Hi guys, I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that: 1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference. 2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears. This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both. I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library. Has anyone run into this? Any advice? Here is the relevant excerpt from my dialplan. <condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems. On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
Has anyone run into this? Any advice?
Here is the relevant excerpt from my dialplan.
<condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765 And it seems to work. Try this in vars.xml <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> I don't think the silence suppression works very well in the browser. On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
Has anyone run into this? Any advice?
Here is the relevant excerpt from my dialplan.
<condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Thanks Anthony! I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side. Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b Would it help if gave you my test ws server and sip address so you could reproduce this? Is it possible that the problem is with my AWS instance or CentOS 5.9? On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765
And it seems to work.
Try this in vars.xml
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
I don't think the silence suppression works very well in the browser.
On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
Has anyone run into this? Any advice?
Here is the relevant excerpt from my dialplan.
<condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master. On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony!
I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
Would it help if gave you my test ws server and sip address so you could reproduce this?
Is it possible that the problem is with my AWS instance or CentOS 5.9?
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765
And it seems to work.
Try this in vars.xml
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
I don't think the silence suppression works very well in the browser.
On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
Has anyone run into this? Any advice?
Here is the relevant excerpt from my dialplan.
<condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there. I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen. However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound. Is there a way to assign conference params per user (like member flags)? On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony!
I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
Would it help if gave you my test ws server and sip address so you could reproduce this?
Is it possible that the problem is with my AWS instance or CentOS 5.9?
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765
And it seems to work.
Try this in vars.xml
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
I don't think the silence suppression works very well in the browser.
On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
Has anyone run into this? Any advice?
Here is the relevant excerpt from my dialplan.
<condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time. On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
Is there a way to assign conference params per user (like member flags)?
On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony!
I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
Would it help if gave you my test ws server and sip address so you could reproduce this?
Is it possible that the problem is with my AWS instance or CentOS 5.9?
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765
And it seems to work.
Try this in vars.xml
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
I don't think the silence suppression works very well in the browser.
On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi guys,
I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
Has anyone run into this? Any advice?
Here is the relevant excerpt from my dialplan.
<condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better. Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag? On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
Is there a way to assign conference params per user (like member flags)?
On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony!
I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
Would it help if gave you my test ws server and sip address so you could reproduce this?
Is it possible that the problem is with my AWS instance or CentOS 5.9?
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765
And it seems to work.
Try this in vars.xml
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
I don't think the silence suppression works very well in the browser.
On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Hi guys, > > I am connecting to FreeSWITCH using WebRTC and after a user calls > in, I play a beep and put them into a conference. Two strange things > happen after that: > > 1. There is a noise in the user's leg that someone described like a > wind in the tunnel. The noise lasts for several seconds, then the user can > hear the hold music. The noise does not seem to happen if I don't play the > beep before the conference. > > 2. About 3 seconds into the call, there is another very short beep - > like a small portion of the original beep that the user hears. > > This absolutely does not happen with softphones - only with WebRTC. > I tried it with Chrome and Opera - same thing is happening on both. > > I tried increasing the answer delay from 2 seconds to 3 and the > noise now happens more rarely but the extra beep is still there almost > every time. I am using JsSIP 3.0 but since it only handles signalling I > don't think the problem is with the JS library. > > Has anyone run into this? Any advice? > > Here is the relevant excerpt from my dialplan. > > <condition field="destination_number" expression="^conf-(\S+)$"> > <action application="set" data="answer_delay=3000"/> > <action application="answer"/> > <action application="playback" > data="tone_stream://%(1000,0,600)" /> > <action application="conference" data="$1@default"/> > </condition> > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
What if you play an audio file? is it the same? I can only guess really. On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" />
Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
Is there a way to assign conference params per user (like member flags)?
On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony!
I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
Would it help if gave you my test ws server and sip address so you could reproduce this?
Is it possible that the problem is with my AWS instance or CentOS 5.9?
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765
And it seems to work.
Try this in vars.xml
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
I don't think the silence suppression works very well in the browser.
On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> In case it was not clear from the previous email - this only happens > with conferences. If after the beep, instead of placing the call into a > conference, I just play some music, there are no problems. > > > On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com > > wrote: > >> Hi guys, >> >> I am connecting to FreeSWITCH using WebRTC and after a user calls >> in, I play a beep and put them into a conference. Two strange things >> happen after that: >> >> 1. There is a noise in the user's leg that someone described like a >> wind in the tunnel. The noise lasts for several seconds, then the user can >> hear the hold music. The noise does not seem to happen if I don't play the >> beep before the conference. >> >> 2. About 3 seconds into the call, there is another very short beep >> - like a small portion of the original beep that the user hears. >> >> This absolutely does not happen with softphones - only with WebRTC. >> I tried it with Chrome and Opera - same thing is happening on both. >> >> I tried increasing the answer delay from 2 seconds to 3 and the >> noise now happens more rarely but the extra beep is still there almost >> every time. I am using JsSIP 3.0 but since it only handles signalling I >> don't think the problem is with the JS library. >> >> Has anyone run into this? Any advice? >> >> Here is the relevant excerpt from my dialplan. >> >> <condition field="destination_number" expression="^conf-(\S+)$"> >> <action application="set" data="answer_delay=3000"/> >> <action application="answer"/> >> <action application="playback" >> data="tone_stream://%(1000,0,600)" /> >> <action application="conference" data="$1@default"/> >> </condition> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Yep it's the same. Thank you Oleg On Jun 4, 2014 4:05 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
What if you play an audio file? is it the same? I can only guess really.
On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" />
Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
Is there a way to assign conference params per user (like member flags)?
On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony!
I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
Would it help if gave you my test ws server and sip address so you could reproduce this?
Is it possible that the problem is with my AWS instance or CentOS 5.9?
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
> I set up your test ext here: > https://webrtc.freeswitch.org/sipjs/index.html#8765 > > And it seems to work. > > Try this in vars.xml > > <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> > > I don't think the silence suppression works very well in the browser. > > > > > > On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > >> In case it was not clear from the previous email - this only >> happens with conferences. If after the beep, instead of placing the call >> into a conference, I just play some music, there are no problems. >> >> >> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >> olegstolyar@gmail.com> wrote: >> >>> Hi guys, >>> >>> I am connecting to FreeSWITCH using WebRTC and after a user calls >>> in, I play a beep and put them into a conference. Two strange things >>> happen after that: >>> >>> 1. There is a noise in the user's leg that someone described like >>> a wind in the tunnel. The noise lasts for several seconds, then the user >>> can hear the hold music. The noise does not seem to happen if I don't play >>> the beep before the conference. >>> >>> 2. About 3 seconds into the call, there is another very short beep >>> - like a small portion of the original beep that the user hears. >>> >>> This absolutely does not happen with softphones - only with >>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>> >>> I tried increasing the answer delay from 2 seconds to 3 and the >>> noise now happens more rarely but the extra beep is still there almost >>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>> don't think the problem is with the JS library. >>> >>> Has anyone run into this? Any advice? >>> >>> Here is the relevant excerpt from my dialplan. >>> >>> <condition field="destination_number" expression="^conf-(\S+)$"> >>> <action application="set" data="answer_delay=3000"/> >>> <action application="answer"/> >>> <action application="playback" >>> data="tone_stream://%(1000,0,600)" /> >>> <action application="conference" data="$1@default"/> >>> </condition> >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Anthony, I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue. It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at. On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yep it's the same.
Thank you Oleg On Jun 4, 2014 4:05 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
What if you play an audio file? is it the same? I can only guess really.
On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" />
Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
Is there a way to assign conference params per user (like member flags)?
On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Thanks Anthony! > > I tried adding this var and also tried uncommenting the suppress_cng > line in my sip profile. It did not seem to make a difference. I pointed > my client code to your sample and it worked fine - no noise, so it's not a > problem with the client side. > > Do I need to upgrade to the latest master? I am using master from > 5/22 version 1.5.13b > > Would it help if gave you my test ws server and sip address so you > could reproduce this? > > Is it possible that the problem is with my AWS instance or CentOS > 5.9? > > > > > On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < > anthony.minessale@gmail.com> wrote: > >> I set up your test ext here: >> https://webrtc.freeswitch.org/sipjs/index.html#8765 >> >> And it seems to work. >> >> Try this in vars.xml >> >> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >> >> I don't think the silence suppression works very well in the >> browser. >> >> >> >> >> >> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com >> > wrote: >> >>> In case it was not clear from the previous email - this only >>> happens with conferences. If after the beep, instead of placing the call >>> into a conference, I just play some music, there are no problems. >>> >>> >>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> Hi guys, >>>> >>>> I am connecting to FreeSWITCH using WebRTC and after a user calls >>>> in, I play a beep and put them into a conference. Two strange things >>>> happen after that: >>>> >>>> 1. There is a noise in the user's leg that someone described like >>>> a wind in the tunnel. The noise lasts for several seconds, then the user >>>> can hear the hold music. The noise does not seem to happen if I don't play >>>> the beep before the conference. >>>> >>>> 2. About 3 seconds into the call, there is another very short >>>> beep - like a small portion of the original beep that the user hears. >>>> >>>> This absolutely does not happen with softphones - only with >>>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>>> >>>> I tried increasing the answer delay from 2 seconds to 3 and the >>>> noise now happens more rarely but the extra beep is still there almost >>>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>>> don't think the problem is with the JS library. >>>> >>>> Has anyone run into this? Any advice? >>>> >>>> Here is the relevant excerpt from my dialplan. >>>> >>>> <condition field="destination_number" expression="^conf-(\S+)$"> >>>> <action application="set" data="answer_delay=3000"/> >>>> <action application="answer"/> >>>> <action application="playback" >>>> data="tone_stream://%(1000,0,600)" /> >>>> <action application="conference" data="$1@default"/> >>>> </condition> >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Anthony, Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well. I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using? On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yep it's the same.
Thank you Oleg On Jun 4, 2014 4:05 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
What if you play an audio file? is it the same? I can only guess really.
On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" />
Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
Is there a way to assign conference params per user (like member flags)?
On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
> If you opt to use master, ALWAYS use latest master since that is the > nature of master. I have no idea if your problem is related to those > things but the server I pointed you to test is running basically latest > master. > > > > On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > >> Thanks Anthony! >> >> I tried adding this var and also tried uncommenting the >> suppress_cng line in my sip profile. It did not seem to make a difference. >> I pointed my client code to your sample and it worked fine - no noise, so >> it's not a problem with the client side. >> >> Do I need to upgrade to the latest master? I am using master from >> 5/22 version 1.5.13b >> >> Would it help if gave you my test ws server and sip address so you >> could reproduce this? >> >> Is it possible that the problem is with my AWS instance or CentOS >> 5.9? >> >> >> >> >> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >> anthony.minessale@gmail.com> wrote: >> >>> I set up your test ext here: >>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>> >>> And it seems to work. >>> >>> Try this in vars.xml >>> >>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>> >>> I don't think the silence suppression works very well in the >>> browser. >>> >>> >>> >>> >>> >>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> In case it was not clear from the previous email - this only >>>> happens with conferences. If after the beep, instead of placing the call >>>> into a conference, I just play some music, there are no problems. >>>> >>>> >>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> Hi guys, >>>>> >>>>> I am connecting to FreeSWITCH using WebRTC and after a user >>>>> calls in, I play a beep and put them into a conference. Two strange things >>>>> happen after that: >>>>> >>>>> 1. There is a noise in the user's leg that someone described >>>>> like a wind in the tunnel. The noise lasts for several seconds, then the >>>>> user can hear the hold music. The noise does not seem to happen if I don't >>>>> play the beep before the conference. >>>>> >>>>> 2. About 3 seconds into the call, there is another very short >>>>> beep - like a small portion of the original beep that the user hears. >>>>> >>>>> This absolutely does not happen with softphones - only with >>>>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>>>> >>>>> I tried increasing the answer delay from 2 seconds to 3 and the >>>>> noise now happens more rarely but the extra beep is still there almost >>>>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>>>> don't think the problem is with the JS library. >>>>> >>>>> Has anyone run into this? Any advice? >>>>> >>>>> Here is the relevant excerpt from my dialplan. >>>>> >>>>> <condition field="destination_number" expression="^conf-(\S+)$"> >>>>> <action application="set" data="answer_delay=3000"/> >>>>> <action application="answer"/> >>>>> <action application="playback" >>>>> data="tone_stream://%(1000,0,600)" /> >>>>> <action application="conference" data="$1@default"/> >>>>> </condition> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>> >>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>> http://twitter.com/FreeSWITCH >>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>> <http://freeswitch.org/g+>* >>> >>> ClueCon Weekly Development Call >>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Anthony,
Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yep it's the same.
Thank you Oleg On Jun 4, 2014 4:05 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
What if you play an audio file? is it the same? I can only guess really.
On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" />
Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> OK, thanks. It's still happening with the latest master, so I have > to assume that this is something to do with the AWS VMs. It would not be > the first time strange things happen there. > > I was trying to work around the problem and noticed that if instead > of playing the beep using playback app before placing the user into the > conference, I make the beep the enter-sound of the conference, the problem > does not happen. > > However, I need to only play this sound for users joining the > conference from one dialplan but not from another. I cannot find a way to > do it. I looked at > <action application="set" > data="conference_enter_sound=silence_stream://10"/> > but it changes the enter sound for the conference for everyone, so > anyone entering after the user who set this up will hear the same sound. > > Is there a way to assign conference params per user (like member > flags)? > > > On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < > anthony.minessale@gmail.com> wrote: > >> If you opt to use master, ALWAYS use latest master since that is >> the nature of master. I have no idea if your problem is related to those >> things but the server I pointed you to test is running basically latest >> master. >> >> >> >> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com >> > wrote: >> >>> Thanks Anthony! >>> >>> I tried adding this var and also tried uncommenting the >>> suppress_cng line in my sip profile. It did not seem to make a difference. >>> I pointed my client code to your sample and it worked fine - no noise, so >>> it's not a problem with the client side. >>> >>> Do I need to upgrade to the latest master? I am using master from >>> 5/22 version 1.5.13b >>> >>> Would it help if gave you my test ws server and sip address so you >>> could reproduce this? >>> >>> Is it possible that the problem is with my AWS instance or CentOS >>> 5.9? >>> >>> >>> >>> >>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>> anthony.minessale@gmail.com> wrote: >>> >>>> I set up your test ext here: >>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>> >>>> And it seems to work. >>>> >>>> Try this in vars.xml >>>> >>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>> >>>> I don't think the silence suppression works very well in the >>>> browser. >>>> >>>> >>>> >>>> >>>> >>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> In case it was not clear from the previous email - this only >>>>> happens with conferences. If after the beep, instead of placing the call >>>>> into a conference, I just play some music, there are no problems. >>>>> >>>>> >>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>> olegstolyar@gmail.com> wrote: >>>>> >>>>>> Hi guys, >>>>>> >>>>>> I am connecting to FreeSWITCH using WebRTC and after a user >>>>>> calls in, I play a beep and put them into a conference. Two strange things >>>>>> happen after that: >>>>>> >>>>>> 1. There is a noise in the user's leg that someone described >>>>>> like a wind in the tunnel. The noise lasts for several seconds, then the >>>>>> user can hear the hold music. The noise does not seem to happen if I don't >>>>>> play the beep before the conference. >>>>>> >>>>>> 2. About 3 seconds into the call, there is another very short >>>>>> beep - like a small portion of the original beep that the user hears. >>>>>> >>>>>> This absolutely does not happen with softphones - only with >>>>>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>>>>> >>>>>> I tried increasing the answer delay from 2 seconds to 3 and the >>>>>> noise now happens more rarely but the extra beep is still there almost >>>>>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>>>>> don't think the problem is with the JS library. >>>>>> >>>>>> Has anyone run into this? Any advice? >>>>>> >>>>>> Here is the relevant excerpt from my dialplan. >>>>>> >>>>>> <condition field="destination_number" expression="^conf-(\S+)$"> >>>>>> <action application="set" data="answer_delay=3000"/> >>>>>> <action application="answer"/> >>>>>> <action application="playback" >>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>> <action application="conference" data="$1@default"/> >>>>>> </condition> >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>> >>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>> http://twitter.com/FreeSWITCH >>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>> <http://freeswitch.org/g+>* >>>> >>>> ClueCon Weekly Development Call >>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Also it's using sipjs.com not jssip. On Jun 8, 2014 8:14 AM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Anthony,
Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yep it's the same.
Thank you Oleg On Jun 4, 2014 4:05 PM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
What if you play an audio file? is it the same? I can only guess really.
On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" />
Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
> Try setting the channel var timer_name=soft before playing the beep. > It may be jittery media flow causing your problem. > My only advice on amazon is use the more expensive one because the > lower cost ones are not reliable for cpu time. > > > > On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com > > wrote: > >> OK, thanks. It's still happening with the latest master, so I have >> to assume that this is something to do with the AWS VMs. It would not be >> the first time strange things happen there. >> >> I was trying to work around the problem and noticed that if instead >> of playing the beep using playback app before placing the user into the >> conference, I make the beep the enter-sound of the conference, the problem >> does not happen. >> >> However, I need to only play this sound for users joining the >> conference from one dialplan but not from another. I cannot find a way to >> do it. I looked at >> <action application="set" >> data="conference_enter_sound=silence_stream://10"/> >> but it changes the enter sound for the conference for everyone, so >> anyone entering after the user who set this up will hear the same sound. >> >> Is there a way to assign conference params per user (like member >> flags)? >> >> >> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >> anthony.minessale@gmail.com> wrote: >> >>> If you opt to use master, ALWAYS use latest master since that is >>> the nature of master. I have no idea if your problem is related to those >>> things but the server I pointed you to test is running basically latest >>> master. >>> >>> >>> >>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> Thanks Anthony! >>>> >>>> I tried adding this var and also tried uncommenting the >>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>> I pointed my client code to your sample and it worked fine - no noise, so >>>> it's not a problem with the client side. >>>> >>>> Do I need to upgrade to the latest master? I am using master >>>> from 5/22 version 1.5.13b >>>> >>>> Would it help if gave you my test ws server and sip address so >>>> you could reproduce this? >>>> >>>> Is it possible that the problem is with my AWS instance or CentOS >>>> 5.9? >>>> >>>> >>>> >>>> >>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>> anthony.minessale@gmail.com> wrote: >>>> >>>>> I set up your test ext here: >>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>> >>>>> And it seems to work. >>>>> >>>>> Try this in vars.xml >>>>> >>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>> >>>>> I don't think the silence suppression works very well in the >>>>> browser. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>> olegstolyar@gmail.com> wrote: >>>>> >>>>>> In case it was not clear from the previous email - this only >>>>>> happens with conferences. If after the beep, instead of placing the call >>>>>> into a conference, I just play some music, there are no problems. >>>>>> >>>>>> >>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>> olegstolyar@gmail.com> wrote: >>>>>> >>>>>>> Hi guys, >>>>>>> >>>>>>> I am connecting to FreeSWITCH using WebRTC and after a user >>>>>>> calls in, I play a beep and put them into a conference. Two strange things >>>>>>> happen after that: >>>>>>> >>>>>>> 1. There is a noise in the user's leg that someone described >>>>>>> like a wind in the tunnel. The noise lasts for several seconds, then the >>>>>>> user can hear the hold music. The noise does not seem to happen if I don't >>>>>>> play the beep before the conference. >>>>>>> >>>>>>> 2. About 3 seconds into the call, there is another very short >>>>>>> beep - like a small portion of the original beep that the user hears. >>>>>>> >>>>>>> This absolutely does not happen with softphones - only with >>>>>>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>>>>>> >>>>>>> I tried increasing the answer delay from 2 seconds to 3 and >>>>>>> the noise now happens more rarely but the extra beep is still there almost >>>>>>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>>>>>> don't think the problem is with the JS library. >>>>>>> >>>>>>> Has anyone run into this? Any advice? >>>>>>> >>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>> >>>>>>> <condition field="destination_number" >>>>>>> expression="^conf-(\S+)$"> >>>>>>> <action application="set" data="answer_delay=3000"/> >>>>>>> <action application="answer"/> >>>>>>> <action application="playback" >>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>> <action application="conference" data="$1@default"/> >>>>>>> </condition> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>> >>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>> http://twitter.com/FreeSWITCH >>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>> <http://freeswitch.org/g+>* >>>>> >>>>> ClueCon Weekly Development Call >>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>> >>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>> http://twitter.com/FreeSWITCH >>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>> <http://freeswitch.org/g+>* >>> >>> ClueCon Weekly Development Call >>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging. On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Also it's using sipjs.com not jssip. On Jun 8, 2014 8:14 AM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Anthony,
Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yep it's the same.
Thank you Oleg On Jun 4, 2014 4:05 PM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
What if you play an audio file? is it the same? I can only guess really.
On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> I tried this: > <action application="set" data="timer_name=soft"/> > <action application="playback" > data="tone_stream://%(1000,0,600)" /> > > Didn't help unfortunately (unless this is the wrong syntax for > setting the variable). Completely agreed on the AWS instances - in fact we > started on an older m1 instance and the voice quality was pretty bad and > there were frequent delays. Then we switch to an m3 and it was much better. > > Any advice on setting enter-sound per member? Or setting a > no-enter-sound flag - similar to the nomoh flag? > > > > On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < > anthony.minessale@gmail.com> wrote: > >> Try setting the channel var timer_name=soft before playing the >> beep. It may be jittery media flow causing your problem. >> My only advice on amazon is use the more expensive one because the >> lower cost ones are not reliable for cpu time. >> >> >> >> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >> olegstolyar@gmail.com> wrote: >> >>> OK, thanks. It's still happening with the latest master, so I >>> have to assume that this is something to do with the AWS VMs. It would not >>> be the first time strange things happen there. >>> >>> I was trying to work around the problem and noticed that if >>> instead of playing the beep using playback app before placing the user into >>> the conference, I make the beep the enter-sound of the conference, the >>> problem does not happen. >>> >>> However, I need to only play this sound for users joining the >>> conference from one dialplan but not from another. I cannot find a way to >>> do it. I looked at >>> <action application="set" >>> data="conference_enter_sound=silence_stream://10"/> >>> but it changes the enter sound for the conference for everyone, so >>> anyone entering after the user who set this up will hear the same sound. >>> >>> Is there a way to assign conference params per user (like member >>> flags)? >>> >>> >>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>> anthony.minessale@gmail.com> wrote: >>> >>>> If you opt to use master, ALWAYS use latest master since that is >>>> the nature of master. I have no idea if your problem is related to those >>>> things but the server I pointed you to test is running basically latest >>>> master. >>>> >>>> >>>> >>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> Thanks Anthony! >>>>> >>>>> I tried adding this var and also tried uncommenting the >>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>> it's not a problem with the client side. >>>>> >>>>> Do I need to upgrade to the latest master? I am using master >>>>> from 5/22 version 1.5.13b >>>>> >>>>> Would it help if gave you my test ws server and sip address so >>>>> you could reproduce this? >>>>> >>>>> Is it possible that the problem is with my AWS instance or >>>>> CentOS 5.9? >>>>> >>>>> >>>>> >>>>> >>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>> anthony.minessale@gmail.com> wrote: >>>>> >>>>>> I set up your test ext here: >>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>> >>>>>> And it seems to work. >>>>>> >>>>>> Try this in vars.xml >>>>>> >>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>> >>>>>> I don't think the silence suppression works very well in the >>>>>> browser. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>> olegstolyar@gmail.com> wrote: >>>>>> >>>>>>> In case it was not clear from the previous email - this only >>>>>>> happens with conferences. If after the beep, instead of placing the call >>>>>>> into a conference, I just play some music, there are no problems. >>>>>>> >>>>>>> >>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> Hi guys, >>>>>>>> >>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a user >>>>>>>> calls in, I play a beep and put them into a conference. Two strange things >>>>>>>> happen after that: >>>>>>>> >>>>>>>> 1. There is a noise in the user's leg that someone described >>>>>>>> like a wind in the tunnel. The noise lasts for several seconds, then the >>>>>>>> user can hear the hold music. The noise does not seem to happen if I don't >>>>>>>> play the beep before the conference. >>>>>>>> >>>>>>>> 2. About 3 seconds into the call, there is another very short >>>>>>>> beep - like a small portion of the original beep that the user hears. >>>>>>>> >>>>>>>> This absolutely does not happen with softphones - only with >>>>>>>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>>>>>>> >>>>>>>> I tried increasing the answer delay from 2 seconds to 3 and >>>>>>>> the noise now happens more rarely but the extra beep is still there almost >>>>>>>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>>>>>>> don't think the problem is with the JS library. >>>>>>>> >>>>>>>> Has anyone run into this? Any advice? >>>>>>>> >>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>> >>>>>>>> <condition field="destination_number" >>>>>>>> expression="^conf-(\S+)$"> >>>>>>>> <action application="set" data="answer_delay=3000"/> >>>>>>>> <action application="answer"/> >>>>>>>> <action application="playback" >>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>> <action application="conference" data="$1@default"/> >>>>>>>> </condition> >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>> >>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>> http://twitter.com/FreeSWITCH >>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>> <http://freeswitch.org/g+>* >>>>>> >>>>>> ClueCon Weekly Development Call >>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>> >>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>> http://twitter.com/FreeSWITCH >>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>> <http://freeswitch.org/g+>* >>>> >>>> ClueCon Weekly Development Call >>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hey Anthony, just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example? On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Also it's using sipjs.com not jssip. On Jun 8, 2014 8:14 AM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Anthony,
Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yep it's the same.
Thank you Oleg On Jun 4, 2014 4:05 PM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
> What if you play an audio file? is it the same? > I can only guess really. > > > > On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com > > wrote: > >> I tried this: >> <action application="set" data="timer_name=soft"/> >> <action application="playback" >> data="tone_stream://%(1000,0,600)" /> >> >> Didn't help unfortunately (unless this is the wrong syntax for >> setting the variable). Completely agreed on the AWS instances - in fact we >> started on an older m1 instance and the voice quality was pretty bad and >> there were frequent delays. Then we switch to an m3 and it was much better. >> >> Any advice on setting enter-sound per member? Or setting a >> no-enter-sound flag - similar to the nomoh flag? >> >> >> >> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >> anthony.minessale@gmail.com> wrote: >> >>> Try setting the channel var timer_name=soft before playing the >>> beep. It may be jittery media flow causing your problem. >>> My only advice on amazon is use the more expensive one because the >>> lower cost ones are not reliable for cpu time. >>> >>> >>> >>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> OK, thanks. It's still happening with the latest master, so I >>>> have to assume that this is something to do with the AWS VMs. It would not >>>> be the first time strange things happen there. >>>> >>>> I was trying to work around the problem and noticed that if >>>> instead of playing the beep using playback app before placing the user into >>>> the conference, I make the beep the enter-sound of the conference, the >>>> problem does not happen. >>>> >>>> However, I need to only play this sound for users joining the >>>> conference from one dialplan but not from another. I cannot find a way to >>>> do it. I looked at >>>> <action application="set" >>>> data="conference_enter_sound=silence_stream://10"/> >>>> but it changes the enter sound for the conference for everyone, >>>> so anyone entering after the user who set this up will hear the same sound. >>>> >>>> Is there a way to assign conference params per user (like member >>>> flags)? >>>> >>>> >>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>> anthony.minessale@gmail.com> wrote: >>>> >>>>> If you opt to use master, ALWAYS use latest master since that is >>>>> the nature of master. I have no idea if your problem is related to those >>>>> things but the server I pointed you to test is running basically latest >>>>> master. >>>>> >>>>> >>>>> >>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>> olegstolyar@gmail.com> wrote: >>>>> >>>>>> Thanks Anthony! >>>>>> >>>>>> I tried adding this var and also tried uncommenting the >>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>> it's not a problem with the client side. >>>>>> >>>>>> Do I need to upgrade to the latest master? I am using master >>>>>> from 5/22 version 1.5.13b >>>>>> >>>>>> Would it help if gave you my test ws server and sip address so >>>>>> you could reproduce this? >>>>>> >>>>>> Is it possible that the problem is with my AWS instance or >>>>>> CentOS 5.9? >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>> anthony.minessale@gmail.com> wrote: >>>>>> >>>>>>> I set up your test ext here: >>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>> >>>>>>> And it seems to work. >>>>>>> >>>>>>> Try this in vars.xml >>>>>>> >>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>> >>>>>>> I don't think the silence suppression works very well in the >>>>>>> browser. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> In case it was not clear from the previous email - this only >>>>>>>> happens with conferences. If after the beep, instead of placing the call >>>>>>>> into a conference, I just play some music, there are no problems. >>>>>>>> >>>>>>>> >>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi guys, >>>>>>>>> >>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a user >>>>>>>>> calls in, I play a beep and put them into a conference. Two strange things >>>>>>>>> happen after that: >>>>>>>>> >>>>>>>>> 1. There is a noise in the user's leg that someone described >>>>>>>>> like a wind in the tunnel. The noise lasts for several seconds, then the >>>>>>>>> user can hear the hold music. The noise does not seem to happen if I don't >>>>>>>>> play the beep before the conference. >>>>>>>>> >>>>>>>>> 2. About 3 seconds into the call, there is another very >>>>>>>>> short beep - like a small portion of the original beep that the user hears. >>>>>>>>> >>>>>>>>> This absolutely does not happen with softphones - only with >>>>>>>>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>>>>>>>> >>>>>>>>> I tried increasing the answer delay from 2 seconds to 3 and >>>>>>>>> the noise now happens more rarely but the extra beep is still there almost >>>>>>>>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>>>>>>>> don't think the problem is with the JS library. >>>>>>>>> >>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>> >>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>> >>>>>>>>> <condition field="destination_number" >>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>> <action application="set" data="answer_delay=3000"/> >>>>>>>>> <action application="answer"/> >>>>>>>>> <action application="playback" >>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>> <action application="conference" data="$1@default"/> >>>>>>>>> </condition> >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>> >>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>> http://twitter.com/FreeSWITCH >>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>> <http://freeswitch.org/g+>* >>>>>>> >>>>>>> ClueCon Weekly Development Call >>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>> >>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>> http://twitter.com/FreeSWITCH >>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>> <http://freeswitch.org/g+>* >>>>> >>>>> ClueCon Weekly Development Call >>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>> >>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>> http://twitter.com/FreeSWITCH >>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>> <http://freeswitch.org/g+>* >>> >>> ClueCon Weekly Development Call >>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Anthony, could you send me the XML for the extension you used to set up this example? On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hey Anthony,
just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Also it's using sipjs.com not jssip. On Jun 8, 2014 8:14 AM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Anthony,
Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Yep it's the same. > > Thank you > Oleg > On Jun 4, 2014 4:05 PM, "Anthony Minessale" < > anthony.minessale@gmail.com> wrote: > >> What if you play an audio file? is it the same? >> I can only guess really. >> >> >> >> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >> olegstolyar@gmail.com> wrote: >> >>> I tried this: >>> <action application="set" data="timer_name=soft"/> >>> <action application="playback" >>> data="tone_stream://%(1000,0,600)" /> >>> >>> Didn't help unfortunately (unless this is the wrong syntax for >>> setting the variable). Completely agreed on the AWS instances - in fact we >>> started on an older m1 instance and the voice quality was pretty bad and >>> there were frequent delays. Then we switch to an m3 and it was much better. >>> >>> Any advice on setting enter-sound per member? Or setting a >>> no-enter-sound flag - similar to the nomoh flag? >>> >>> >>> >>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>> anthony.minessale@gmail.com> wrote: >>> >>>> Try setting the channel var timer_name=soft before playing the >>>> beep. It may be jittery media flow causing your problem. >>>> My only advice on amazon is use the more expensive one because >>>> the lower cost ones are not reliable for cpu time. >>>> >>>> >>>> >>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> OK, thanks. It's still happening with the latest master, so I >>>>> have to assume that this is something to do with the AWS VMs. It would not >>>>> be the first time strange things happen there. >>>>> >>>>> I was trying to work around the problem and noticed that if >>>>> instead of playing the beep using playback app before placing the user into >>>>> the conference, I make the beep the enter-sound of the conference, the >>>>> problem does not happen. >>>>> >>>>> However, I need to only play this sound for users joining the >>>>> conference from one dialplan but not from another. I cannot find a way to >>>>> do it. I looked at >>>>> <action application="set" >>>>> data="conference_enter_sound=silence_stream://10"/> >>>>> but it changes the enter sound for the conference for everyone, >>>>> so anyone entering after the user who set this up will hear the same sound. >>>>> >>>>> Is there a way to assign conference params per user (like member >>>>> flags)? >>>>> >>>>> >>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>> anthony.minessale@gmail.com> wrote: >>>>> >>>>>> If you opt to use master, ALWAYS use latest master since that >>>>>> is the nature of master. I have no idea if your problem is related to >>>>>> those things but the server I pointed you to test is running basically >>>>>> latest master. >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>> olegstolyar@gmail.com> wrote: >>>>>> >>>>>>> Thanks Anthony! >>>>>>> >>>>>>> I tried adding this var and also tried uncommenting the >>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>> it's not a problem with the client side. >>>>>>> >>>>>>> Do I need to upgrade to the latest master? I am using master >>>>>>> from 5/22 version 1.5.13b >>>>>>> >>>>>>> Would it help if gave you my test ws server and sip address so >>>>>>> you could reproduce this? >>>>>>> >>>>>>> Is it possible that the problem is with my AWS instance or >>>>>>> CentOS 5.9? >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>> >>>>>>>> I set up your test ext here: >>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>> >>>>>>>> And it seems to work. >>>>>>>> >>>>>>>> Try this in vars.xml >>>>>>>> >>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>> >>>>>>>> I don't think the silence suppression works very well in the >>>>>>>> browser. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>> >>>>>>>>> In case it was not clear from the previous email - this only >>>>>>>>> happens with conferences. If after the beep, instead of placing the call >>>>>>>>> into a conference, I just play some music, there are no problems. >>>>>>>>> >>>>>>>>> >>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Hi guys, >>>>>>>>>> >>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a user >>>>>>>>>> calls in, I play a beep and put them into a conference. Two strange things >>>>>>>>>> happen after that: >>>>>>>>>> >>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>> >>>>>>>>>> 2. About 3 seconds into the call, there is another very >>>>>>>>>> short beep - like a small portion of the original beep that the user hears. >>>>>>>>>> >>>>>>>>>> This absolutely does not happen with softphones - only with >>>>>>>>>> WebRTC. I tried it with Chrome and Opera - same thing is happening on both. >>>>>>>>>> >>>>>>>>>> I tried increasing the answer delay from 2 seconds to 3 and >>>>>>>>>> the noise now happens more rarely but the extra beep is still there almost >>>>>>>>>> every time. I am using JsSIP 3.0 but since it only handles signalling I >>>>>>>>>> don't think the problem is with the JS library. >>>>>>>>>> >>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>> >>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>> >>>>>>>>>> <condition field="destination_number" >>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>> <action application="set" data="answer_delay=3000"/> >>>>>>>>>> <action application="answer"/> >>>>>>>>>> <action application="playback" >>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>> <action application="conference" data="$1@default >>>>>>>>>> "/> >>>>>>>>>> </condition> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>> >>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>> <http://freeswitch.org/g+>* >>>>>>>> >>>>>>>> ClueCon Weekly Development Call >>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>> >>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>> http://twitter.com/FreeSWITCH >>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>> <http://freeswitch.org/g+>* >>>>>> >>>>>> ClueCon Weekly Development Call >>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>> >>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>> http://twitter.com/FreeSWITCH >>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>> <http://freeswitch.org/g+>* >>>> >>>> ClueCon Weekly Development Call >>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
<extension name="8765"> <condition field="destination_number" expression="^8765$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition> </extension> On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
could you send me the XML for the extension you used to set up this example?
On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hey Anthony,
just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Also it's using sipjs.com not jssip. On Jun 8, 2014 8:14 AM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Anthony,
Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Hi Anthony, > > I dug further and I can reproduce the issue on a completely > different VM (Debian on Windows host). I installed the latest FS master > and even switched to the vanilla conf to eliminate the possibility of my > config causing this issue. > > It's still happening. If you have time to look into this some more, > I'll be happy to upload my whole VM for someone to take a look at. > > > On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > >> Yep it's the same. >> >> Thank you >> Oleg >> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >> anthony.minessale@gmail.com> wrote: >> >>> What if you play an audio file? is it the same? >>> I can only guess really. >>> >>> >>> >>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> I tried this: >>>> <action application="set" data="timer_name=soft"/> >>>> <action application="playback" >>>> data="tone_stream://%(1000,0,600)" /> >>>> >>>> Didn't help unfortunately (unless this is the wrong syntax for >>>> setting the variable). Completely agreed on the AWS instances - in fact we >>>> started on an older m1 instance and the voice quality was pretty bad and >>>> there were frequent delays. Then we switch to an m3 and it was much better. >>>> >>>> Any advice on setting enter-sound per member? Or setting a >>>> no-enter-sound flag - similar to the nomoh flag? >>>> >>>> >>>> >>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>> anthony.minessale@gmail.com> wrote: >>>> >>>>> Try setting the channel var timer_name=soft before playing the >>>>> beep. It may be jittery media flow causing your problem. >>>>> My only advice on amazon is use the more expensive one because >>>>> the lower cost ones are not reliable for cpu time. >>>>> >>>>> >>>>> >>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>> olegstolyar@gmail.com> wrote: >>>>> >>>>>> OK, thanks. It's still happening with the latest master, so I >>>>>> have to assume that this is something to do with the AWS VMs. It would not >>>>>> be the first time strange things happen there. >>>>>> >>>>>> I was trying to work around the problem and noticed that if >>>>>> instead of playing the beep using playback app before placing the user into >>>>>> the conference, I make the beep the enter-sound of the conference, the >>>>>> problem does not happen. >>>>>> >>>>>> However, I need to only play this sound for users joining the >>>>>> conference from one dialplan but not from another. I cannot find a way to >>>>>> do it. I looked at >>>>>> <action application="set" >>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>> but it changes the enter sound for the conference for everyone, >>>>>> so anyone entering after the user who set this up will hear the same sound. >>>>>> >>>>>> Is there a way to assign conference params per user (like >>>>>> member flags)? >>>>>> >>>>>> >>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>> anthony.minessale@gmail.com> wrote: >>>>>> >>>>>>> If you opt to use master, ALWAYS use latest master since that >>>>>>> is the nature of master. I have no idea if your problem is related to >>>>>>> those things but the server I pointed you to test is running basically >>>>>>> latest master. >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> Thanks Anthony! >>>>>>>> >>>>>>>> I tried adding this var and also tried uncommenting the >>>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>>> it's not a problem with the client side. >>>>>>>> >>>>>>>> Do I need to upgrade to the latest master? I am using master >>>>>>>> from 5/22 version 1.5.13b >>>>>>>> >>>>>>>> Would it help if gave you my test ws server and sip address >>>>>>>> so you could reproduce this? >>>>>>>> >>>>>>>> Is it possible that the problem is with my AWS instance or >>>>>>>> CentOS 5.9? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>> >>>>>>>>> I set up your test ext here: >>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>> >>>>>>>>> And it seems to work. >>>>>>>>> >>>>>>>>> Try this in vars.xml >>>>>>>>> >>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>> >>>>>>>>> I don't think the silence suppression works very well in the >>>>>>>>> browser. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> In case it was not clear from the previous email - this >>>>>>>>>> only happens with conferences. If after the beep, instead of placing the >>>>>>>>>> call into a conference, I just play some music, there are no problems. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Hi guys, >>>>>>>>>>> >>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a >>>>>>>>>>> user calls in, I play a beep and put them into a conference. Two strange >>>>>>>>>>> things happen after that: >>>>>>>>>>> >>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>> >>>>>>>>>>> 2. About 3 seconds into the call, there is another very >>>>>>>>>>> short beep - like a small portion of the original beep that the user hears. >>>>>>>>>>> >>>>>>>>>>> This absolutely does not happen with softphones - only >>>>>>>>>>> with WebRTC. I tried it with Chrome and Opera - same thing is happening on >>>>>>>>>>> both. >>>>>>>>>>> >>>>>>>>>>> I tried increasing the answer delay from 2 seconds to 3 >>>>>>>>>>> and the noise now happens more rarely but the extra beep is still there >>>>>>>>>>> almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>> >>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>> >>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>> >>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>> <action application="set" >>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>> <action application="answer"/> >>>>>>>>>>> <action application="playback" >>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>> <action application="conference" data="$1@default >>>>>>>>>>> "/> >>>>>>>>>>> </condition> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>> >>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>> >>>>>>>>> ClueCon Weekly Development Call >>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>> >>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>> http://twitter.com/FreeSWITCH >>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>> <http://freeswitch.org/g+>* >>>>>>> >>>>>>> ClueCon Weekly Development Call >>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>> >>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>> http://twitter.com/FreeSWITCH >>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>> <http://freeswitch.org/g+>* >>>>> >>>>> ClueCon Weekly Development Call >>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>> >>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>> http://twitter.com/FreeSWITCH >>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>> <http://freeswitch.org/g+>* >>> >>> ClueCon Weekly Development Call >>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad. I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH. Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening. On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
<extension name="8765">
<condition field="destination_number" expression="^8765$">
<action application="set" data="answer_delay=3000"/>
<action application="answer"/>
<action application="playback" data="tone_stream://%(1000,0,600)" />
<action application="conference" data="$1@default"/>
</condition>
</extension>
On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
could you send me the XML for the extension you used to set up this example?
On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hey Anthony,
just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Also it's using sipjs.com not jssip. On Jun 8, 2014 8:14 AM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
> Anthony, > > Another update - I thought that the only thing my original > installation and the Debian VM had in common was that both were VMs. So I > went ahead and installed FreeSWITCH on my local Windows machine. Same > problem is happening here as well. > > I am extremely curious what's different between your installation > and my three setups (CentOS, Debian and Windows) that accounts for the > difference in this behavior. Could you tell me what machine/OS you are > using? > > > On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > >> Hi Anthony, >> >> I dug further and I can reproduce the issue on a completely >> different VM (Debian on Windows host). I installed the latest FS master >> and even switched to the vanilla conf to eliminate the possibility of my >> config causing this issue. >> >> It's still happening. If you have time to look into this some >> more, I'll be happy to upload my whole VM for someone to take a look at. >> >> >> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com >> > wrote: >> >>> Yep it's the same. >>> >>> Thank you >>> Oleg >>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>> anthony.minessale@gmail.com> wrote: >>> >>>> What if you play an audio file? is it the same? >>>> I can only guess really. >>>> >>>> >>>> >>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> I tried this: >>>>> <action application="set" data="timer_name=soft"/> >>>>> <action application="playback" >>>>> data="tone_stream://%(1000,0,600)" /> >>>>> >>>>> Didn't help unfortunately (unless this is the wrong syntax for >>>>> setting the variable). Completely agreed on the AWS instances - in fact we >>>>> started on an older m1 instance and the voice quality was pretty bad and >>>>> there were frequent delays. Then we switch to an m3 and it was much better. >>>>> >>>>> Any advice on setting enter-sound per member? Or setting a >>>>> no-enter-sound flag - similar to the nomoh flag? >>>>> >>>>> >>>>> >>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>> anthony.minessale@gmail.com> wrote: >>>>> >>>>>> Try setting the channel var timer_name=soft before playing the >>>>>> beep. It may be jittery media flow causing your problem. >>>>>> My only advice on amazon is use the more expensive one because >>>>>> the lower cost ones are not reliable for cpu time. >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>> olegstolyar@gmail.com> wrote: >>>>>> >>>>>>> OK, thanks. It's still happening with the latest master, so I >>>>>>> have to assume that this is something to do with the AWS VMs. It would not >>>>>>> be the first time strange things happen there. >>>>>>> >>>>>>> I was trying to work around the problem and noticed that if >>>>>>> instead of playing the beep using playback app before placing the user into >>>>>>> the conference, I make the beep the enter-sound of the conference, the >>>>>>> problem does not happen. >>>>>>> >>>>>>> However, I need to only play this sound for users joining the >>>>>>> conference from one dialplan but not from another. I cannot find a way to >>>>>>> do it. I looked at >>>>>>> <action application="set" >>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>> but it changes the enter sound for the conference for >>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>> same sound. >>>>>>> >>>>>>> Is there a way to assign conference params per user (like >>>>>>> member flags)? >>>>>>> >>>>>>> >>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>> >>>>>>>> If you opt to use master, ALWAYS use latest master since that >>>>>>>> is the nature of master. I have no idea if your problem is related to >>>>>>>> those things but the server I pointed you to test is running basically >>>>>>>> latest master. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>> >>>>>>>>> Thanks Anthony! >>>>>>>>> >>>>>>>>> I tried adding this var and also tried uncommenting the >>>>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>>>> it's not a problem with the client side. >>>>>>>>> >>>>>>>>> Do I need to upgrade to the latest master? I am using >>>>>>>>> master from 5/22 version 1.5.13b >>>>>>>>> >>>>>>>>> Would it help if gave you my test ws server and sip address >>>>>>>>> so you could reproduce this? >>>>>>>>> >>>>>>>>> Is it possible that the problem is with my AWS instance or >>>>>>>>> CentOS 5.9? >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> I set up your test ext here: >>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>> >>>>>>>>>> And it seems to work. >>>>>>>>>> >>>>>>>>>> Try this in vars.xml >>>>>>>>>> >>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>> >>>>>>>>>> I don't think the silence suppression works very well in >>>>>>>>>> the browser. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> In case it was not clear from the previous email - this >>>>>>>>>>> only happens with conferences. If after the beep, instead of placing the >>>>>>>>>>> call into a conference, I just play some music, there are no problems. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Hi guys, >>>>>>>>>>>> >>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a >>>>>>>>>>>> user calls in, I play a beep and put them into a conference. Two strange >>>>>>>>>>>> things happen after that: >>>>>>>>>>>> >>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>> >>>>>>>>>>>> 2. About 3 seconds into the call, there is another very >>>>>>>>>>>> short beep - like a small portion of the original beep that the user hears. >>>>>>>>>>>> >>>>>>>>>>>> This absolutely does not happen with softphones - only >>>>>>>>>>>> with WebRTC. I tried it with Chrome and Opera - same thing is happening on >>>>>>>>>>>> both. >>>>>>>>>>>> >>>>>>>>>>>> I tried increasing the answer delay from 2 seconds to 3 >>>>>>>>>>>> and the noise now happens more rarely but the extra beep is still there >>>>>>>>>>>> almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>> >>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>> >>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>> >>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>> <action application="set" >>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>> <action application="playback" >>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>> <action application="conference" data="$1@default >>>>>>>>>>>> "/> >>>>>>>>>>>> </condition> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>> >>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>> >>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>> >>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>> <http://freeswitch.org/g+>* >>>>>>>> >>>>>>>> ClueCon Weekly Development Call >>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>> >>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>> http://twitter.com/FreeSWITCH >>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>> <http://freeswitch.org/g+>* >>>>>> >>>>>> ClueCon Weekly Development Call >>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>> >>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>> http://twitter.com/FreeSWITCH >>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>> <http://freeswitch.org/g+>* >>>> >>>> ClueCon Weekly Development Call >>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel? On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening.
On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
<extension name="8765">
<condition field="destination_number" expression="^8765$">
<action application="set" data="answer_delay=3000"/>
<action application="answer"/>
<action application="playback" data= "tone_stream://%(1000,0,600)" />
<action application="conference" data="$1@default"/>
</condition>
</extension>
On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
could you send me the XML for the extension you used to set up this example?
On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hey Anthony,
just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Also it's using sipjs.com not jssip. On Jun 8, 2014 8:14 AM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
> Debian 7 on proxmox vm. > On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> > wrote: > >> Anthony, >> >> Another update - I thought that the only thing my original >> installation and the Debian VM had in common was that both were VMs. So I >> went ahead and installed FreeSWITCH on my local Windows machine. Same >> problem is happening here as well. >> >> I am extremely curious what's different between your installation >> and my three setups (CentOS, Debian and Windows) that accounts for the >> difference in this behavior. Could you tell me what machine/OS you are >> using? >> >> >> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com >> > wrote: >> >>> Hi Anthony, >>> >>> I dug further and I can reproduce the issue on a completely >>> different VM (Debian on Windows host). I installed the latest FS master >>> and even switched to the vanilla conf to eliminate the possibility of my >>> config causing this issue. >>> >>> It's still happening. If you have time to look into this some >>> more, I'll be happy to upload my whole VM for someone to take a look at. >>> >>> >>> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> Yep it's the same. >>>> >>>> Thank you >>>> Oleg >>>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>>> anthony.minessale@gmail.com> wrote: >>>> >>>>> What if you play an audio file? is it the same? >>>>> I can only guess really. >>>>> >>>>> >>>>> >>>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>>> olegstolyar@gmail.com> wrote: >>>>> >>>>>> I tried this: >>>>>> <action application="set" data="timer_name=soft"/> >>>>>> <action application="playback" >>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>> >>>>>> Didn't help unfortunately (unless this is the wrong syntax for >>>>>> setting the variable). Completely agreed on the AWS instances - in fact we >>>>>> started on an older m1 instance and the voice quality was pretty bad and >>>>>> there were frequent delays. Then we switch to an m3 and it was much better. >>>>>> >>>>>> Any advice on setting enter-sound per member? Or setting a >>>>>> no-enter-sound flag - similar to the nomoh flag? >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>>> anthony.minessale@gmail.com> wrote: >>>>>> >>>>>>> Try setting the channel var timer_name=soft before playing the >>>>>>> beep. It may be jittery media flow causing your problem. >>>>>>> My only advice on amazon is use the more expensive one because >>>>>>> the lower cost ones are not reliable for cpu time. >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> OK, thanks. It's still happening with the latest master, so >>>>>>>> I have to assume that this is something to do with the AWS VMs. It would >>>>>>>> not be the first time strange things happen there. >>>>>>>> >>>>>>>> I was trying to work around the problem and noticed that if >>>>>>>> instead of playing the beep using playback app before placing the user into >>>>>>>> the conference, I make the beep the enter-sound of the conference, the >>>>>>>> problem does not happen. >>>>>>>> >>>>>>>> However, I need to only play this sound for users joining the >>>>>>>> conference from one dialplan but not from another. I cannot find a way to >>>>>>>> do it. I looked at >>>>>>>> <action application="set" >>>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>>> but it changes the enter sound for the conference for >>>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>>> same sound. >>>>>>>> >>>>>>>> Is there a way to assign conference params per user (like >>>>>>>> member flags)? >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>> >>>>>>>>> If you opt to use master, ALWAYS use latest master since >>>>>>>>> that is the nature of master. I have no idea if your problem is related to >>>>>>>>> those things but the server I pointed you to test is running basically >>>>>>>>> latest master. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Thanks Anthony! >>>>>>>>>> >>>>>>>>>> I tried adding this var and also tried uncommenting the >>>>>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>>>>> it's not a problem with the client side. >>>>>>>>>> >>>>>>>>>> Do I need to upgrade to the latest master? I am using >>>>>>>>>> master from 5/22 version 1.5.13b >>>>>>>>>> >>>>>>>>>> Would it help if gave you my test ws server and sip address >>>>>>>>>> so you could reproduce this? >>>>>>>>>> >>>>>>>>>> Is it possible that the problem is with my AWS instance or >>>>>>>>>> CentOS 5.9? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I set up your test ext here: >>>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>>> >>>>>>>>>>> And it seems to work. >>>>>>>>>>> >>>>>>>>>>> Try this in vars.xml >>>>>>>>>>> >>>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>>> >>>>>>>>>>> I don't think the silence suppression works very well in >>>>>>>>>>> the browser. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> In case it was not clear from the previous email - this >>>>>>>>>>>> only happens with conferences. If after the beep, instead of placing the >>>>>>>>>>>> call into a conference, I just play some music, there are no problems. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>> >>>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a >>>>>>>>>>>>> user calls in, I play a beep and put them into a conference. Two strange >>>>>>>>>>>>> things happen after that: >>>>>>>>>>>>> >>>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>>> >>>>>>>>>>>>> 2. About 3 seconds into the call, there is another very >>>>>>>>>>>>> short beep - like a small portion of the original beep that the user hears. >>>>>>>>>>>>> >>>>>>>>>>>>> This absolutely does not happen with softphones - only >>>>>>>>>>>>> with WebRTC. I tried it with Chrome and Opera - same thing is happening on >>>>>>>>>>>>> both. >>>>>>>>>>>>> >>>>>>>>>>>>> I tried increasing the answer delay from 2 seconds to 3 >>>>>>>>>>>>> and the noise now happens more rarely but the extra beep is still there >>>>>>>>>>>>> almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>>> >>>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>>> >>>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>>> >>>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>>> <action application="playback" >>>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>>> <action application="conference" data="$1@default >>>>>>>>>>>>> "/> >>>>>>>>>>>>> </condition> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>> Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> -- >>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>> >>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>> >>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>> >>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>> >>>>>>>>> ClueCon Weekly Development Call >>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>> >>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>> http://twitter.com/FreeSWITCH >>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>> <http://freeswitch.org/g+>* >>>>>>> >>>>>>> ClueCon Weekly Development Call >>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>> >>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>> http://twitter.com/FreeSWITCH >>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>> <http://freeswitch.org/g+>* >>>>> >>>>> ClueCon Weekly Development Call >>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation. You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-) On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel?
On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening.
On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
<extension name="8765">
<condition field="destination_number" expression="^8765$">
<action application="set" data="answer_delay=3000"/>
<action application="answer"/>
<action application="playback" data= "tone_stream://%(1000,0,600)" />
<action application="conference" data="$1@default"/>
</condition>
</extension>
On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
could you send me the XML for the extension you used to set up this example?
On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hey Anthony,
just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
> Also it's using sipjs.com not jssip. > On Jun 8, 2014 8:14 AM, "Anthony Minessale" < > anthony.minessale@gmail.com> wrote: > >> Debian 7 on proxmox vm. >> On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> >> wrote: >> >>> Anthony, >>> >>> Another update - I thought that the only thing my original >>> installation and the Debian VM had in common was that both were VMs. So I >>> went ahead and installed FreeSWITCH on my local Windows machine. Same >>> problem is happening here as well. >>> >>> I am extremely curious what's different between your installation >>> and my three setups (CentOS, Debian and Windows) that accounts for the >>> difference in this behavior. Could you tell me what machine/OS you are >>> using? >>> >>> >>> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> Hi Anthony, >>>> >>>> I dug further and I can reproduce the issue on a completely >>>> different VM (Debian on Windows host). I installed the latest FS master >>>> and even switched to the vanilla conf to eliminate the possibility of my >>>> config causing this issue. >>>> >>>> It's still happening. If you have time to look into this some >>>> more, I'll be happy to upload my whole VM for someone to take a look at. >>>> >>>> >>>> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> Yep it's the same. >>>>> >>>>> Thank you >>>>> Oleg >>>>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>>>> anthony.minessale@gmail.com> wrote: >>>>> >>>>>> What if you play an audio file? is it the same? >>>>>> I can only guess really. >>>>>> >>>>>> >>>>>> >>>>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>>>> olegstolyar@gmail.com> wrote: >>>>>> >>>>>>> I tried this: >>>>>>> <action application="set" data="timer_name=soft"/> >>>>>>> <action application="playback" >>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>> >>>>>>> Didn't help unfortunately (unless this is the wrong syntax for >>>>>>> setting the variable). Completely agreed on the AWS instances - in fact we >>>>>>> started on an older m1 instance and the voice quality was pretty bad and >>>>>>> there were frequent delays. Then we switch to an m3 and it was much better. >>>>>>> >>>>>>> Any advice on setting enter-sound per member? Or setting a >>>>>>> no-enter-sound flag - similar to the nomoh flag? >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>> >>>>>>>> Try setting the channel var timer_name=soft before playing >>>>>>>> the beep. It may be jittery media flow causing your problem. >>>>>>>> My only advice on amazon is use the more expensive one >>>>>>>> because the lower cost ones are not reliable for cpu time. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>> >>>>>>>>> OK, thanks. It's still happening with the latest master, so >>>>>>>>> I have to assume that this is something to do with the AWS VMs. It would >>>>>>>>> not be the first time strange things happen there. >>>>>>>>> >>>>>>>>> I was trying to work around the problem and noticed that if >>>>>>>>> instead of playing the beep using playback app before placing the user into >>>>>>>>> the conference, I make the beep the enter-sound of the conference, the >>>>>>>>> problem does not happen. >>>>>>>>> >>>>>>>>> However, I need to only play this sound for users joining >>>>>>>>> the conference from one dialplan but not from another. I cannot find a way >>>>>>>>> to do it. I looked at >>>>>>>>> <action application="set" >>>>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>>>> but it changes the enter sound for the conference for >>>>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>>>> same sound. >>>>>>>>> >>>>>>>>> Is there a way to assign conference params per user (like >>>>>>>>> member flags)? >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> If you opt to use master, ALWAYS use latest master since >>>>>>>>>> that is the nature of master. I have no idea if your problem is related to >>>>>>>>>> those things but the server I pointed you to test is running basically >>>>>>>>>> latest master. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Thanks Anthony! >>>>>>>>>>> >>>>>>>>>>> I tried adding this var and also tried uncommenting the >>>>>>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>>>>>> it's not a problem with the client side. >>>>>>>>>>> >>>>>>>>>>> Do I need to upgrade to the latest master? I am using >>>>>>>>>>> master from 5/22 version 1.5.13b >>>>>>>>>>> >>>>>>>>>>> Would it help if gave you my test ws server and sip >>>>>>>>>>> address so you could reproduce this? >>>>>>>>>>> >>>>>>>>>>> Is it possible that the problem is with my AWS instance or >>>>>>>>>>> CentOS 5.9? >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> I set up your test ext here: >>>>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>>>> >>>>>>>>>>>> And it seems to work. >>>>>>>>>>>> >>>>>>>>>>>> Try this in vars.xml >>>>>>>>>>>> >>>>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>>>> >>>>>>>>>>>> I don't think the silence suppression works very well in >>>>>>>>>>>> the browser. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> In case it was not clear from the previous email - this >>>>>>>>>>>>> only happens with conferences. If after the beep, instead of placing the >>>>>>>>>>>>> call into a conference, I just play some music, there are no problems. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>> >>>>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a >>>>>>>>>>>>>> user calls in, I play a beep and put them into a conference. Two strange >>>>>>>>>>>>>> things happen after that: >>>>>>>>>>>>>> >>>>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>>>> >>>>>>>>>>>>>> 2. About 3 seconds into the call, there is another very >>>>>>>>>>>>>> short beep - like a small portion of the original beep that the user hears. >>>>>>>>>>>>>> >>>>>>>>>>>>>> This absolutely does not happen with softphones - only >>>>>>>>>>>>>> with WebRTC. I tried it with Chrome and Opera - same thing is happening on >>>>>>>>>>>>>> both. >>>>>>>>>>>>>> >>>>>>>>>>>>>> I tried increasing the answer delay from 2 seconds to 3 >>>>>>>>>>>>>> and the noise now happens more rarely but the extra beep is still there >>>>>>>>>>>>>> almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>>>> >>>>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>>>> <action application="playback" >>>>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>>>> <action application="conference" >>>>>>>>>>>>>> data="$1@default"/> >>>>>>>>>>>>>> </condition> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>> Server >>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>> >>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>> >>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>> Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>> >>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>> >>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>> >>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>> <http://freeswitch.org/g+>* >>>>>>>> >>>>>>>> ClueCon Weekly Development Call >>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>> >>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>> http://twitter.com/FreeSWITCH >>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>> <http://freeswitch.org/g+>* >>>>>> >>>>>> ClueCon Weekly Development Call >>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You can email consulting@freeswitch.org and see if we can find a way. On Thu, Jun 12, 2014 at 1:28 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation.
You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-)
On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel?
On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening.
On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
<extension name="8765">
<condition field="destination_number" expression="^8765$">
<action application="set" data="answer_delay=3000"/>
<action application="answer"/>
<action application="playback" data= "tone_stream://%(1000,0,600)" />
<action application="conference" data="$1@default"/>
</condition>
</extension>
On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
could you send me the XML for the extension you used to set up this example?
On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hey Anthony,
just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your > example and there was no noise, so it's not the client. I also am setting > my stun servers array to empty to speed up the connection but that also > does not seem to matter. I'll keep digging. > > > On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < > anthony.minessale@gmail.com> wrote: > >> Also it's using sipjs.com not jssip. >> On Jun 8, 2014 8:14 AM, "Anthony Minessale" < >> anthony.minessale@gmail.com> wrote: >> >>> Debian 7 on proxmox vm. >>> On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> >>> wrote: >>> >>>> Anthony, >>>> >>>> Another update - I thought that the only thing my original >>>> installation and the Debian VM had in common was that both were VMs. So I >>>> went ahead and installed FreeSWITCH on my local Windows machine. Same >>>> problem is happening here as well. >>>> >>>> I am extremely curious what's different between your installation >>>> and my three setups (CentOS, Debian and Windows) that accounts for the >>>> difference in this behavior. Could you tell me what machine/OS you are >>>> using? >>>> >>>> >>>> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> Hi Anthony, >>>>> >>>>> I dug further and I can reproduce the issue on a completely >>>>> different VM (Debian on Windows host). I installed the latest FS master >>>>> and even switched to the vanilla conf to eliminate the possibility of my >>>>> config causing this issue. >>>>> >>>>> It's still happening. If you have time to look into this some >>>>> more, I'll be happy to upload my whole VM for someone to take a look at. >>>>> >>>>> >>>>> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar < >>>>> olegstolyar@gmail.com> wrote: >>>>> >>>>>> Yep it's the same. >>>>>> >>>>>> Thank you >>>>>> Oleg >>>>>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>>>>> anthony.minessale@gmail.com> wrote: >>>>>> >>>>>>> What if you play an audio file? is it the same? >>>>>>> I can only guess really. >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> I tried this: >>>>>>>> <action application="set" data="timer_name=soft"/> >>>>>>>> <action application="playback" >>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>> >>>>>>>> Didn't help unfortunately (unless this is the wrong syntax >>>>>>>> for setting the variable). Completely agreed on the AWS instances - in >>>>>>>> fact we started on an older m1 instance and the voice quality was pretty >>>>>>>> bad and there were frequent delays. Then we switch to an m3 and it was >>>>>>>> much better. >>>>>>>> >>>>>>>> Any advice on setting enter-sound per member? Or setting a >>>>>>>> no-enter-sound flag - similar to the nomoh flag? >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>> >>>>>>>>> Try setting the channel var timer_name=soft before playing >>>>>>>>> the beep. It may be jittery media flow causing your problem. >>>>>>>>> My only advice on amazon is use the more expensive one >>>>>>>>> because the lower cost ones are not reliable for cpu time. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> OK, thanks. It's still happening with the latest master, >>>>>>>>>> so I have to assume that this is something to do with the AWS VMs. It >>>>>>>>>> would not be the first time strange things happen there. >>>>>>>>>> >>>>>>>>>> I was trying to work around the problem and noticed that if >>>>>>>>>> instead of playing the beep using playback app before placing the user into >>>>>>>>>> the conference, I make the beep the enter-sound of the conference, the >>>>>>>>>> problem does not happen. >>>>>>>>>> >>>>>>>>>> However, I need to only play this sound for users joining >>>>>>>>>> the conference from one dialplan but not from another. I cannot find a way >>>>>>>>>> to do it. I looked at >>>>>>>>>> <action application="set" >>>>>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>>>>> but it changes the enter sound for the conference for >>>>>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>>>>> same sound. >>>>>>>>>> >>>>>>>>>> Is there a way to assign conference params per user (like >>>>>>>>>> member flags)? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> If you opt to use master, ALWAYS use latest master since >>>>>>>>>>> that is the nature of master. I have no idea if your problem is related to >>>>>>>>>>> those things but the server I pointed you to test is running basically >>>>>>>>>>> latest master. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Thanks Anthony! >>>>>>>>>>>> >>>>>>>>>>>> I tried adding this var and also tried uncommenting the >>>>>>>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>>>>>>> it's not a problem with the client side. >>>>>>>>>>>> >>>>>>>>>>>> Do I need to upgrade to the latest master? I am using >>>>>>>>>>>> master from 5/22 version 1.5.13b >>>>>>>>>>>> >>>>>>>>>>>> Would it help if gave you my test ws server and sip >>>>>>>>>>>> address so you could reproduce this? >>>>>>>>>>>> >>>>>>>>>>>> Is it possible that the problem is with my AWS instance >>>>>>>>>>>> or CentOS 5.9? >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> I set up your test ext here: >>>>>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>>>>> >>>>>>>>>>>>> And it seems to work. >>>>>>>>>>>>> >>>>>>>>>>>>> Try this in vars.xml >>>>>>>>>>>>> >>>>>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>>>>> >>>>>>>>>>>>> I don't think the silence suppression works very well in >>>>>>>>>>>>> the browser. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> In case it was not clear from the previous email - this >>>>>>>>>>>>>> only happens with conferences. If after the beep, instead of placing the >>>>>>>>>>>>>> call into a conference, I just play some music, there are no problems. >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after a >>>>>>>>>>>>>>> user calls in, I play a beep and put them into a conference. Two strange >>>>>>>>>>>>>>> things happen after that: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> 2. About 3 seconds into the call, there is another >>>>>>>>>>>>>>> very short beep - like a small portion of the original beep that the user >>>>>>>>>>>>>>> hears. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> This absolutely does not happen with softphones - only >>>>>>>>>>>>>>> with WebRTC. I tried it with Chrome and Opera - same thing is happening on >>>>>>>>>>>>>>> both. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I tried increasing the answer delay from 2 seconds to >>>>>>>>>>>>>>> 3 and the noise now happens more rarely but the extra beep is still there >>>>>>>>>>>>>>> almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>>>>> <action application="playback" >>>>>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>>>>> <action application="conference" >>>>>>>>>>>>>>> data="$1@default"/> >>>>>>>>>>>>>>> </condition> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>> Server >>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> -- >>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>>> >>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>> >>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>> Server >>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>> Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> -- >>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>> >>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>> >>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>> >>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>> >>>>>>>>> ClueCon Weekly Development Call >>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>> >>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>> http://twitter.com/FreeSWITCH >>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>> <http://freeswitch.org/g+>* >>>>>>> >>>>>>> ClueCon Weekly Development Call >>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Thanks Anthony, I did that. No response yet. However, I did get a VM with Debian 7, and indeed this problem does not happen there. I am not sure where to go from here. I am kind of stuck with CentOS as my production system at least for the next several months. On Thu, Jun 12, 2014 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You can email consulting@freeswitch.org and see if we can find a way.
On Thu, Jun 12, 2014 at 1:28 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation.
You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-)
On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel?
On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening.
On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
<extension name="8765">
<condition field="destination_number" expression="^8765$">
<action application="set" data="answer_delay=3000"/>
<action application="answer"/>
<action application="playback" data= "tone_stream://%(1000,0,600)" />
<action application="conference" data="$1@default"/>
</condition>
</extension>
On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Hi Anthony,
could you send me the XML for the extension you used to set up this example?
On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
> Hey Anthony, > > just in case I am missing something obvious, could you send me the > extension in the dialplan you set up with my example? > > > On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > >> Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your >> example and there was no noise, so it's not the client. I also am setting >> my stun servers array to empty to speed up the connection but that also >> does not seem to matter. I'll keep digging. >> >> >> On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < >> anthony.minessale@gmail.com> wrote: >> >>> Also it's using sipjs.com not jssip. >>> On Jun 8, 2014 8:14 AM, "Anthony Minessale" < >>> anthony.minessale@gmail.com> wrote: >>> >>>> Debian 7 on proxmox vm. >>>> On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> >>>> wrote: >>>> >>>>> Anthony, >>>>> >>>>> Another update - I thought that the only thing my original >>>>> installation and the Debian VM had in common was that both were VMs. So I >>>>> went ahead and installed FreeSWITCH on my local Windows machine. Same >>>>> problem is happening here as well. >>>>> >>>>> I am extremely curious what's different between your >>>>> installation and my three setups (CentOS, Debian and Windows) that accounts >>>>> for the difference in this behavior. Could you tell me what machine/OS you >>>>> are using? >>>>> >>>>> >>>>> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar < >>>>> olegstolyar@gmail.com> wrote: >>>>> >>>>>> Hi Anthony, >>>>>> >>>>>> I dug further and I can reproduce the issue on a completely >>>>>> different VM (Debian on Windows host). I installed the latest FS master >>>>>> and even switched to the vanilla conf to eliminate the possibility of my >>>>>> config causing this issue. >>>>>> >>>>>> It's still happening. If you have time to look into this some >>>>>> more, I'll be happy to upload my whole VM for someone to take a look at. >>>>>> >>>>>> >>>>>> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar < >>>>>> olegstolyar@gmail.com> wrote: >>>>>> >>>>>>> Yep it's the same. >>>>>>> >>>>>>> Thank you >>>>>>> Oleg >>>>>>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>> >>>>>>>> What if you play an audio file? is it the same? >>>>>>>> I can only guess really. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>> >>>>>>>>> I tried this: >>>>>>>>> <action application="set" data="timer_name=soft"/> >>>>>>>>> <action application="playback" >>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>> >>>>>>>>> Didn't help unfortunately (unless this is the wrong syntax >>>>>>>>> for setting the variable). Completely agreed on the AWS instances - in >>>>>>>>> fact we started on an older m1 instance and the voice quality was pretty >>>>>>>>> bad and there were frequent delays. Then we switch to an m3 and it was >>>>>>>>> much better. >>>>>>>>> >>>>>>>>> Any advice on setting enter-sound per member? Or setting a >>>>>>>>> no-enter-sound flag - similar to the nomoh flag? >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Try setting the channel var timer_name=soft before playing >>>>>>>>>> the beep. It may be jittery media flow causing your problem. >>>>>>>>>> My only advice on amazon is use the more expensive one >>>>>>>>>> because the lower cost ones are not reliable for cpu time. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> OK, thanks. It's still happening with the latest master, >>>>>>>>>>> so I have to assume that this is something to do with the AWS VMs. It >>>>>>>>>>> would not be the first time strange things happen there. >>>>>>>>>>> >>>>>>>>>>> I was trying to work around the problem and noticed that >>>>>>>>>>> if instead of playing the beep using playback app before placing the user >>>>>>>>>>> into the conference, I make the beep the enter-sound of the conference, the >>>>>>>>>>> problem does not happen. >>>>>>>>>>> >>>>>>>>>>> However, I need to only play this sound for users joining >>>>>>>>>>> the conference from one dialplan but not from another. I cannot find a way >>>>>>>>>>> to do it. I looked at >>>>>>>>>>> <action application="set" >>>>>>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>>>>>> but it changes the enter sound for the conference for >>>>>>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>>>>>> same sound. >>>>>>>>>>> >>>>>>>>>>> Is there a way to assign conference params per user (like >>>>>>>>>>> member flags)? >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> If you opt to use master, ALWAYS use latest master since >>>>>>>>>>>> that is the nature of master. I have no idea if your problem is related to >>>>>>>>>>>> those things but the server I pointed you to test is running basically >>>>>>>>>>>> latest master. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Thanks Anthony! >>>>>>>>>>>>> >>>>>>>>>>>>> I tried adding this var and also tried uncommenting the >>>>>>>>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>>>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>>>>>>>> it's not a problem with the client side. >>>>>>>>>>>>> >>>>>>>>>>>>> Do I need to upgrade to the latest master? I am using >>>>>>>>>>>>> master from 5/22 version 1.5.13b >>>>>>>>>>>>> >>>>>>>>>>>>> Would it help if gave you my test ws server and sip >>>>>>>>>>>>> address so you could reproduce this? >>>>>>>>>>>>> >>>>>>>>>>>>> Is it possible that the problem is with my AWS instance >>>>>>>>>>>>> or CentOS 5.9? >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> I set up your test ext here: >>>>>>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>>>>>> >>>>>>>>>>>>>> And it seems to work. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Try this in vars.xml >>>>>>>>>>>>>> >>>>>>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>>>>>> >>>>>>>>>>>>>> I don't think the silence suppression works very well >>>>>>>>>>>>>> in the browser. >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> In case it was not clear from the previous email - >>>>>>>>>>>>>>> this only happens with conferences. If after the beep, instead of placing >>>>>>>>>>>>>>> the call into a conference, I just play some music, there are no problems. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after >>>>>>>>>>>>>>>> a user calls in, I play a beep and put them into a conference. Two strange >>>>>>>>>>>>>>>> things happen after that: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> 2. About 3 seconds into the call, there is another >>>>>>>>>>>>>>>> very short beep - like a small portion of the original beep that the user >>>>>>>>>>>>>>>> hears. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> This absolutely does not happen with softphones - >>>>>>>>>>>>>>>> only with WebRTC. I tried it with Chrome and Opera - same thing is >>>>>>>>>>>>>>>> happening on both. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> I tried increasing the answer delay from 2 seconds to >>>>>>>>>>>>>>>> 3 and the noise now happens more rarely but the extra beep is still there >>>>>>>>>>>>>>>> almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>>>>>> <action application="playback" >>>>>>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>>>>>> <action application="conference" >>>>>>>>>>>>>>>> data="$1@default"/> >>>>>>>>>>>>>>>> </condition> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> -- >>>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>>>> >>>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>>> >>>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>> Server >>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>> Server >>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>> >>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>> >>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>> Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>> >>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>> >>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>> >>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>> <http://freeswitch.org/g+>* >>>>>>>> >>>>>>>> ClueCon Weekly Development Call >>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
My only suggestion is check the version of openssl? On Sat, Jun 14, 2014 at 12:26 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony, I did that. No response yet.
However, I did get a VM with Debian 7, and indeed this problem does not happen there.
I am not sure where to go from here. I am kind of stuck with CentOS as my production system at least for the next several months.
On Thu, Jun 12, 2014 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You can email consulting@freeswitch.org and see if we can find a way.
On Thu, Jun 12, 2014 at 1:28 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation.
You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-)
On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel?
On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening.
On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
<extension name="8765">
<condition field="destination_number" expression="^8765$">
<action application="set" data="answer_delay=3000"/>
<action application="answer"/>
<action application="playback" data= "tone_stream://%(1000,0,600)" />
<action application="conference" data="$1@default"/>
</condition>
</extension>
On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com > wrote:
> Hi Anthony, > > could you send me the XML for the extension you used to set up this > example? > > > > On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > >> Hey Anthony, >> >> just in case I am missing something obvious, could you send me the >> extension in the dialplan you set up with my example? >> >> >> On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com >> > wrote: >> >>> Yeah, I saw sipjs vs jssp but I pointed my own jssip client to >>> your example and there was no noise, so it's not the client. I also am >>> setting my stun servers array to empty to speed up the connection but that >>> also does not seem to matter. I'll keep digging. >>> >>> >>> On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < >>> anthony.minessale@gmail.com> wrote: >>> >>>> Also it's using sipjs.com not jssip. >>>> On Jun 8, 2014 8:14 AM, "Anthony Minessale" < >>>> anthony.minessale@gmail.com> wrote: >>>> >>>>> Debian 7 on proxmox vm. >>>>> On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> >>>>> wrote: >>>>> >>>>>> Anthony, >>>>>> >>>>>> Another update - I thought that the only thing my original >>>>>> installation and the Debian VM had in common was that both were VMs. So I >>>>>> went ahead and installed FreeSWITCH on my local Windows machine. Same >>>>>> problem is happening here as well. >>>>>> >>>>>> I am extremely curious what's different between your >>>>>> installation and my three setups (CentOS, Debian and Windows) that accounts >>>>>> for the difference in this behavior. Could you tell me what machine/OS you >>>>>> are using? >>>>>> >>>>>> >>>>>> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar < >>>>>> olegstolyar@gmail.com> wrote: >>>>>> >>>>>>> Hi Anthony, >>>>>>> >>>>>>> I dug further and I can reproduce the issue on a completely >>>>>>> different VM (Debian on Windows host). I installed the latest FS master >>>>>>> and even switched to the vanilla conf to eliminate the possibility of my >>>>>>> config causing this issue. >>>>>>> >>>>>>> It's still happening. If you have time to look into this some >>>>>>> more, I'll be happy to upload my whole VM for someone to take a look at. >>>>>>> >>>>>>> >>>>>>> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> Yep it's the same. >>>>>>>> >>>>>>>> Thank you >>>>>>>> Oleg >>>>>>>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>> >>>>>>>>> What if you play an audio file? is it the same? >>>>>>>>> I can only guess really. >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> I tried this: >>>>>>>>>> <action application="set" data="timer_name=soft"/> >>>>>>>>>> <action application="playback" >>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>> >>>>>>>>>> Didn't help unfortunately (unless this is the wrong syntax >>>>>>>>>> for setting the variable). Completely agreed on the AWS instances - in >>>>>>>>>> fact we started on an older m1 instance and the voice quality was pretty >>>>>>>>>> bad and there were frequent delays. Then we switch to an m3 and it was >>>>>>>>>> much better. >>>>>>>>>> >>>>>>>>>> Any advice on setting enter-sound per member? Or setting a >>>>>>>>>> no-enter-sound flag - similar to the nomoh flag? >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> Try setting the channel var timer_name=soft before playing >>>>>>>>>>> the beep. It may be jittery media flow causing your problem. >>>>>>>>>>> My only advice on amazon is use the more expensive one >>>>>>>>>>> because the lower cost ones are not reliable for cpu time. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> OK, thanks. It's still happening with the latest master, >>>>>>>>>>>> so I have to assume that this is something to do with the AWS VMs. It >>>>>>>>>>>> would not be the first time strange things happen there. >>>>>>>>>>>> >>>>>>>>>>>> I was trying to work around the problem and noticed that >>>>>>>>>>>> if instead of playing the beep using playback app before placing the user >>>>>>>>>>>> into the conference, I make the beep the enter-sound of the conference, the >>>>>>>>>>>> problem does not happen. >>>>>>>>>>>> >>>>>>>>>>>> However, I need to only play this sound for users joining >>>>>>>>>>>> the conference from one dialplan but not from another. I cannot find a way >>>>>>>>>>>> to do it. I looked at >>>>>>>>>>>> <action application="set" >>>>>>>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>>>>>>> but it changes the enter sound for the conference for >>>>>>>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>>>>>>> same sound. >>>>>>>>>>>> >>>>>>>>>>>> Is there a way to assign conference params per user (like >>>>>>>>>>>> member flags)? >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> If you opt to use master, ALWAYS use latest master since >>>>>>>>>>>>> that is the nature of master. I have no idea if your problem is related to >>>>>>>>>>>>> those things but the server I pointed you to test is running basically >>>>>>>>>>>>> latest master. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> Thanks Anthony! >>>>>>>>>>>>>> >>>>>>>>>>>>>> I tried adding this var and also tried uncommenting the >>>>>>>>>>>>>> suppress_cng line in my sip profile. It did not seem to make a difference. >>>>>>>>>>>>>> I pointed my client code to your sample and it worked fine - no noise, so >>>>>>>>>>>>>> it's not a problem with the client side. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Do I need to upgrade to the latest master? I am using >>>>>>>>>>>>>> master from 5/22 version 1.5.13b >>>>>>>>>>>>>> >>>>>>>>>>>>>> Would it help if gave you my test ws server and sip >>>>>>>>>>>>>> address so you could reproduce this? >>>>>>>>>>>>>> >>>>>>>>>>>>>> Is it possible that the problem is with my AWS instance >>>>>>>>>>>>>> or CentOS 5.9? >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> I set up your test ext here: >>>>>>>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> And it seems to work. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Try this in vars.xml >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I don't think the silence suppression works very well >>>>>>>>>>>>>>> in the browser. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> In case it was not clear from the previous email - >>>>>>>>>>>>>>>> this only happens with conferences. If after the beep, instead of placing >>>>>>>>>>>>>>>> the call into a conference, I just play some music, there are no problems. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and after >>>>>>>>>>>>>>>>> a user calls in, I play a beep and put them into a conference. Two strange >>>>>>>>>>>>>>>>> things happen after that: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> 2. About 3 seconds into the call, there is another >>>>>>>>>>>>>>>>> very short beep - like a small portion of the original beep that the user >>>>>>>>>>>>>>>>> hears. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> This absolutely does not happen with softphones - >>>>>>>>>>>>>>>>> only with WebRTC. I tried it with Chrome and Opera - same thing is >>>>>>>>>>>>>>>>> happening on both. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I tried increasing the answer delay from 2 seconds >>>>>>>>>>>>>>>>> to 3 and the noise now happens more rarely but the extra beep is still >>>>>>>>>>>>>>>>> there almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>>>>>>> <action application="playback" >>>>>>>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>>>>>>> <action application="conference" >>>>>>>>>>>>>>>>> data="$1@default"/> >>>>>>>>>>>>>>>>> </condition> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>> Server >>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> -- >>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>>> >>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>> >>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>> Server >>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>> Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> -- >>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>> >>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>> >>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> -- >>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>> >>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>> >>>>>>>>> ClueCon Weekly Development Call >>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> _________________________________________________________________________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting@freeswitch.org >>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>> >>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>> http://www.cudatel.com >>>>>>>>> >>>>>>>>> Official FreeSWITCH Sites >>>>>>>>> http://www.freeswitch.org >>>>>>>>> http://wiki.freeswitch.org >>>>>>>>> http://www.cluecon.com >>>>>>>>> >>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE: >>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
They all have different OpenSSL versions but they all have the latest available for them. I tried building FreeSWITCH on my CentOS 5 with the latest OpenSSL using these instructions: http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_v.1.4.2Fmaster... I had to fudge it a little to make it work with Brian's Makefile. The problem is still there but I am not sure switching to the new OpenSSL worked. The active version of OpenSSL on the OS is still old. Is there a way to check which OpenSSL version FreeSWITCH is using? On Sat, Jun 14, 2014 at 2:15 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
My only suggestion is check the version of openssl?
On Sat, Jun 14, 2014 at 12:26 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony, I did that. No response yet.
However, I did get a VM with Debian 7, and indeed this problem does not happen there.
I am not sure where to go from here. I am kind of stuck with CentOS as my production system at least for the next several months.
On Thu, Jun 12, 2014 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You can email consulting@freeswitch.org and see if we can find a way.
On Thu, Jun 12, 2014 at 1:28 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation.
You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-)
On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel?
On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening.
On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
> <extension name="8765"> > > > > <condition field="destination_number" expression="^8765$"> > > > > <action application="set" data="answer_delay=3000"/> > > > > <action application="answer"/> > > > > <action application="playback" data= > "tone_stream://%(1000,0,600)" /> > > > <action application="conference" data="$1@default"/> > > > > </condition> > > > > </extension> > > > On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar < > olegstolyar@gmail.com> wrote: > >> Hi Anthony, >> >> could you send me the XML for the extension you used to set up this >> example? >> >> >> >> On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com >> > wrote: >> >>> Hey Anthony, >>> >>> just in case I am missing something obvious, could you send me the >>> extension in the dialplan you set up with my example? >>> >>> >>> On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> Yeah, I saw sipjs vs jssp but I pointed my own jssip client to >>>> your example and there was no noise, so it's not the client. I also am >>>> setting my stun servers array to empty to speed up the connection but that >>>> also does not seem to matter. I'll keep digging. >>>> >>>> >>>> On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < >>>> anthony.minessale@gmail.com> wrote: >>>> >>>>> Also it's using sipjs.com not jssip. >>>>> On Jun 8, 2014 8:14 AM, "Anthony Minessale" < >>>>> anthony.minessale@gmail.com> wrote: >>>>> >>>>>> Debian 7 on proxmox vm. >>>>>> On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> >>>>>> wrote: >>>>>> >>>>>>> Anthony, >>>>>>> >>>>>>> Another update - I thought that the only thing my original >>>>>>> installation and the Debian VM had in common was that both were VMs. So I >>>>>>> went ahead and installed FreeSWITCH on my local Windows machine. Same >>>>>>> problem is happening here as well. >>>>>>> >>>>>>> I am extremely curious what's different between your >>>>>>> installation and my three setups (CentOS, Debian and Windows) that accounts >>>>>>> for the difference in this behavior. Could you tell me what machine/OS you >>>>>>> are using? >>>>>>> >>>>>>> >>>>>>> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> Hi Anthony, >>>>>>>> >>>>>>>> I dug further and I can reproduce the issue on a completely >>>>>>>> different VM (Debian on Windows host). I installed the latest FS master >>>>>>>> and even switched to the vanilla conf to eliminate the possibility of my >>>>>>>> config causing this issue. >>>>>>>> >>>>>>>> It's still happening. If you have time to look into this >>>>>>>> some more, I'll be happy to upload my whole VM for someone to take a look >>>>>>>> at. >>>>>>>> >>>>>>>> >>>>>>>> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar < >>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>> >>>>>>>>> Yep it's the same. >>>>>>>>> >>>>>>>>> Thank you >>>>>>>>> Oleg >>>>>>>>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> What if you play an audio file? is it the same? >>>>>>>>>> I can only guess really. >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I tried this: >>>>>>>>>>> <action application="set" data="timer_name=soft"/> >>>>>>>>>>> <action application="playback" >>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>> >>>>>>>>>>> Didn't help unfortunately (unless this is the wrong syntax >>>>>>>>>>> for setting the variable). Completely agreed on the AWS instances - in >>>>>>>>>>> fact we started on an older m1 instance and the voice quality was pretty >>>>>>>>>>> bad and there were frequent delays. Then we switch to an m3 and it was >>>>>>>>>>> much better. >>>>>>>>>>> >>>>>>>>>>> Any advice on setting enter-sound per member? Or setting >>>>>>>>>>> a no-enter-sound flag - similar to the nomoh flag? >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> Try setting the channel var timer_name=soft before >>>>>>>>>>>> playing the beep. It may be jittery media flow causing your problem. >>>>>>>>>>>> My only advice on amazon is use the more expensive one >>>>>>>>>>>> because the lower cost ones are not reliable for cpu time. >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> OK, thanks. It's still happening with the latest >>>>>>>>>>>>> master, so I have to assume that this is something to do with the AWS VMs. >>>>>>>>>>>>> It would not be the first time strange things happen there. >>>>>>>>>>>>> >>>>>>>>>>>>> I was trying to work around the problem and noticed that >>>>>>>>>>>>> if instead of playing the beep using playback app before placing the user >>>>>>>>>>>>> into the conference, I make the beep the enter-sound of the conference, the >>>>>>>>>>>>> problem does not happen. >>>>>>>>>>>>> >>>>>>>>>>>>> However, I need to only play this sound for users >>>>>>>>>>>>> joining the conference from one dialplan but not from another. I cannot >>>>>>>>>>>>> find a way to do it. I looked at >>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>>>>>>>> but it changes the enter sound for the conference for >>>>>>>>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>>>>>>>> same sound. >>>>>>>>>>>>> >>>>>>>>>>>>> Is there a way to assign conference params per user >>>>>>>>>>>>> (like member flags)? >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> If you opt to use master, ALWAYS use latest master >>>>>>>>>>>>>> since that is the nature of master. I have no idea if your problem is >>>>>>>>>>>>>> related to those things but the server I pointed you to test is running >>>>>>>>>>>>>> basically latest master. >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> Thanks Anthony! >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> I tried adding this var and also tried uncommenting >>>>>>>>>>>>>>> the suppress_cng line in my sip profile. It did not seem to make a >>>>>>>>>>>>>>> difference. I pointed my client code to your sample and it worked fine - >>>>>>>>>>>>>>> no noise, so it's not a problem with the client side. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Do I need to upgrade to the latest master? I am using >>>>>>>>>>>>>>> master from 5/22 version 1.5.13b >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Would it help if gave you my test ws server and sip >>>>>>>>>>>>>>> address so you could reproduce this? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Is it possible that the problem is with my AWS >>>>>>>>>>>>>>> instance or CentOS 5.9? >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> I set up your test ext here: >>>>>>>>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> And it seems to work. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Try this in vars.xml >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> I don't think the silence suppression works very well >>>>>>>>>>>>>>>> in the browser. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> In case it was not clear from the previous email - >>>>>>>>>>>>>>>>> this only happens with conferences. If after the beep, instead of placing >>>>>>>>>>>>>>>>> the call into a conference, I just play some music, there are no problems. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and >>>>>>>>>>>>>>>>>> after a user calls in, I play a beep and put them into a conference. Two >>>>>>>>>>>>>>>>>> strange things happen after that: >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> 2. About 3 seconds into the call, there is another >>>>>>>>>>>>>>>>>> very short beep - like a small portion of the original beep that the user >>>>>>>>>>>>>>>>>> hears. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> This absolutely does not happen with softphones - >>>>>>>>>>>>>>>>>> only with WebRTC. I tried it with Chrome and Opera - same thing is >>>>>>>>>>>>>>>>>> happening on both. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> I tried increasing the answer delay from 2 seconds >>>>>>>>>>>>>>>>>> to 3 and the noise now happens more rarely but the extra beep is still >>>>>>>>>>>>>>>>>> there almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>>>>>>>> <action application="playback" >>>>>>>>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>>>>>>>> <action application="conference" >>>>>>>>>>>>>>>>>> data="$1@default"/> >>>>>>>>>>>>>>>>>> </condition> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ >>>>>>>>>>>>>>>> @FreeSWITCH ♬ >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> -- >>>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>>>> >>>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>>> >>>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>> Server >>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>> Server >>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>> >>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>> >>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>> Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> -- >>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>> >>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>> >>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> _________________________________________________________________________ >>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>> consulting@freeswitch.org >>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>> http://www.cudatel.com >>>>>>>>>> >>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>> http://www.cluecon.com >>>>>>>>>> >>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>> >>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>> http://www.freeswitch.org >>>>>>>>>> >>>>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting@freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>> http://www.cudatel.com >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE: >>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > <http://freeswitch.org/g+>* > > ClueCon Weekly Development Call > ☎ sip:888@conference.freeswitch.org ☎ +19193869900 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
And looking closer at Brian's Makefile it seems that my CentOS system already used OpenSSL 1.0.1g. Let me know if there is a way to verify the OpenSSL version FreeSWITCH uses. Otherwise, I will see what my OS options are. Thanks for all your help and I want to make sure it's clear that this is a small problem compared to the amazing functionality FreeSWITCH provides. On Sat, Jun 14, 2014 at 8:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
They all have different OpenSSL versions but they all have the latest available for them.
I tried building FreeSWITCH on my CentOS 5 with the latest OpenSSL using these instructions:
http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_v.1.4.2Fmaster...
I had to fudge it a little to make it work with Brian's Makefile.
The problem is still there but I am not sure switching to the new OpenSSL worked. The active version of OpenSSL on the OS is still old.
Is there a way to check which OpenSSL version FreeSWITCH is using?
On Sat, Jun 14, 2014 at 2:15 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
My only suggestion is check the version of openssl?
On Sat, Jun 14, 2014 at 12:26 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Anthony, I did that. No response yet.
However, I did get a VM with Debian 7, and indeed this problem does not happen there.
I am not sure where to go from here. I am kind of stuck with CentOS as my production system at least for the next several months.
On Thu, Jun 12, 2014 at 11:51 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You can email consulting@freeswitch.org and see if we can find a way.
On Thu, Jun 12, 2014 at 1:28 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation.
You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-)
On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel?
On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com > wrote:
> Thanks, this is identical to what I have. The problem for me is > getting worse on Chrome 36 beta. On 35 or earlier I was at least able to > find a delay setting where the symptoms were minimized. Now on 36 it's > always bad. > > I have an out of the box FreeSWITCH installation on Windows where > all I did was uncomment in the ws-binding in internal.xml sip profile and > added this extension to public.xml dial plan as the very first extension. > Then I changed the prefs on > http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this > FreeSWITCH. > > Could I get some commercial support for this? Should I email > consulting@freeswitch.com? I'd be happy to give you access to my > installations where this is happening. > > > On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale < > anthony.minessale@gmail.com> wrote: > >> <extension name="8765"> >> >> >> >> <condition field="destination_number" expression="^8765$"> >> >> >> >> <action application="set" data="answer_delay=3000"/> >> >> >> >> <action application="answer"/> >> >> >> >> <action application="playback" data= >> "tone_stream://%(1000,0,600)" /> >> >> >> <action application="conference" data="$1@default"/> >> >> >> >> </condition> >> >> >> >> </extension> >> >> >> On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar < >> olegstolyar@gmail.com> wrote: >> >>> Hi Anthony, >>> >>> could you send me the XML for the extension you used to set up >>> this example? >>> >>> >>> >>> On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar < >>> olegstolyar@gmail.com> wrote: >>> >>>> Hey Anthony, >>>> >>>> just in case I am missing something obvious, could you send me >>>> the extension in the dialplan you set up with my example? >>>> >>>> >>>> On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar < >>>> olegstolyar@gmail.com> wrote: >>>> >>>>> Yeah, I saw sipjs vs jssp but I pointed my own jssip client to >>>>> your example and there was no noise, so it's not the client. I also am >>>>> setting my stun servers array to empty to speed up the connection but that >>>>> also does not seem to matter. I'll keep digging. >>>>> >>>>> >>>>> On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale < >>>>> anthony.minessale@gmail.com> wrote: >>>>> >>>>>> Also it's using sipjs.com not jssip. >>>>>> On Jun 8, 2014 8:14 AM, "Anthony Minessale" < >>>>>> anthony.minessale@gmail.com> wrote: >>>>>> >>>>>>> Debian 7 on proxmox vm. >>>>>>> On Jun 6, 2014 10:18 PM, "Oleg Stolyar" < >>>>>>> olegstolyar@gmail.com> wrote: >>>>>>> >>>>>>>> Anthony, >>>>>>>> >>>>>>>> Another update - I thought that the only thing my original >>>>>>>> installation and the Debian VM had in common was that both were VMs. So I >>>>>>>> went ahead and installed FreeSWITCH on my local Windows machine. Same >>>>>>>> problem is happening here as well. >>>>>>>> >>>>>>>> I am extremely curious what's different between your >>>>>>>> installation and my three setups (CentOS, Debian and Windows) that accounts >>>>>>>> for the difference in this behavior. Could you tell me what machine/OS you >>>>>>>> are using? >>>>>>>> >>>>>>>> >>>>>>>> On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar < >>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi Anthony, >>>>>>>>> >>>>>>>>> I dug further and I can reproduce the issue on a completely >>>>>>>>> different VM (Debian on Windows host). I installed the latest FS master >>>>>>>>> and even switched to the vanilla conf to eliminate the possibility of my >>>>>>>>> config causing this issue. >>>>>>>>> >>>>>>>>> It's still happening. If you have time to look into this >>>>>>>>> some more, I'll be happy to upload my whole VM for someone to take a look >>>>>>>>> at. >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar < >>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Yep it's the same. >>>>>>>>>> >>>>>>>>>> Thank you >>>>>>>>>> Oleg >>>>>>>>>> On Jun 4, 2014 4:05 PM, "Anthony Minessale" < >>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> What if you play an audio file? is it the same? >>>>>>>>>>> I can only guess really. >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar < >>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>> >>>>>>>>>>>> I tried this: >>>>>>>>>>>> <action application="set" data="timer_name=soft"/> >>>>>>>>>>>> <action application="playback" >>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>> >>>>>>>>>>>> Didn't help unfortunately (unless this is the wrong >>>>>>>>>>>> syntax for setting the variable). Completely agreed on the AWS instances - >>>>>>>>>>>> in fact we started on an older m1 instance and the voice quality was pretty >>>>>>>>>>>> bad and there were frequent delays. Then we switch to an m3 and it was >>>>>>>>>>>> much better. >>>>>>>>>>>> >>>>>>>>>>>> Any advice on setting enter-sound per member? Or setting >>>>>>>>>>>> a no-enter-sound flag - similar to the nomoh flag? >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale < >>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>> >>>>>>>>>>>>> Try setting the channel var timer_name=soft before >>>>>>>>>>>>> playing the beep. It may be jittery media flow causing your problem. >>>>>>>>>>>>> My only advice on amazon is use the more expensive one >>>>>>>>>>>>> because the lower cost ones are not reliable for cpu time. >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar < >>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>> >>>>>>>>>>>>>> OK, thanks. It's still happening with the latest >>>>>>>>>>>>>> master, so I have to assume that this is something to do with the AWS VMs. >>>>>>>>>>>>>> It would not be the first time strange things happen there. >>>>>>>>>>>>>> >>>>>>>>>>>>>> I was trying to work around the problem and noticed >>>>>>>>>>>>>> that if instead of playing the beep using playback app before placing the >>>>>>>>>>>>>> user into the conference, I make the beep the enter-sound of the >>>>>>>>>>>>>> conference, the problem does not happen. >>>>>>>>>>>>>> >>>>>>>>>>>>>> However, I need to only play this sound for users >>>>>>>>>>>>>> joining the conference from one dialplan but not from another. I cannot >>>>>>>>>>>>>> find a way to do it. I looked at >>>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>>> data="conference_enter_sound=silence_stream://10"/> >>>>>>>>>>>>>> but it changes the enter sound for the conference for >>>>>>>>>>>>>> everyone, so anyone entering after the user who set this up will hear the >>>>>>>>>>>>>> same sound. >>>>>>>>>>>>>> >>>>>>>>>>>>>> Is there a way to assign conference params per user >>>>>>>>>>>>>> (like member flags)? >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale < >>>>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>>>> >>>>>>>>>>>>>>> If you opt to use master, ALWAYS use latest master >>>>>>>>>>>>>>> since that is the nature of master. I have no idea if your problem is >>>>>>>>>>>>>>> related to those things but the server I pointed you to test is running >>>>>>>>>>>>>>> basically latest master. >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar < >>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Thanks Anthony! >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> I tried adding this var and also tried uncommenting >>>>>>>>>>>>>>>> the suppress_cng line in my sip profile. It did not seem to make a >>>>>>>>>>>>>>>> difference. I pointed my client code to your sample and it worked fine - >>>>>>>>>>>>>>>> no noise, so it's not a problem with the client side. >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Do I need to upgrade to the latest master? I am >>>>>>>>>>>>>>>> using master from 5/22 version 1.5.13b >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Would it help if gave you my test ws server and sip >>>>>>>>>>>>>>>> address so you could reproduce this? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Is it possible that the problem is with my AWS >>>>>>>>>>>>>>>> instance or CentOS 5.9? >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale < >>>>>>>>>>>>>>>> anthony.minessale@gmail.com> wrote: >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I set up your test ext here: >>>>>>>>>>>>>>>>> https://webrtc.freeswitch.org/sipjs/index.html#8765 >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> And it seems to work. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Try this in vars.xml >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> <X-PRE-PROCESS cmd="set" data="suppress_cng=true"/> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> I don't think the silence suppression works very >>>>>>>>>>>>>>>>> well in the browser. >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar < >>>>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> In case it was not clear from the previous email - >>>>>>>>>>>>>>>>>> this only happens with conferences. If after the beep, instead of placing >>>>>>>>>>>>>>>>>> the call into a conference, I just play some music, there are no problems. >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar < >>>>>>>>>>>>>>>>>> olegstolyar@gmail.com> wrote: >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> Hi guys, >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> I am connecting to FreeSWITCH using WebRTC and >>>>>>>>>>>>>>>>>>> after a user calls in, I play a beep and put them into a conference. Two >>>>>>>>>>>>>>>>>>> strange things happen after that: >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> 1. There is a noise in the user's leg that someone >>>>>>>>>>>>>>>>>>> described like a wind in the tunnel. The noise lasts for several seconds, >>>>>>>>>>>>>>>>>>> then the user can hear the hold music. The noise does not seem to happen >>>>>>>>>>>>>>>>>>> if I don't play the beep before the conference. >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> 2. About 3 seconds into the call, there is another >>>>>>>>>>>>>>>>>>> very short beep - like a small portion of the original beep that the user >>>>>>>>>>>>>>>>>>> hears. >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> This absolutely does not happen with softphones - >>>>>>>>>>>>>>>>>>> only with WebRTC. I tried it with Chrome and Opera - same thing is >>>>>>>>>>>>>>>>>>> happening on both. >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> I tried increasing the answer delay from 2 seconds >>>>>>>>>>>>>>>>>>> to 3 and the noise now happens more rarely but the extra beep is still >>>>>>>>>>>>>>>>>>> there almost every time. I am using JsSIP 3.0 but since it only handles >>>>>>>>>>>>>>>>>>> signalling I don't think the problem is with the JS library. >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> Has anyone run into this? Any advice? >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> Here is the relevant excerpt from my dialplan. >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> <condition field="destination_number" >>>>>>>>>>>>>>>>>>> expression="^conf-(\S+)$"> >>>>>>>>>>>>>>>>>>> <action application="set" >>>>>>>>>>>>>>>>>>> data="answer_delay=3000"/> >>>>>>>>>>>>>>>>>>> <action application="answer"/> >>>>>>>>>>>>>>>>>>> <action application="playback" >>>>>>>>>>>>>>>>>>> data="tone_stream://%(1000,0,600)" /> >>>>>>>>>>>>>>>>>>> <action application="conference" >>>>>>>>>>>>>>>>>>> data="$1@default"/> >>>>>>>>>>>>>>>>>>> </condition> >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel >>>>>>>>>>>>>>>>>> Communication Server >>>>>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ >>>>>>>>>>>>>>>>> @FreeSWITCH ♬ >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> -- >>>>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>>> Server >>>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>>> >>>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>>> Server >>>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>>> >>>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>>> >>>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> -- >>>>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>>>> >>>>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>>>> >>>>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>>> Server >>>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>>> >>>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>>> >>>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>>> >>>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>>>>>>>>>>> Server >>>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>>> >>>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>>> >>>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> -- >>>>>>>>>>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>>>>>>> >>>>>>>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>>>>>>> http://twitter.com/FreeSWITCH >>>>>>>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>>>>>>> <http://freeswitch.org/g+>* >>>>>>>>>>> >>>>>>>>>>> ClueCon Weekly Development Call >>>>>>>>>>> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> _________________________________________________________________________ >>>>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>>>> consulting@freeswitch.org >>>>>>>>>>> http://www.freeswitchsolutions.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>>>>> http://www.cudatel.com >>>>>>>>>>> >>>>>>>>>>> Official FreeSWITCH Sites >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> http://wiki.freeswitch.org >>>>>>>>>>> http://www.cluecon.com >>>>>>>>>>> >>>>>>>>>>> FreeSWITCH-users mailing list >>>>>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>>>> UNSUBSCRIBE: >>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>>>>> http://www.freeswitch.org >>>>>>>>>>> >>>>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting@freeswitch.org >>>>>>>> http://www.freeswitchsolutions.com >>>>>>>> >>>>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>>>> http://www.cudatel.com >>>>>>>> >>>>>>>> Official FreeSWITCH Sites >>>>>>>> http://www.freeswitch.org >>>>>>>> http://wiki.freeswitch.org >>>>>>>> http://www.cluecon.com >>>>>>>> >>>>>>>> FreeSWITCH-users mailing list >>>>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE: >>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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use ldd? On Jun 15, 2014, at 3:09 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
And looking closer at Brian's Makefile it seems that my CentOS system already used OpenSSL 1.0.1g.
Let me know if there is a way to verify the OpenSSL version FreeSWITCH uses.
Otherwise, I will see what my OS options are.
Thanks for all your help and I want to make sure it's clear that this is a small problem compared to the amazing functionality FreeSWITCH provides.
On Sat, Jun 14, 2014 at 8:01 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: They all have different OpenSSL versions but they all have the latest available for them.
I tried building FreeSWITCH on my CentOS 5 with the latest OpenSSL using these instructions: http://wiki.freeswitch.org/wiki/Installation_Guide#FreeSWITCH_v.1.4.2Fmaster...
I had to fudge it a little to make it work with Brian's Makefile.
The problem is still there but I am not sure switching to the new OpenSSL worked. The active version of OpenSSL on the OS is still old.
Is there a way to check which OpenSSL version FreeSWITCH is using?
On Sat, Jun 14, 2014 at 2:15 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: My only suggestion is check the version of openssl?
On Sat, Jun 14, 2014 at 12:26 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Thanks Anthony, I did that. No response yet.
However, I did get a VM with Debian 7, and indeed this problem does not happen there.
I am not sure where to go from here. I am kind of stuck with CentOS as my production system at least for the next several months.
On Thu, Jun 12, 2014 at 11:51 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: You can email consulting@freeswitch.org and see if we can find a way.
On Thu, Jun 12, 2014 at 1:28 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: I have not tried it on the dev channel but it's happening on both stable and is worse on Beta. It is not happening on your site even with the same stable or beta browser, so it has to be something different with my installation.
You are absolutely right - the first thing to do is reproduce it. I understand that this may take some time and perhaps getting on the phone together, that's why I am asking for commercial support. I don't expect guaranteed results of course - just need some sanity check :-)
On Thu, Jun 12, 2014 at 11:14 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: I'm using 37 (dev channel). I am not sure what we can do to fix the problem if its not something we can reproduce. Have you tried it on canary or dev channel?
On Thu, Jun 12, 2014 at 12:21 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Thanks, this is identical to what I have. The problem for me is getting worse on Chrome 36 beta. On 35 or earlier I was at least able to find a delay setting where the symptoms were minimized. Now on 36 it's always bad.
I have an out of the box FreeSWITCH installation on Windows where all I did was uncomment in the ws-binding in internal.xml sip profile and added this extension to public.xml dial plan as the very first extension. Then I changed the prefs on http://webrtc.freeswitch.org/sipjs/index.html#8765 to point to this FreeSWITCH.
Could I get some commercial support for this? Should I email consulting@freeswitch.com? I'd be happy to give you access to my installations where this is happening.
On Thu, Jun 12, 2014 at 9:55 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: <extension name="8765">
<condition field="destination_number" expression="^8765$">
<action application="set" data="answer_delay=3000"/>
<action application="answer"/>
<action application="playback" data="tone_stream://%(1000,0,600)" />
<action application="conference" data="$1@default"/>
</condition>
</extension>
On Thu, Jun 12, 2014 at 11:47 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Hi Anthony,
could you send me the XML for the extension you used to set up this example?
On Sun, Jun 8, 2014 at 7:11 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Hey Anthony,
just in case I am missing something obvious, could you send me the extension in the dialplan you set up with my example?
On Sun, Jun 8, 2014 at 6:59 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Yeah, I saw sipjs vs jssp but I pointed my own jssip client to your example and there was no noise, so it's not the client. I also am setting my stun servers array to empty to speed up the connection but that also does not seem to matter. I'll keep digging.
On Sun, Jun 8, 2014 at 6:15 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: Also it's using sipjs.com not jssip.
On Jun 8, 2014 8:14 AM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote: Debian 7 on proxmox vm. On Jun 6, 2014 10:18 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote: Anthony,
Another update - I thought that the only thing my original installation and the Debian VM had in common was that both were VMs. So I went ahead and installed FreeSWITCH on my local Windows machine. Same problem is happening here as well.
I am extremely curious what's different between your installation and my three setups (CentOS, Debian and Windows) that accounts for the difference in this behavior. Could you tell me what machine/OS you are using?
On Fri, Jun 6, 2014 at 1:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Hi Anthony,
I dug further and I can reproduce the issue on a completely different VM (Debian on Windows host). I installed the latest FS master and even switched to the vanilla conf to eliminate the possibility of my config causing this issue.
It's still happening. If you have time to look into this some more, I'll be happy to upload my whole VM for someone to take a look at.
On Wed, Jun 4, 2014 at 4:50 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Yep it's the same.
Thank you Oleg
On Jun 4, 2014 4:05 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote: What if you play an audio file? is it the same? I can only guess really.
On Wed, Jun 4, 2014 at 10:25 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote: I tried this: <action application="set" data="timer_name=soft"/> <action application="playback" data="tone_stream://%(1000,0,600)" />
Didn't help unfortunately (unless this is the wrong syntax for setting the variable). Completely agreed on the AWS instances - in fact we started on an older m1 instance and the voice quality was pretty bad and there were frequent delays. Then we switch to an m3 and it was much better.
Any advice on setting enter-sound per member? Or setting a no-enter-sound flag - similar to the nomoh flag?
On Wed, Jun 4, 2014 at 7:39 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: Try setting the channel var timer_name=soft before playing the beep. It may be jittery media flow causing your problem. My only advice on amazon is use the more expensive one because the lower cost ones are not reliable for cpu time.
On Tue, Jun 3, 2014 at 10:30 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: OK, thanks. It's still happening with the latest master, so I have to assume that this is something to do with the AWS VMs. It would not be the first time strange things happen there.
I was trying to work around the problem and noticed that if instead of playing the beep using playback app before placing the user into the conference, I make the beep the enter-sound of the conference, the problem does not happen.
However, I need to only play this sound for users joining the conference from one dialplan but not from another. I cannot find a way to do it. I looked at <action application="set" data="conference_enter_sound=silence_stream://10"/> but it changes the enter sound for the conference for everyone, so anyone entering after the user who set this up will hear the same sound.
Is there a way to assign conference params per user (like member flags)?
On Tue, Jun 3, 2014 at 12:25 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote: If you opt to use master, ALWAYS use latest master since that is the nature of master. I have no idea if your problem is related to those things but the server I pointed you to test is running basically latest master.
On Tue, Jun 3, 2014 at 2:07 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Thanks Anthony!
I tried adding this var and also tried uncommenting the suppress_cng line in my sip profile. It did not seem to make a difference. I pointed my client code to your sample and it worked fine - no noise, so it's not a problem with the client side.
Do I need to upgrade to the latest master? I am using master from 5/22 version 1.5.13b
Would it help if gave you my test ws server and sip address so you could reproduce this?
Is it possible that the problem is with my AWS instance or CentOS 5.9?
On Tue, Jun 3, 2014 at 11:23 AM, Anthony Minessale <anthony.minessale@gmail.com> wrote: I set up your test ext here: https://webrtc.freeswitch.org/sipjs/index.html#8765
And it seems to work.
Try this in vars.xml
<X-PRE-PROCESS cmd="set" data="suppress_cng=true"/>
I don't think the silence suppression works very well in the browser.
On Mon, Jun 2, 2014 at 7:23 AM, Oleg Stolyar <olegstolyar@gmail.com> wrote: In case it was not clear from the previous email - this only happens with conferences. If after the beep, instead of placing the call into a conference, I just play some music, there are no problems.
On Sat, May 31, 2014 at 5:46 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote: Hi guys,
I am connecting to FreeSWITCH using WebRTC and after a user calls in, I play a beep and put them into a conference. Two strange things happen after that:
1. There is a noise in the user's leg that someone described like a wind in the tunnel. The noise lasts for several seconds, then the user can hear the hold music. The noise does not seem to happen if I don't play the beep before the conference.
2. About 3 seconds into the call, there is another very short beep - like a small portion of the original beep that the user hears.
This absolutely does not happen with softphones - only with WebRTC. I tried it with Chrome and Opera - same thing is happening on both.
I tried increasing the answer delay from 2 seconds to 3 and the noise now happens more rarely but the extra beep is still there almost every time. I am using JsSIP 3.0 but since it only handles signalling I don't think the problem is with the JS library.
Has anyone run into this? Any advice?
Here is the relevant excerpt from my dialplan.
<condition field="destination_number" expression="^conf-(\S+)$"> <action application="set" data="answer_delay=3000"/> <action application="answer"/> <action application="playback" data="tone_stream://%(1000,0,600)" /> <action application="conference" data="$1@default"/> </condition>
Just use my make file to build FreeSWITCH too, make freeswitch, otherwise you'll have to manually say where all your newly built libs are located. See the configure section of the Makefile.centos5 PKG_CONFIG_PATH=$(PREFIX)/lib/pkgconfig ./configure LDFLAGS='-L$(PREFIX)/lib -Wl,-rpath=$(PREFIX)/lib' CFLAGS='-I$(PREFIX)/include' --prefix=$(FSPREFIX) this will ensure that you always use the libs that my makefile built. Its a kludge but it prevents the configure and build from accidentally using something else in your systems ld path. -- Brian West brian@freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Jun 14, 2014, at 10:09 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
And looking closer at Brian's Makefile it seems that my CentOS system already used OpenSSL 1.0.1g.
Let me know if there is a way to verify the OpenSSL version FreeSWITCH uses.
Otherwise, I will see what my OS options are.
Never underestimate the power of desperation and monkeying around with things you don't understand :-) In case someone else runs into this problem: After all else failed I figured that this may have something to do with conference settings since this problem never happened outside of a conference. I started changing conference profile params. Lo and behold, after I changed the sample rate param to 12000 (from the default 8000) the problem went away. Interestingly, at 11999 it still happened but at 12000 or above, it completely disappeared. Frankly, I have no idea why but I'll take it :-) On Mon, Jun 16, 2014 at 11:39 AM, Brian West <brian@freeswitch.org> wrote:
Just use my make file to build FreeSWITCH too, make freeswitch, otherwise you'll have to manually say where all your newly built libs are located. See the configure section of the Makefile.centos5
PKG_CONFIG_PATH=$(PREFIX)/lib/pkgconfig ./configure LDFLAGS='-L$(PREFIX)/lib -Wl,-rpath=$(PREFIX)/lib' CFLAGS='-I$(PREFIX)/include' --prefix=$(FSPREFIX)
this will ensure that you always use the libs that my makefile built. Its a kludge but it prevents the configure and build from accidentally using something else in your systems ld path.
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 14, 2014, at 10:09 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
And looking closer at Brian's Makefile it seems that my CentOS system already used OpenSSL 1.0.1g.
Let me know if there is a way to verify the OpenSSL version FreeSWITCH uses.
Otherwise, I will see what my OS options are.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
What you're saying is IMPOSSIBLE to have happened. The rates are checked, the only values accepted are 8000/12000/16000/24000/32000/48000, Had you set it to 11999 it would have defaulted to the 8000. So don't stop looking because you've yet to find the real problem. -- Brian West brian@freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Jun 16, 2014, at 1:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Never underestimate the power of desperation and monkeying around with things you don't understand :-)
In case someone else runs into this problem:
After all else failed I figured that this may have something to do with conference settings since this problem never happened outside of a conference. I started changing conference profile params. Lo and behold, after I changed the sample rate param to 12000 (from the default 8000) the problem went away. Interestingly, at 11999 it still happened but at 12000 or above, it completely disappeared.
Frankly, I have no idea why but I'll take it :-)
Ah, but that's completely consistent with what I am seeing. This explains why 11999 still exhibited the problem - it simply kept using 8000. The question for me is why did the problem go away at 12000. On Mon, Jun 16, 2014 at 1:25 PM, Brian West <brian@freeswitch.org> wrote:
What you're saying is IMPOSSIBLE to have happened. The rates are checked, the only values accepted are 8000/12000/16000/24000/32000/48000, Had you set it to 11999 it would have defaulted to the 8000.
So don't stop looking because you've yet to find the real problem.
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 1:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Never underestimate the power of desperation and monkeying around with things you don't understand :-)
In case someone else runs into this problem:
After all else failed I figured that this may have something to do with conference settings since this problem never happened outside of a conference. I started changing conference profile params. Lo and behold, after I changed the sample rate param to 12000 (from the default 8000) the problem went away. Interestingly, at 11999 it still happened but at 12000 or above, it completely disappeared.
Frankly, I have no idea why but I'll take it :-)
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Just realized that I may have confused everyone is by saying "12000 and above". The only value I I really tried above 12000 was 16000. Thanks for clearing it up Brian! On Mon, Jun 16, 2014 at 1:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Ah, but that's completely consistent with what I am seeing. This explains why 11999 still exhibited the problem - it simply kept using 8000.
The question for me is why did the problem go away at 12000.
On Mon, Jun 16, 2014 at 1:25 PM, Brian West <brian@freeswitch.org> wrote:
What you're saying is IMPOSSIBLE to have happened. The rates are checked, the only values accepted are 8000/12000/16000/24000/32000/48000, Had you set it to 11999 it would have defaulted to the 8000.
So don't stop looking because you've yet to find the real problem.
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 1:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Never underestimate the power of desperation and monkeying around with things you don't understand :-)
In case someone else runs into this problem:
After all else failed I figured that this may have something to do with conference settings since this problem never happened outside of a conference. I started changing conference profile params. Lo and behold, after I changed the sample rate param to 12000 (from the default 8000) the problem went away. Interestingly, at 11999 it still happened but at 12000 or above, it completely disappeared.
Frankly, I have no idea why but I'll take it :-)
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
That sir is the Mystery, So you're doing WebRTC into an 8k CONF right? -- Brian West brian@freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Jun 16, 2014, at 3:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Ah, but that's completely consistent with what I am seeing. This explains why 11999 still exhibited the problem - it simply kept using 8000.
The question for me is why did the problem go away at 12000.
On Mon, Jun 16, 2014 at 1:25 PM, Brian West <brian@freeswitch.org> wrote: What you're saying is IMPOSSIBLE to have happened. The rates are checked, the only values accepted are 8000/12000/16000/24000/32000/48000, Had you set it to 11999 it would have defaulted to the 8000.
So don't stop looking because you've yet to find the real problem.
-- Brian West brian@freeswitch.org
Originally when the problem was occurring? Yes, I was using the default conference profile from the vanilla conf. To recap, the problem happened on Windows 7, CentOS 5.9, Debian 6 but it worked fine on Debian 7. On Mon, Jun 16, 2014 at 2:03 PM, Brian West <brian@freeswitch.org> wrote:
That sir is the Mystery, So you're doing WebRTC into an 8k CONF right?
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 3:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Ah, but that's completely consistent with what I am seeing. This explains why 11999 still exhibited the problem - it simply kept using 8000.
The question for me is why did the problem go away at 12000.
On Mon, Jun 16, 2014 at 1:25 PM, Brian West <brian@freeswitch.org> wrote:
What you're saying is IMPOSSIBLE to have happened. The rates are checked, the only values accepted are 8000/12000/16000/24000/32000/48000, Had you set it to 11999 it would have defaulted to the 8000.
So don't stop looking because you've yet to find the real problem.
-- Brian West brian@freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Platform problem then. -- Brian West brian@freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Jun 16, 2014, at 4:10 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
it worked fine on Debian 7.
opus is 48khz so you should really be calling the cdquality conference profile when using webrtc. 3300 in the default config. On Mon, Jun 16, 2014 at 4:13 PM, Brian West <brian@freeswitch.org> wrote:
Platform problem then.
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 4:10 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
it worked fine on Debian 7.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that? My system is in production already and nobody's been complaining about voice quality yet. On Mon, Jun 16, 2014 at 6:29 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
opus is 48khz so you should really be calling the cdquality conference profile when using webrtc. 3300 in the default config.
On Mon, Jun 16, 2014 at 4:13 PM, Brian West <brian@freeswitch.org> wrote:
Platform problem then.
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 4:10 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
it worked fine on Debian 7.
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Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly. You should also be using 1.0.1h, NOT g. -- Brian West brian@freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
Totally agree about the desire to figure it out. If we can do it, that would great. PCMU vs OPUS is a separate issue from the noise after beep. The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var. However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile. Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it? Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time. On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
-- Brian West brian@freeswitch.org
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu. On Jun 17, 2014, at 1:26 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine.
On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote: Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote: Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
-- Brian West brian@freeswitch.org
<whmcslogo.png>
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other. On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu.
On Jun 17, 2014, at 1:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
-- Brian West brian@freeswitch.org
<whmcslogo.png>
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Looks like if you add <action application="set" data="rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS" /> before you call answer or the conference it fixes it. So the media engine is getting confused by our timestamps at the beginning of the call. On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other.
On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu.
On Jun 17, 2014, at 1:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
-- Brian West brian@freeswitch.org
<whmcslogo.png>
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
Thanks Tony! Will there be any side effects if I use this setting? Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other.
On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu.
On Jun 17, 2014, at 1:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
-- Brian West brian@freeswitch.org
<whmcslogo.png>
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
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No, it just disables the proper behavior of our rtp stack which is to have time still pass when not sending RTP. With the param enabled, the timestamps only go up when we are actually writing packets which is incorrect but does fix your problem ;) On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Tony!
Will there be any side effects if I use this setting?
Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other.
On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu.
On Jun 17, 2014, at 1:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
Why aren't you doing OPUS? Are you not liking the ability to hear and understand EVERYONE in the conference? Makes little sense to me, Sounds like your browser has a bug when PCMU is used at 8k I would really love to know what the problem is and the scenario that triggers it. Its unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have nobody complaining is they've never really experienced WebRTC properly.
You should also be using 1.0.1h, NOT g.
-- Brian West brian@freeswitch.org
<whmcslogo.png>
Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com
T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
My negotiations say I am using PCMU, so since 12000 works for me and 48000 will take extra CPU, any reason I can't just use that?
My system is in production already and nobody's been complaining about voice quality yet.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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It won't disable rtp timeout though, right? Thank you Oleg On Aug 5, 2014 1:34 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, it just disables the proper behavior of our rtp stack which is to have time still pass when not sending RTP. With the param enabled, the timestamps only go up when we are actually writing packets which is incorrect but does fix your problem ;)
On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Tony!
Will there be any side effects if I use this setting?
Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other.
On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu.
On Jun 17, 2014, at 1:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
Totally agree about the desire to figure it out. If we can do it, that would great.
PCMU vs OPUS is a separate issue from the noise after beep.
The PCMU part is happening because even though I am using latest master, my conf is old and I don't have OPUS in my global_codec_prefs var.
However, the original problem of noise after beep in the conference happens with the latest conf using OPUS as well. Easily reproducible on vanilla config on Windows 7. Of course switching to cdquality fixed it as did switching to 12000 in the default conference profile.
Not sure I understand the remark about the ability to hear and understand everyone in the conference. I have three types of clients who all mix and match in the conference - a PSTN termination provider (with SIP connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since the non-WebRTC end points don't support OPUS, I am not sure using it will help in this case. Will it?
Also, if I do use OPUS, any advice on how I can limit the bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth limit in it. My users are all in controlled bandwidth environments, so I need to make sure a single user does not use more than 80-100 kpbs (in each direction) even if more is available at the time.
On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> wrote:
> Why aren't you doing OPUS? Are you not liking the ability to hear > and understand EVERYONE in the conference? Makes little sense to me, > Sounds like your browser has a bug when PCMU is used at 8k I would really > love to know what the problem is and the scenario that triggers it. Its > unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have > nobody complaining is they've never really experienced WebRTC properly. > > You should also be using 1.0.1h, NOT g. > > -- > Brian West > brian@freeswitch.org > > <whmcslogo.png> > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9001 > ISN: 410*543 > Skype:briankwest > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) > > > > > > > > > > > > > > > > On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> > wrote: > > My negotiations say I am using PCMU, so since 12000 works for me and > 48000 will take extra CPU, any reason I can't just use that? > > My system is in production already and nobody's been complaining > about voice quality yet. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
No, RTP timeout will still work. I enabled this flag by default now when webrtc is enabled since every browser seems to suffer from it. On Tue, Aug 5, 2014 at 2:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
It won't disable rtp timeout though, right?
Thank you Oleg On Aug 5, 2014 1:34 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, it just disables the proper behavior of our rtp stack which is to have time still pass when not sending RTP. With the param enabled, the timestamps only go up when we are actually writing packets which is incorrect but does fix your problem ;)
On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Tony!
Will there be any side effects if I use this setting?
Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other.
On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
if you have to up the rate to get around this issue, setting it to 16k instead of 12k when your endpoints are all 8k should use less cpu.
On Jun 17, 2014, at 1:26 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
For what it's worth, opus usually costs about the same bandwidth of ulaw. If you are using pstn callers anyway ulaw is fine. On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> wrote:
> Totally agree about the desire to figure it out. If we can do it, > that would great. > > PCMU vs OPUS is a separate issue from the noise after beep. > > The PCMU part is happening because even though I am using latest > master, my conf is old and I don't have OPUS in my global_codec_prefs var. > > However, the original problem of noise after beep in the conference > happens with the latest conf using OPUS as well. Easily reproducible on > vanilla config on Windows 7. Of course switching to cdquality fixed it as > did switching to 12000 in the default conference profile. > > Not sure I understand the remark about the ability to hear and > understand everyone in the conference. I have three types of clients who > all mix and match in the conference - a PSTN termination provider (with SIP > connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since > the non-WebRTC end points don't support OPUS, I am not sure using it will > help in this case. Will it? > > Also, if I do use OPUS, any advice on how I can limit the bandwidth > it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth > limit in it. My users are all in controlled bandwidth environments, so I > need to make sure a single user does not use more than 80-100 kpbs (in each > direction) even if more is available at the time. > > > > On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> > wrote: > >> Why aren't you doing OPUS? Are you not liking the ability to hear >> and understand EVERYONE in the conference? Makes little sense to me, >> Sounds like your browser has a bug when PCMU is used at 8k I would really >> love to know what the problem is and the scenario that triggers it. Its >> unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have >> nobody complaining is they've never really experienced WebRTC properly. >> >> You should also be using 1.0.1h, NOT g. >> >> -- >> Brian West >> brian@freeswitch.org >> >> <whmcslogo.png> >> >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >> iNUM: +883 5100 1420 9001 >> ISN: 410*543 >> Skype:briankwest >> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> >> wrote: >> >> My negotiations say I am using PCMU, so since 12000 works for me >> and 48000 will take extra CPU, any reason I can't just use that? >> >> My system is in production already and nobody's been complaining >> about voice quality yet. >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Out of curiosity - why was this not happening on Debian 7? Thank you Oleg On Aug 5, 2014 3:29 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, RTP timeout will still work. I enabled this flag by default now when webrtc is enabled since every browser seems to suffer from it.
On Tue, Aug 5, 2014 at 2:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
It won't disable rtp timeout though, right?
Thank you Oleg On Aug 5, 2014 1:34 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, it just disables the proper behavior of our rtp stack which is to have time still pass when not sending RTP. With the param enabled, the timestamps only go up when we are actually writing packets which is incorrect but does fix your problem ;)
On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Tony!
Will there be any side effects if I use this setting?
Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Michael! I've been testing 12000 and 16000 for a couple of days now. So far 12000 seems to use marginally (very marginally) less CPU than 16000. I'll monitor CPU for several more days before settling on one or the other.
On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> wrote:
> if you have to up the rate to get around this issue, setting it to > 16k instead of 12k when your endpoints are all 8k should use less cpu. > > On Jun 17, 2014, at 1:26 PM, Anthony Minessale < > anthony.minessale@gmail.com> wrote: > > For what it's worth, opus usually costs about the same bandwidth of > ulaw. If you are using pstn callers anyway ulaw is fine. > On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> > wrote: > >> Totally agree about the desire to figure it out. If we can do it, >> that would great. >> >> PCMU vs OPUS is a separate issue from the noise after beep. >> >> The PCMU part is happening because even though I am using latest >> master, my conf is old and I don't have OPUS in my global_codec_prefs var. >> >> However, the original problem of noise after beep in the conference >> happens with the latest conf using OPUS as well. Easily reproducible on >> vanilla config on Windows 7. Of course switching to cdquality fixed it as >> did switching to 12000 in the default conference profile. >> >> Not sure I understand the remark about the ability to hear and >> understand everyone in the conference. I have three types of clients who >> all mix and match in the conference - a PSTN termination provider (with SIP >> connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since >> the non-WebRTC end points don't support OPUS, I am not sure using it will >> help in this case. Will it? >> >> Also, if I do use OPUS, any advice on how I can limit the bandwidth >> it uses? I saw opus.conf.xml but it was not clear how to set a bandwidth >> limit in it. My users are all in controlled bandwidth environments, so I >> need to make sure a single user does not use more than 80-100 kpbs (in each >> direction) even if more is available at the time. >> >> >> >> On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> >> wrote: >> >>> Why aren't you doing OPUS? Are you not liking the ability to hear >>> and understand EVERYONE in the conference? Makes little sense to me, >>> Sounds like your browser has a bug when PCMU is used at 8k I would really >>> love to know what the problem is and the scenario that triggers it. Its >>> unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have >>> nobody complaining is they've never really experienced WebRTC properly. >>> >>> You should also be using 1.0.1h, NOT g. >>> >>> -- >>> Brian West >>> brian@freeswitch.org >>> >>> <whmcslogo.png> >>> >>> Twitter: @FreeSWITCH , @briankwest >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>> iNUM: +883 5100 1420 9001 >>> ISN: 410*543 >>> Skype:briankwest >>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> >>> wrote: >>> >>> My negotiations say I am using PCMU, so since 12000 works for me >>> and 48000 will take extra CPU, any reason I can't just use that? >>> >>> My system is in production already and nobody's been complaining >>> about voice quality yet. >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I have no idea really. Maybe it did and you missed a step reproducing it? On Tue, Aug 5, 2014 at 3:51 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Out of curiosity - why was this not happening on Debian 7?
Thank you Oleg On Aug 5, 2014 3:29 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, RTP timeout will still work. I enabled this flag by default now when webrtc is enabled since every browser seems to suffer from it.
On Tue, Aug 5, 2014 at 2:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
It won't disable rtp timeout though, right?
Thank you Oleg On Aug 5, 2014 1:34 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, it just disables the proper behavior of our rtp stack which is to have time still pass when not sending RTP. With the param enabled, the timestamps only go up when we are actually writing packets which is incorrect but does fix your problem ;)
On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Tony!
Will there be any side effects if I use this setting?
Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com > wrote:
> Thanks Michael! I've been testing 12000 and 16000 for a couple of > days now. So far 12000 seems to use marginally (very marginally) less CPU > than 16000. I'll monitor CPU for several more days before settling on one > or the other. > > > On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> > wrote: > >> if you have to up the rate to get around this issue, setting it to >> 16k instead of 12k when your endpoints are all 8k should use less cpu. >> >> On Jun 17, 2014, at 1:26 PM, Anthony Minessale < >> anthony.minessale@gmail.com> wrote: >> >> For what it's worth, opus usually costs about the same bandwidth of >> ulaw. If you are using pstn callers anyway ulaw is fine. >> On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> >> wrote: >> >>> Totally agree about the desire to figure it out. If we can do it, >>> that would great. >>> >>> PCMU vs OPUS is a separate issue from the noise after beep. >>> >>> The PCMU part is happening because even though I am using latest >>> master, my conf is old and I don't have OPUS in my global_codec_prefs var. >>> >>> However, the original problem of noise after beep in the >>> conference happens with the latest conf using OPUS as well. Easily >>> reproducible on vanilla config on Windows 7. Of course switching to >>> cdquality fixed it as did switching to 12000 in the default conference >>> profile. >>> >>> Not sure I understand the remark about the ability to hear and >>> understand everyone in the conference. I have three types of clients who >>> all mix and match in the conference - a PSTN termination provider (with SIP >>> connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since >>> the non-WebRTC end points don't support OPUS, I am not sure using it will >>> help in this case. Will it? >>> >>> Also, if I do use OPUS, any advice on how I can limit the >>> bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a >>> bandwidth limit in it. My users are all in controlled bandwidth >>> environments, so I need to make sure a single user does not use more than >>> 80-100 kpbs (in each direction) even if more is available at the time. >>> >>> >>> >>> On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> >>> wrote: >>> >>>> Why aren't you doing OPUS? Are you not liking the ability to hear >>>> and understand EVERYONE in the conference? Makes little sense to me, >>>> Sounds like your browser has a bug when PCMU is used at 8k I would really >>>> love to know what the problem is and the scenario that triggers it. Its >>>> unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have >>>> nobody complaining is they've never really experienced WebRTC properly. >>>> >>>> You should also be using 1.0.1h, NOT g. >>>> >>>> -- >>>> Brian West >>>> brian@freeswitch.org >>>> >>>> <whmcslogo.png> >>>> >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> >>>> wrote: >>>> >>>> My negotiations say I am using PCMU, so since 12000 works for me >>>> and 48000 will take extra CPU, any reason I can't just use that? >>>> >>>> My system is in production already and nobody's been complaining >>>> about voice quality yet. >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
It was not happening on your webrtc demo when you originally tried to reproduce it with 8765 extension. Then it was not happening when I tried it on Debian 7. It's probably not worth investigating - I was just curious. Thank you Oleg I have no idea really. Maybe it did and you missed a step reproducing it? On Tue, Aug 5, 2014 at 3:51 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Out of curiosity - why was this not happening on Debian 7?
Thank you Oleg On Aug 5, 2014 3:29 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, RTP timeout will still work. I enabled this flag by default now when webrtc is enabled since every browser seems to suffer from it.
On Tue, Aug 5, 2014 at 2:06 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
It won't disable rtp timeout though, right?
Thank you Oleg On Aug 5, 2014 1:34 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
No, it just disables the proper behavior of our rtp stack which is to have time still pass when not sending RTP. With the param enabled, the timestamps only go up when we are actually writing packets which is incorrect but does fix your problem ;)
On Tue, Aug 5, 2014 at 12:48 PM, Oleg Stolyar <olegstolyar@gmail.com> wrote:
Thanks Tony!
Will there be any side effects if I use this setting?
Oleg On Aug 5, 2014 12:06 PM, "Anthony Minessale" < anthony.minessale@gmail.com> wrote:
Looks like if you add
<action application="set" data= "rtp_manual_rtp_bugs=SEND_LINEAR_TIMESTAMPS"/>
before you call answer or the conference it fixes it.
So the media engine is getting confused by our timestamps at the beginning of the call.
On Tue, Jun 17, 2014 at 12:27 PM, Oleg Stolyar <olegstolyar@gmail.com > wrote:
> Thanks Michael! I've been testing 12000 and 16000 for a couple of > days now. So far 12000 seems to use marginally (very marginally) less CPU > than 16000. I'll monitor CPU for several more days before settling on one > or the other. > > > On Tue, Jun 17, 2014 at 7:27 AM, Michael Jerris <mike@jerris.com> > wrote: > >> if you have to up the rate to get around this issue, setting it to >> 16k instead of 12k when your endpoints are all 8k should use less cpu. >> >> On Jun 17, 2014, at 1:26 PM, Anthony Minessale < >> anthony.minessale@gmail.com> wrote: >> >> For what it's worth, opus usually costs about the same bandwidth of >> ulaw. If you are using pstn callers anyway ulaw is fine. >> On Jun 16, 2014 11:23 PM, "Oleg Stolyar" <olegstolyar@gmail.com> >> wrote: >> >>> Totally agree about the desire to figure it out. If we can do it, >>> that would great. >>> >>> PCMU vs OPUS is a separate issue from the noise after beep. >>> >>> The PCMU part is happening because even though I am using latest >>> master, my conf is old and I don't have OPUS in my global_codec_prefs var. >>> >>> However, the original problem of noise after beep in the >>> conference happens with the latest conf using OPUS as well. Easily >>> reproducible on vanilla config on Windows 7. Of course switching to >>> cdquality fixed it as did switching to 12000 in the default conference >>> profile. >>> >>> Not sure I understand the remark about the ability to hear and >>> understand everyone in the conference. I have three types of clients who >>> all mix and match in the conference - a PSTN termination provider (with SIP >>> connection/PCMU), SIP softphone (PCMU) and Chrome WebRTC (JsSIP). Since >>> the non-WebRTC end points don't support OPUS, I am not sure using it will >>> help in this case. Will it? >>> >>> Also, if I do use OPUS, any advice on how I can limit the >>> bandwidth it uses? I saw opus.conf.xml but it was not clear how to set a >>> bandwidth limit in it. My users are all in controlled bandwidth >>> environments, so I need to make sure a single user does not use more than >>> 80-100 kpbs (in each direction) even if more is available at the time. >>> >>> >>> >>> On Mon, Jun 16, 2014 at 8:14 PM, Brian West <brian@freeswitch.org> >>> wrote: >>> >>>> Why aren't you doing OPUS? Are you not liking the ability to hear >>>> and understand EVERYONE in the conference? Makes little sense to me, >>>> Sounds like your browser has a bug when PCMU is used at 8k I would really >>>> love to know what the problem is and the scenario that triggers it. Its >>>> unacceptable to say 'Oh set it to 12000 it fixes it'. The reason you have >>>> nobody complaining is they've never really experienced WebRTC properly. >>>> >>>> You should also be using 1.0.1h, NOT g. >>>> >>>> -- >>>> Brian West >>>> brian@freeswitch.org >>>> >>>> <whmcslogo.png> >>>> >>>> Twitter: @FreeSWITCH , @briankwest >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>> iNUM: +883 5100 1420 9001 >>>> ISN: 410*543 >>>> Skype:briankwest >>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> On Jun 16, 2014, at 9:55 PM, Oleg Stolyar <olegstolyar@gmail.com> >>>> wrote: >>>> >>>> My negotiations say I am using PCMU, so since 12000 works for me >>>> and 48000 will take extra CPU, any reason I can't just use that? >>>> >>>> My system is in production already and nobody's been complaining >>>> about voice quality yet. >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting@freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
-
Anthony Minessale
-
Brian West
-
Michael Jerris
-
Oleg Stolyar