[Freeswitch-users] Strange media behavior on WebRTC

Anthony Minessale anthony.minessale at gmail.com
Wed Jun 4 18:39:27 MSD 2014 

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 at 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 at 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 at 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 at 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 at 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 at 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 at default"/>
>>>>>> </condition>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
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>>>>> UNSUBSCRIBE:
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>>>>> 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+>*
>>>>
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>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
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>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
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>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>>
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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 at conference.freeswitch.org  ☎ +19193869900
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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