Hello, I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup. 1) What is the best codec to use when there is going to be jitter and some packetloss? 2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles? The problem that I am trying to solve is this... Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues. Any help with this would be appreciated! Thanks in advance. Josh
Hi Josh, Use FS master, it has a new audio Jitter Buffer. Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it. You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/> Regards, Dragos On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Dragos, Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”. Best regards /Ola From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Dragos Oancea Sent: den 12 november 2015 10:14 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Jitterbuffer Hi Josh, Use FS master, it has a new audio Jitter Buffer. Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it. You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/> Regards, Dragos On 11/12/2015 07:51 AM, Josh Richesin wrote: Hello, I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup. 1) What is the best codec to use when there is going to be jitter and some packetloss? 2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles? The problem that I am trying to solve is this... Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues. Any help with this would be appreciated! Thanks in advance. Josh _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
We probably need to wipe confluence clean of the incorrect jitter buffer docs now that its been completely rewritten in the past few months. On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström <Ola.Backstrom@loxysoft.se> wrote:
Dragos,
Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”.
Best regards
/Ola
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Dragos Oancea *Sent:* den 12 november 2015 10:14 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Jitterbuffer
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Hello all! Thanks for the recommendations on upgrading to FS master, (Assuming that is 1.6.2?) and using the new Jitter Buffer and OPUS codec. Unfortunately, after researching and working with our ATA vendor OBI, they do not support OPUS. So that has me form a few more questions on this issue. 1) Is there any documentation on the new Jitter Buffer being used? 2) Is there any reason to believe that upgrading from our current Free Switch Version 1.4.18 to 1.6.2 would fix the issue with the current codec of g711, even tho I can not use opus? 3) Below is the real issue that I am trying to fix: obi200------------> fixed wireless --------> FreeSwitch ------PSTN Call Quality issues are mainly reported by the end user on the PSTN network. Users report that the calls are cutting in and out but our users on the obi200 side hear them perfectly. Any ideas on settings to try to optimize in this situation? I know that it is a network related issue, with high jitter, however I know there must be a fix as other VOIP providers can ride the same network path and have no issues at all. 4) Main reason for not just upgrading is that I am using fushionPBX, and they do not yet support 1.6.2, and also trying to figure out an actual upgrade plan so that I do not loose all the customers voicemail's, VM greetings passwords and such. Any help is greatly appreciated with what direction I should go. Thanks! Josh On Fri, Nov 13, 2015 at 9:45 AM, Brian West <brian@freeswitch.org> wrote:
We probably need to wipe confluence clean of the incorrect jitter buffer docs now that its been completely rewritten in the past few months.
On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström <Ola.Backstrom@loxysoft.se> wrote:
Dragos,
Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”.
Best regards
/Ola
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Dragos Oancea *Sent:* den 12 november 2015 10:14 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Jitterbuffer
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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--
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*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Are you sure the jitter/loss in on the fixed wireless side? Is the freeswitch<->pstn sip? Is freeswitch transcoding between the legs? There is a tool called mtr to measure the loss/jitter along the path. If freeswitch is not transcoding then it doesn't need a jitter buffer. You could actually use bypass_media to take freeswitch out of the media path and see if that helps. https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview On Thu, Nov 19, 2015 at 11:07 PM, Josh Richesin <jlrichesin@gmail.com> wrote:
Hello all!
Thanks for the recommendations on upgrading to FS master, (Assuming that is 1.6.2?) and using the new Jitter Buffer and OPUS codec. Unfortunately, after researching and working with our ATA vendor OBI, they do not support OPUS. So that has me form a few more questions on this issue.
1) Is there any documentation on the new Jitter Buffer being used?
2) Is there any reason to believe that upgrading from our current Free Switch Version 1.4.18 to 1.6.2 would fix the issue with the current codec of g711, even tho I can not use opus?
3) Below is the real issue that I am trying to fix:
obi200------------> fixed wireless --------> FreeSwitch ------PSTN
Call Quality issues are mainly reported by the end user on the PSTN network. Users report that the calls are cutting in and out but our users on the obi200 side hear them perfectly. Any ideas on settings to try to optimize in this situation?
I know that it is a network related issue, with high jitter, however I know there must be a fix as other VOIP providers can ride the same network path and have no issues at all.
4) Main reason for not just upgrading is that I am using fushionPBX, and they do not yet support 1.6.2, and also trying to figure out an actual upgrade plan so that I do not loose all the customers voicemail's, VM greetings passwords and such.
Any help is greatly appreciated with what direction I should go.
Thanks!
Josh
On Fri, Nov 13, 2015 at 9:45 AM, Brian West <brian@freeswitch.org> wrote:
We probably need to wipe confluence clean of the incorrect jitter buffer docs now that its been completely rewritten in the past few months.
On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström <Ola.Backstrom@loxysoft.se
wrote:
Dragos,
Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”.
Best regards
/Ola
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Dragos Oancea *Sent:* den 12 november 2015 10:14 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Jitterbuffer
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
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*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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 Josh, FS master is not 1.6.2 , it's the latest master branch . Get it from git and compile it. 1.6.2 does not have the new JB. I assume you don't transcode at all and you use G.711 along the path. Try using ILBC between the OBI200 and Freeswitch , so you would do transcoding with Jitter Buffer and PLC enabled. It would be better if you don't transcode at all, but your PSTN provider should offer ILBC or other compressed codec that you should try and you would also need to rely on them for PLC .You should probably enable the Jitter Buffer on the bridge if you do transcoding : <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> Regards, Dragos PS: Obihai supports Opus in their phones: http://www.obihai.com/partner/obi1000 . On 11/20/2015 07:22 AM, Russell Treleaven wrote:
Are you sure the jitter/loss in on the fixed wireless side?
Is the freeswitch<->pstn sip?
Is freeswitch transcoding between the legs?
There is a tool called mtr to measure the loss/jitter along the path.
If freeswitch is not transcoding then it doesn't need a jitter buffer. You could actually use bypass_media to take freeswitch out of the media path and see if that helps. https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview
On Thu, Nov 19, 2015 at 11:07 PM, Josh Richesin <jlrichesin@gmail.com <mailto:jlrichesin@gmail.com>> wrote:
Hello all!
Thanks for the recommendations on upgrading to FS master, (Assuming that is 1.6.2?) and using the new Jitter Buffer and OPUS codec. Unfortunately, after researching and working with our ATA vendor OBI, they do not support OPUS. So that has me form a few more questions on this issue.
1) Is there any documentation on the new Jitter Buffer being used?
2) Is there any reason to believe that upgrading from our current Free Switch Version 1.4.18 to 1.6.2 would fix the issue with the current codec of g711, even tho I can not use opus?
3) Below is the real issue that I am trying to fix:
obi200------------> fixed wireless --------> FreeSwitch ------PSTN
Call Quality issues are mainly reported by the end user on the PSTN network. Users report that the calls are cutting in and out but our users on the obi200 side hear them perfectly. Any ideas on settings to try to optimize in this situation?
I know that it is a network related issue, with high jitter, however I know there must be a fix as other VOIP providers can ride the same network path and have no issues at all.
4) Main reason for not just upgrading is that I am using fushionPBX, and they do not yet support 1.6.2, and also trying to figure out an actual upgrade plan so that I do not loose all the customers voicemail's, VM greetings passwords and such.
Any help is greatly appreciated with what direction I should go.
Thanks!
Josh
On Fri, Nov 13, 2015 at 9:45 AM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote:
We probably need to wipe confluence clean of the incorrect jitter buffer docs now that its been completely rewritten in the past few months.
On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström <Ola.Backstrom@loxysoft.se <mailto:Ola.Backstrom@loxysoft.se>> wrote:
Dragos,
Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
> You can enable the Jitter Buffer in the dial plan: > <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”.
Best regards
/Ola
*From:* freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Dragos Oancea *Sent:* den 12 november 2015 10:14 *To:* freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] Jitterbuffer
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org <mailto:consulting@freeswitch.org>
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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1.6.3 should be rolling out today hopefully. This will be roughly what is in master now. On Friday, November 20, 2015, Dragos Oancea <doancea@orange-vallee.net> wrote:
Hi
Josh, FS master is not 1.6.2 , it's the latest master branch . Get it from git and compile it. 1.6.2 does not have the new JB.
I assume you don't transcode at all and you use G.711 along the path. Try using ILBC between the OBI200 and Freeswitch , so you would do transcoding with Jitter Buffer and PLC enabled. It would be better if you don't transcode at all, but your PSTN provider should offer ILBC or other compressed codec that you should try and you would also need to rely on them for PLC .You should probably enable the Jitter Buffer on the bridge if you do transcoding : <action application="set" data="rtp_jitter_buffer_during_bridge=true"/>
Regards, Dragos
PS: Obihai supports Opus in their phones: http://www.obihai.com/partner/obi1000 .
On 11/20/2015 07:22 AM, Russell Treleaven wrote:
Are you sure the jitter/loss in on the fixed wireless side?
Is the freeswitch<->pstn sip?
Is freeswitch transcoding between the legs?
There is a tool called mtr to measure the loss/jitter along the path.
If freeswitch is not transcoding then it doesn't need a jitter buffer. You could actually use bypass_media to take freeswitch out of the media path and see if that helps. https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview
On Thu, Nov 19, 2015 at 11:07 PM, Josh Richesin <jlrichesin@gmail.com <javascript:_e(%7B%7D,'cvml','jlrichesin@gmail.com');>> wrote:
Hello all!
Thanks for the recommendations on upgrading to FS master, (Assuming that is 1.6.2?) and using the new Jitter Buffer and OPUS codec. Unfortunately, after researching and working with our ATA vendor OBI, they do not support OPUS. So that has me form a few more questions on this issue.
1) Is there any documentation on the new Jitter Buffer being used?
2) Is there any reason to believe that upgrading from our current Free Switch Version 1.4.18 to 1.6.2 would fix the issue with the current codec of g711, even tho I can not use opus?
3) Below is the real issue that I am trying to fix:
obi200------------> fixed wireless --------> FreeSwitch ------PSTN
Call Quality issues are mainly reported by the end user on the PSTN network. Users report that the calls are cutting in and out but our users on the obi200 side hear them perfectly. Any ideas on settings to try to optimize in this situation?
I know that it is a network related issue, with high jitter, however I know there must be a fix as other VOIP providers can ride the same network path and have no issues at all.
4) Main reason for not just upgrading is that I am using fushionPBX, and they do not yet support 1.6.2, and also trying to figure out an actual upgrade plan so that I do not loose all the customers voicemail's, VM greetings passwords and such.
Any help is greatly appreciated with what direction I should go.
Thanks!
Josh
On Fri, Nov 13, 2015 at 9:45 AM, Brian West <brian@freeswitch.org <javascript:_e(%7B%7D,'cvml','brian@freeswitch.org');>> wrote:
We probably need to wipe confluence clean of the incorrect jitter buffer docs now that its been completely rewritten in the past few months.
On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström < Ola.Backstrom@loxysoft.se <javascript:_e(%7B%7D,'cvml','Ola.Backstrom@loxysoft.se');>> wrote:
Dragos,
Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”.
Best regards
/Ola
*From:* freeswitch-users-bounces@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');>] *On Behalf Of *Dragos Oancea *Sent:* den 12 november 2015 10:14 *To:* freeswitch-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','freeswitch-users@lists.freeswitch.org');> *Subject:* Re: [Freeswitch-users] Jitterbuffer
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');>
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users@lists.freeswitch.org');>
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--
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*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
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*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting@freeswitch.org');> http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Thank you for all of the reply's and thoughts! I am sure that the Jitter / Loss is on the fixed wireless side, even tho the QoS is set. I am using some low end ubiquiti devices so I am not convinced that the QoS is working however. Running MTR from the FreeSwitch to the end user bounces between 5ms and 200ms. Same on the return path. Interesting however that most of the users complaints are not noticed by our customer, but buy the person that called them from off net (PSTN) . I would think it to be an obi issue, however we have voip phones, Grandstream and a few Grand stream ATA's that have the same exact issue. When using less jittery network to connect to the FS, everything is fine. For example, customers of ours that are on fiber or using cable modems have not reported any of the issues. Going to the PSTN, we are using SIP trunk to vitelity. The codec that we are using currently, with no trans coding is PCMU/8000. I will try Dragos suggestion of transcoding to ILBC. Thanks! Josh On Fri, Nov 20, 2015 at 7:19 AM, Michael Jerris <mike@jerris.com> wrote:
1.6.3 should be rolling out today hopefully. This will be roughly what is in master now.
On Friday, November 20, 2015, Dragos Oancea <doancea@orange-vallee.net> wrote:
Hi
Josh, FS master is not 1.6.2 , it's the latest master branch . Get it from git and compile it. 1.6.2 does not have the new JB.
I assume you don't transcode at all and you use G.711 along the path. Try using ILBC between the OBI200 and Freeswitch , so you would do transcoding with Jitter Buffer and PLC enabled. It would be better if you don't transcode at all, but your PSTN provider should offer ILBC or other compressed codec that you should try and you would also need to rely on them for PLC .You should probably enable the Jitter Buffer on the bridge if you do transcoding : <action application="set" data="rtp_jitter_buffer_during_bridge=true"/>
Regards, Dragos
PS: Obihai supports Opus in their phones: http://www.obihai.com/partner/obi1000 .
On 11/20/2015 07:22 AM, Russell Treleaven wrote:
Are you sure the jitter/loss in on the fixed wireless side?
Is the freeswitch<->pstn sip?
Is freeswitch transcoding between the legs?
There is a tool called mtr to measure the loss/jitter along the path.
If freeswitch is not transcoding then it doesn't need a jitter buffer. You could actually use bypass_media to take freeswitch out of the media path and see if that helps. https://freeswitch.org/confluence/display/FREESWITCH/Bypass+Media+Overview
On Thu, Nov 19, 2015 at 11:07 PM, Josh Richesin <jlrichesin@gmail.com> wrote:
Hello all!
Thanks for the recommendations on upgrading to FS master, (Assuming that is 1.6.2?) and using the new Jitter Buffer and OPUS codec. Unfortunately, after researching and working with our ATA vendor OBI, they do not support OPUS. So that has me form a few more questions on this issue.
1) Is there any documentation on the new Jitter Buffer being used?
2) Is there any reason to believe that upgrading from our current Free Switch Version 1.4.18 to 1.6.2 would fix the issue with the current codec of g711, even tho I can not use opus?
3) Below is the real issue that I am trying to fix:
obi200------------> fixed wireless --------> FreeSwitch ------PSTN
Call Quality issues are mainly reported by the end user on the PSTN network. Users report that the calls are cutting in and out but our users on the obi200 side hear them perfectly. Any ideas on settings to try to optimize in this situation?
I know that it is a network related issue, with high jitter, however I know there must be a fix as other VOIP providers can ride the same network path and have no issues at all.
4) Main reason for not just upgrading is that I am using fushionPBX, and they do not yet support 1.6.2, and also trying to figure out an actual upgrade plan so that I do not loose all the customers voicemail's, VM greetings passwords and such.
Any help is greatly appreciated with what direction I should go.
Thanks!
Josh
On Fri, Nov 13, 2015 at 9:45 AM, Brian West <brian@freeswitch.org> wrote:
We probably need to wipe confluence clean of the incorrect jitter buffer docs now that its been completely rewritten in the past few months.
On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström < Ola.Backstrom@loxysoft.se> wrote:
Dragos,
Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”.
Best regards
/Ola
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Dragos Oancea *Sent:* den 12 november 2015 10:14 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Jitterbuffer
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
http://confluence.freeswitch.org
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thank you! I will try that out. I will have to migrate over to FS master. Josh On Thu, Nov 12, 2015 at 1:13 AM, Dragos Oancea <doancea@orange-vallee.net> wrote:
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@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 (6)
-
Brian West
-
Dragos Oancea
-
Josh Richesin
-
Michael Jerris
-
Ola Bäckström
-
Russell Treleaven