Timing issues with Verto, SIP client works fine
Hi all, We're using VERTO in an intranet application. If we do anything other than an echo test, we get dropouts on bridge (to SIP phone or PortAudio), which either drops whole sounds (count 1,2,4,5,7,8) or slurs ("se-even"). This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS. Turning off audio processing (googEcho...) has no effect. Changing RTP_TIMER from SOFT to NONE seems to make the dropout more random. It's not a network bandwidth issue - this can happen across a gigabit link. Using exactly the same computers and headsets but using MicroSIP, everything is perfect. Have tried jitterbuffers, but they don't help (as it's not a delay issue) - it really feels like a timing problem going from Verto into the rest of the system. Where should I look now? Have tried 1.10 but no change.
Have done some more testing - the issue appears to be that packets are dropped or a timing resync happens when connecting to PortAudio through the dialplan. Tests: Soft phone (MicroSIP) > Conference > Verto (on Chrome) - sounds great. Verto (on Chrome) > Conference > Verto (on Chrome) - sounds good, some crunchiness, some drops. Soft phone (MicroSIP) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds pretty good. Verto (on Chrome) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds quite poor, lost words. External system is 48KHz and PortAudio > external system > PortAudio sounds perfect. Recordings of the above are at https://bit.ly/3EFFyb0 Jitter buffers aren't making any difference and the timer_test shows good numbers. All are using the Opus CODEC. Any other ideas, please ? ________________________________ From: Support from NetworkedAudio LLC Sent: Wednesday, June 10, 2020 2:35 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Timing issues with Verto, SIP client works fine Hi all, We're using VERTO in an intranet application. If we do anything other than an echo test, we get dropouts on bridge (to SIP phone or PortAudio), which either drops whole sounds (count 1,2,4,5,7,8) or slurs ("se-even"). This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS. Turning off audio processing (googEcho...) has no effect. Changing RTP_TIMER from SOFT to NONE seems to make the dropout more random. It's not a network bandwidth issue - this can happen across a gigabit link. Using exactly the same computers and headsets but using MicroSIP, everything is perfect. Have tried jitterbuffers, but they don't help (as it's not a delay issue) - it really feels like a timing problem going from Verto into the rest of the system. Where should I look now? Have tried 1.10 but no change.
On which OS is FS installed? On Thu, 18 Jun 2020, 18:59 Support from NetworkedAudio LLC, <support@naud.io> wrote:
Have done some more testing - the issue appears to be that packets are dropped or a timing resync happens when connecting to PortAudio through the dialplan.
Tests:
Soft phone (MicroSIP) > Conference > Verto (on Chrome) - sounds great. Verto (on Chrome) > Conference > Verto (on Chrome) - sounds good, some crunchiness, some drops. Soft phone (MicroSIP) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds pretty good. Verto (on Chrome) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds quite poor, lost words.
External system is 48KHz and PortAudio > external system > PortAudio sounds perfect.
Recordings of the above are at https://bit.ly/3EFFyb0
Jitter buffers aren't making any difference and the timer_test shows good numbers. All are using the Opus CODEC.
Any other ideas, please ?
------------------------------ *From:* Support from NetworkedAudio LLC *Sent:* Wednesday, June 10, 2020 2:35 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Timing issues with Verto, SIP client works fine
Hi all,
We're using VERTO in an intranet application.
If we do anything other than an echo test, we get dropouts on bridge (to SIP phone or PortAudio), which either drops whole sounds (count 1,2,4,5,7,8) or slurs ("se-even").
This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS.
Turning off audio processing (googEcho...) has no effect. Changing RTP_TIMER from SOFT to NONE seems to make the dropout more random. It's not a network bandwidth issue - this can happen across a gigabit link.
Using exactly the same computers and headsets but using MicroSIP, everything is perfect.
Have tried jitterbuffers, but they don't help (as it's not a delay issue) - it really feels like a timing problem going from Verto into the rest of the system.
Where should I look now? Have tried 1.10 but no change. _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
Windows Server 2016. ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Gregor Nanger <gregor@infomedia.si> Sent: Thursday, June 18, 2020 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues with Verto, SIP client works fine On which OS is FS installed? On Thu, 18 Jun 2020, 18:59 Support from NetworkedAudio LLC, <support@naud.io<mailto:support@naud.io>> wrote: Have done some more testing - the issue appears to be that packets are dropped or a timing resync happens when connecting to PortAudio through the dialplan. Tests: Soft phone (MicroSIP) > Conference > Verto (on Chrome) - sounds great. Verto (on Chrome) > Conference > Verto (on Chrome) - sounds good, some crunchiness, some drops. Soft phone (MicroSIP) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds pretty good. Verto (on Chrome) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds quite poor, lost words. External system is 48KHz and PortAudio > external system > PortAudio sounds perfect. Recordings of the above are at https://bit.ly/3EFFyb0 Jitter buffers aren't making any difference and the timer_test shows good numbers. All are using the Opus CODEC. Any other ideas, please ? ________________________________ From: Support from NetworkedAudio LLC Sent: Wednesday, June 10, 2020 2:35 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Timing issues with Verto, SIP client works fine Hi all, We're using VERTO in an intranet application. If we do anything other than an echo test, we get dropouts on bridge (to SIP phone or PortAudio), which either drops whole sounds (count 1,2,4,5,7,8) or slurs ("se-even"). This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS. Turning off audio processing (googEcho...) has no effect. Changing RTP_TIMER from SOFT to NONE seems to make the dropout more random. It's not a network bandwidth issue - this can happen across a gigabit link. Using exactly the same computers and headsets but using MicroSIP, everything is perfect. Have tried jitterbuffers, but they don't help (as it's not a delay issue) - it really feels like a timing problem going from Verto into the rest of the system. Where should I look now? Have tried 1.10 but no change. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com
Try set <param name="timer-name" value="none"/> in verto.config On Thu, 18 Jun 2020 at 22:43, Support from NetworkedAudio LLC < support@naud.io> wrote:
Windows Server 2016. ------------------------------ *From:* FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Gregor Nanger <gregor@infomedia.si> *Sent:* Thursday, June 18, 2020 12:22 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Timing issues with Verto, SIP client works fine
On which OS is FS installed?
On Thu, 18 Jun 2020, 18:59 Support from NetworkedAudio LLC, < support@naud.io> wrote:
Have done some more testing - the issue appears to be that packets are dropped or a timing resync happens when connecting to PortAudio through the dialplan.
Tests:
Soft phone (MicroSIP) > Conference > Verto (on Chrome) - sounds great. Verto (on Chrome) > Conference > Verto (on Chrome) - sounds good, some crunchiness, some drops. Soft phone (MicroSIP) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds pretty good. Verto (on Chrome) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds quite poor, lost words.
External system is 48KHz and PortAudio > external system > PortAudio sounds perfect.
Recordings of the above are at https://bit.ly/3EFFyb0
Jitter buffers aren't making any difference and the timer_test shows good numbers. All are using the Opus CODEC.
Any other ideas, please ?
------------------------------ *From:* Support from NetworkedAudio LLC *Sent:* Wednesday, June 10, 2020 2:35 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Timing issues with Verto, SIP client works fine
Hi all,
We're using VERTO in an intranet application.
If we do anything other than an echo test, we get dropouts on bridge (to SIP phone or PortAudio), which either drops whole sounds (count 1,2,4,5,7,8) or slurs ("se-even").
This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS.
Turning off audio processing (googEcho...) has no effect. Changing RTP_TIMER from SOFT to NONE seems to make the dropout more random. It's not a network bandwidth issue - this can happen across a gigabit link.
Using exactly the same computers and headsets but using MicroSIP, everything is perfect.
Have tried jitterbuffers, but they don't help (as it's not a delay issue) - it really feels like a timing problem going from Verto into the rest of the system.
Where should I look now? Have tried 1.10 but no change. _________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
-- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si
This helped a great deal. Thank you. ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Gregor Nanger <gregor@infomedia.si> Sent: Thursday, June 18, 2020 4:37 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues with Verto, SIP client works fine Try set <param name="timer-name" value="none"/> in verto.config On Thu, 18 Jun 2020 at 22:43, Support from NetworkedAudio LLC <support@naud.io<mailto:support@naud.io>> wrote: Windows Server 2016. ________________________________ From: FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> on behalf of Gregor Nanger <gregor@infomedia.si<mailto:gregor@infomedia.si>> Sent: Thursday, June 18, 2020 12:22 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Timing issues with Verto, SIP client works fine On which OS is FS installed? On Thu, 18 Jun 2020, 18:59 Support from NetworkedAudio LLC, <support@naud.io<mailto:support@naud.io>> wrote: Have done some more testing - the issue appears to be that packets are dropped or a timing resync happens when connecting to PortAudio through the dialplan. Tests: Soft phone (MicroSIP) > Conference > Verto (on Chrome) - sounds great. Verto (on Chrome) > Conference > Verto (on Chrome) - sounds good, some crunchiness, some drops. Soft phone (MicroSIP) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds pretty good. Verto (on Chrome) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds quite poor, lost words. External system is 48KHz and PortAudio > external system > PortAudio sounds perfect. Recordings of the above are at https://bit.ly/3EFFyb0 Jitter buffers aren't making any difference and the timer_test shows good numbers. All are using the Opus CODEC. Any other ideas, please ? ________________________________ From: Support from NetworkedAudio LLC Sent: Wednesday, June 10, 2020 2:35 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Timing issues with Verto, SIP client works fine Hi all, We're using VERTO in an intranet application. If we do anything other than an echo test, we get dropouts on bridge (to SIP phone or PortAudio), which either drops whole sounds (count 1,2,4,5,7,8) or slurs ("se-even"). This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS. Turning off audio processing (googEcho...) has no effect. Changing RTP_TIMER from SOFT to NONE seems to make the dropout more random. It's not a network bandwidth issue - this can happen across a gigabit link. Using exactly the same computers and headsets but using MicroSIP, everything is perfect. Have tried jitterbuffers, but they don't help (as it's not a delay issue) - it really feels like a timing problem going from Verto into the rest of the system. Where should I look now? Have tried 1.10 but no change. _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com _________________________________________________________________________ The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform. Join our online community to chat in real time https://signalwire.community Professional FreeSWITCH Services sales@freeswitch.com<mailto:sales@freeswitch.com> https://freeswitch.com Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://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 https://freeswitch.com -- Gregor Nanger CTO t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si<http://www.infomedia.si>
participants (2)
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Gregor Nanger
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Support from NetworkedAudio LLC