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<div dir="ltr">Windows Server 2016.</div>

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<div id="divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif"><b>From:</b> FreeSWITCH-users <freeswitch-users-bounces@lists.freeswitch.org> on behalf of Gregor Nanger <gregor@infomedia.si><br>

<b>Sent:</b> Thursday, June 18, 2020 12:22 PM<br>

<b>To:</b> FreeSWITCH Users Help<br>

<b>Subject:</b> Re: [Freeswitch-users] Timing issues with Verto, SIP client works fine

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<div dir="auto">On which OS is FS installed?</div>

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<div dir="ltr" class="gmail_attr">On Thu, 18 Jun 2020, 18:59 Support from NetworkedAudio LLC, <<a href="mailto:support@naud.io">support@naud.io</a>> wrote:<br>

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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.</div>

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Tests:</div>

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Soft phone (MicroSIP) > Conference > Verto (on Chrome) - sounds great.</div>

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Verto (on Chrome) > Conference > Verto (on Chrome) - sounds good, some crunchiness, some drops.</div>

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Soft phone (MicroSIP) > PortAudio > external system > PortAudio > Verto (on Chrome) sounds pretty good.</div>

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Verto (on Chrome) <span style="font-family:Calibri,Arial,Helvetica,sans-serif; background-color:rgb(255,255,255); display:inline!important">> PortAudio > external system > PortAudio > Verto (on Chrome) sounds quite poor, lost words.</span></div>

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<span style="font-family:Calibri,Arial,Helvetica,sans-serif; background-color:rgb(255,255,255); display:inline!important">External system is 48KHz and PortAudio > external system > PortAudio sounds perfect.</span></div>

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<span style="font-family:Calibri,Arial,Helvetica,sans-serif; background-color:rgb(255,255,255); display:inline!important">Recordings of the above are at <a href="https://bit.ly/3EFFyb0" target="_blank" rel="noreferrer">https://bit.ly/3EFFyb0</a></span></div>

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Jitter buffers aren't making any difference and the timer_test shows good numbers. All are using the Opus CODEC.</div>

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Any other ideas, please ?</div>

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<div id="m_524414677411593440divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>From:</b> Support from NetworkedAudio LLC<br>

<b>Sent:</b> Wednesday, June 10, 2020 2:35 PM<br>

<b>To:</b> FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank" rel="noreferrer">freeswitch-users@lists.freeswitch.org</a>><br>

<b>Subject:</b> Timing issues with Verto, SIP client works fine</font>

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Hi all,</div>

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We're using VERTO in an intranet application.<br>

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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"). </div>

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This happens with MS Edge (worse) and Chrome (better), with PCMU and OPUS.</div>

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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.</div>

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Using exactly the same computers and headsets but using MicroSIP, everything is perfect.</div>

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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.</div>

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Where should I look now? Have tried 1.10 but no change.</div>

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