[Freeswitch-users] Timing issues with Verto, SIP client works fine
Support from NetworkedAudio LLC
support at naud.io
Wed Jun 10 19:35:01 UTC 2020
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.
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the FreeSWITCH-users