[Freeswitch-users] Audio delays and jitter

[Danny Gershman]

We are using the OPUS codec with Chrome and WebRTC to make SIP calls into
FS conferences.  We have packet captures at the edge of network that sound
clear, but when they get to FreeSWITCH they sound distorted via delays and
clipping.

I've tried several combinations of jitterbuffer settings, disabled/enabling
RTP timers, conference.  I have a mechanism I use in which I replay the RTP
packets using SIPP and try them out.  Is there a way to dynamically adjust
the jitterbuffer so that the right size is being used?  Right now I have it
set at 180:540:540.  Maybe there is some other setting I could try that
would help?

Perhaps there's a way to force OPUS to use less bandwidth at the expense of
quality?

I'm willing to share the PCAPs some examples.

Thanks in advance.

--Danny Gershman
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