[Freeswitch-users] Audio delays and jitter

[Brian West]

Which revision of FreeSWITCH have you tested with?

On Wed, Sep 30, 2015 at 4:15 PM, Danny Gershman <danny.gershman at gmail.com>
wrote:

> 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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-- 

*Brian West*
brian at freeswitch.org


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