[Freeswitch-users] Audio delays and jitter

[Ken Rice]

You should really be testing the latest… not to mention there are multiple bugs resolved since 1.4.19

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Danny Gershman
Sent: Wednesday, September 30, 2015 4:54 PM
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Audio delays and jitter

 

Currently tested with 1.4.19

 

On Wed, Sep 30, 2015 at 5:23 PM Brian West <brian at freeswitch.org <mailto:brian at freeswitch.org> > wrote:

Which revision of FreeSWITCH have you tested with?

 

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