[Freeswitch-users] Audio delays and jitter
danny.gershman at gmail.com
Thu Oct 1 01:54:10 MSD 2015
Currently tested with 1.4.19
On Wed, Sep 30, 2015 at 5:23 PM Brian West <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>
>> 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
>> 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
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> Official FreeSWITCH Sites
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
> *Brian West*
> brian at freeswitch.org
> *Twitter: @FreeSWITCH , @briankwest*
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