[Freeswitch-users] Audio delays and jitter

[Danny Gershman]

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>
> 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
>
>
> *Twitter: @FreeSWITCH , @briankwest*
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>
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>
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