[Freeswitch-users] jitters generated from freeswitch to trunk in conference call

Dragos Oancea dragos at freeswitch.org
Wed Oct 6 10:25:25 UTC 2021

Make sure you're using a timer.
eg: <param name="rtp-timer-name" value="soft"/>

"jitter buffers generated from freeswitch to trunk" - does this mean you
noticed jitter or you enabled the jitter buffer in FS ?
If you don't use the jitter buffer, start using it.

On Wed, Oct 6, 2021 at 5:40 AM Achintha <achinthau at gmail.com> wrote:

> Hi all,
> I configured 2 freeswitch servers (FreeSWITCH version:
> 1.10.5-release-17-25569c1631~64bit) on Debian 10.4 as media servers. one
> opensips server is located in front of the freeswitch servers and opensips
> act as SIP load balancer. media directly connect with sip trunk and
> freeswitch.We use dynamic dialplan to generate calls and use g711.
> Anyway i generate below mentioned scenario
>    1. registered as extension
>    2. dial outbound to mobile call through the sip trunk and talk
>    3. put hold the first call
>    4. dial another outbound call to another mobile through the same sip
> trunk and talk.
>    5. connect both calls (conference).
> in the 5th step jitter buffers generated from freeswitch to trunk. voice
> not clear.
> I have changed codecs and tests (opus,G711a,G711u)
> but the issue is not fixed. how to solve this problem.
> Best Regards..
> Achintha Udukumbura
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