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. https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer On Wed, Oct 6, 2021 at 5:40 AM Achintha <achinthau@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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