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

Achintha achinthau at gmail.com
Wed Oct 6 02:39:28 UTC 2021

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20211006/13e727ed/attachment.html>

More information about the FreeSWITCH-users mailing list