I guess you are talking about latency issues caused by the jitter buffer.
Does the problem go away when you put this 

<action application="set" data="sip_jitter_buffer_during_bridge=false"/>

in the dialplan?


On Tue, Nov 19, 2013 at 5:32 PM, I put the Who? in Mishehu <mishehu@freeswitch.org> wrote:
I do not have this problem, and I use FS for years.  So this is not normal or expected behavior.  But that being said,  you provided no information about your set up - such as distro, FS version, what type of system you were running on (virtual or bare metal), and what kind of network you were running this over.

I can only guess that you're running on a virtual server and are experiencing some sort of clock skew, but that's just a guess.

-- 
Yossi Neiman


On 11/19/2013 02:28 PM, Ashwin Jain wrote:
Hi,

We have setup a single freeswitch server which receives a call from a DID provider (ex: voipms) and do a outbound call another provider (ex: lesnet) (we also record call). 

For short duration call, everything is working fine, but for calls with long duration (like more than 10 minutes), audio is getting out of sync.

Has anyone had this issue? Where could be the problem? 

--
Thanks and Regards,
Ashwin Jain


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_________________________________________________________________________
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