[Freeswitch-users] Audio out of sync
Russell Treleaven
rtreleaven at bunnykick.ca
Wed Nov 20 02:04:36 MSK 2013
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 at 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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