Hi Ranga,

I am positive that RTP is being buffered (in loss for better word). I clearly 
see rtp coming into port 30508 (as seen in my example) and being "buffered. 
This happens consequently until 180 Ringing comes x seconds later. Then and 
only then are the "buffered" rtp packets relayed to the conference. The X 
seconds corresponds to the time between INVITE going out and 180 Ringing coming 
in.

Can you please give me some clues where to look. Is port 30508 a sipXrelay 
port? And port 12536 a conference bridge port?

Which logs will give me a clue to this problem?

Regards,
Sven

 

-----Original Message-----
From: M. Ranganathan [mailto:mra...@gmail.com] 
Sent: 07 July 2010 02:58
To: Sven Evensen
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] RTP being buffer in sipX, causing delayed audio.

On Tue, Jul 6, 2010 at 10:21 AM, Sven Evensen <sven.even...@onrelay.com> wrote:
> Our product, which sits kind of on top of sipX, generates calls, redirects
> calls etc.
>
>
>
>
> So my question is : Why does RTP get buffered in sipX?


The answer is - it is not.



There is no buffer. If it is being delayed, that is for a different
reason and that is something to investigate independently. SipXrelay
does not buffer packets.





>
> And sorry, I do not know whether to send pcaps or snapshots...


Neither.

You probably have a problem elsewhere. Could be anything at all from
process scheduling to overload.


Ranga


>
>
>
> Regards,
>
> Sven
>
>
>
> Sven Evensen, Technical Support Consultant
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-- 
M. Ranganathan
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