Thanks for the tip Ranga! I will continue using my SIP microscope!

Sven

-----Original Message-----
From: M. Ranganathan [mailto:mra...@gmail.com] 
Sent: 07 July 2010 16:54
To: Sven Evensen
Cc: Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] RTP being buffer in sipX, causing delayed
audio.

SipXrelay will not just relay whatever media is sent to it. It will
only relay media after it knows where to send the media. Please
examine the signaling carefully using sipviewer and take note of what
point in the processing this can be asserted. Look at the signaling
for the sip call setup.


Nothing is being buffered but packets can be dropped.

( It is highly recommended that you switch 4.2 )

Ranga


On Wed, Jul 7, 2010 at 11:43 AM, Sven Evensen <sven.even...@onrelay.com>
wrote:
> Thanks Tony, two of my cases are running EC2, the third one is on a
high
> spec'ed server, so I am sure it is not down to that. The "buffered"
> media is released when 180 Ringing is received, dead certain. :)
>
> Sven
>
> -----Original Message-----
> From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> Sent: 07 July 2010 14:38
> To: Sven Evensen; mra...@gmail.com
> Cc: sipx-users@list.sipfoundry.org
> Subject: Re: [sipx-users] RTP being buffer in sipX, causing delayed
> audio.
>
> Er... Just keep in mind that if this running as a virtual
> somewhere/somehow
> (EC2 included), that latency (storage method, etc.) and media might
> result
> in what you are seeing.
>
> You might find different results by using 4.2 in the same basic
> configuration.
> ============================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
>
> Email: tgrazi...@myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> ----- Original Message -----
> From: sipx-users-boun...@list.sipfoundry.org
> <sipx-users-boun...@list.sipfoundry.org>
> To: M. Ranganathan <mra...@gmail.com>
> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org>
> Sent: Wed Jul 07 09:34:28 2010
> Subject: Re: [sipx-users] RTP being buffer in sipX, causing delayed
> audio.
>
> 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
>> OnRelay
>> Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) 207 902
> 8123
>> |
>> sven.even...@onrelay.com | www.onrelay.com
>>
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>
>
>
> --
> M. Ranganathan
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-- 
M. Ranganathan
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