Our product, which sits kind of on top of sipX, generates calls,
redirects calls etc.

 

This issue is for sipX 4.0.4 using sipXBridge and the fault is seen on
several

different ITSPs

 

In this scenario we have an external call redirected out to an other
external party.

We then want to add a party, so the two original call legs are moved
into a conference

bridge and the third leg (external) is added. This is where the problem
starts.

This last INVITE goes out and RTP starts coming in immediately. This RTP
stream

gets buffered in sipX (sipXBridge?) until the 180 Ringing arrives. Then
sipXBridge

starts sending the buffered RTP into the conference where it is sent out
with no delay.

The ring tone coming as result of 180 is delayed until all the buffered
packets are sent to

the conference bridge. Then the voice is delayed when the ring tone
packets are finished

(after 200 OK).

 

This is from WireShark, the RTP packet comes in from SIP trunk
(123.102.11.15) and is relayed to conf bridge 3 seconds later.

 

8650     38.547256         123.102.11.15   40036   192.168.0.10
30508   RTP     PT=ITU-T G.711 PCMA, SSRC=0x8D56FB95, Seq=35378,
Time=200

 

10722   41.856764         192.168.0.10     30510   192.168.0.10
12536   RTP     PT=ITU-T G.711 PCMA, SSRC=0x8D56FB95, Seq=35378,
Time=200

 

So my question is : Why does RTP get buffered in sipX?

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

 

Regards,

Sven

 

Sven Evensen, Technical Support Consultant
OnRelay
Elizabeth House | 39 York Road, London SE1 7NQ, UK | +44 (0) 207 902
8123 | [email protected] | <http://www.onrelay.com/>
www.onrelay.com <http://www.onrelay.com/> 

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