Ethereal-dev: [Ethereal-dev] SIP VoIP calls - lost packets?

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From: "Mike Kennedy" <Mike.Kennedy@xxxxxxxxxxxxxxx>
Date: Wed, 17 May 2006 12:19:01 +0100

Hi all

 

I am writing a SIP protocol stack and am having a few issues with conference calls.

If I have 1 call on hold and another in progress and then join them all together I seem to get a file that sounds all wrong and distorted.

 

Using ethereal and looking at the RTP stream analysis, one part of the conversation has 95%+ lost packets.  Looking at the individual packets it seems that upon the joining of the 3 calls the sequence number jumps up quite a large amount and corrupts(?) that stream???

 

Has anyone else noticed this??

 

Cheers

Mike


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