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

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From: "Andreas Sikkema" <andreas.sikkema@xxxxxxxx>
Date: Wed, 17 May 2006 14:44:27 +0200
Mike,


> 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???

All RTP streams have their own sequence number "ranges", so 
your application / stack needs to make sure they are correct. 
Either you do that by proxying the RTP or by reopening the 
media streams.

I'm not sure if this is on-topic for this list though....

-- 
Andreas Sikkema                       BBned NV
Software Engineer                Planeetbaan 4
+31 (0)23 7074342            2132 HZ Hoofddorp