RE: SIP VoIP calls - lost packets?
"Andreas Sikkema" <[email protected]>
| Newsgroups | gmane.network.ethereal.devel |
|---|---|
| Message-ID | <5B97FFB7F28E3E4294B049C968CBF47B01F8D097@SERVER-EX02A.INTERN.BBNED.NL> |
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