SIP VoIP calls - lost packets?
"Mike Kennedy" <[email protected]>
| Newsgroups | gmane.network.ethereal.devel |
|---|---|
| Message-ID | <[email protected]> |
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 _______________________________________________________________________ This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. _______________________________________________________________________ _______________________________________________ Ethereal-dev mailing list [email protected] http://www.ethereal.com/mailman/listinfo/ethereal-dev