> > On Wed, Apr 9, 2008 at 7:05 AM, Andrea <[EMAIL PROTECTED]> wrote: > > > Yu Marilyn-Q12239 wrote: > > > > Look at your RTP stream in the trace. Asterisk is not only a SIP > B2BUA, > > > > it is a RTP B2B too. It doesn't need to let B know A's address change > to > > > > forward RTP. > > > > > > > > > > > [EMAIL PROTECTED] wrote: > > > > From: Andrea <[EMAIL PROTECTED]> > > > > > > > > Yes i traced the whole session. you can get the file here: > > > > http://www.fileden.com/files/2008/3/26/1837579/test1.pcap > > > > Asterisk is 49.8 > > > > Client A is 47.103 -> 49.115 > > > > Client B is 49.116 > > > > The only thing i dont traced is that if B hangup the call, asterisk > > > > forward "BYE" message to 47.103 (the old address of A) > > > > Thanks in advance > > > > > > > > It sounds like Asterisk is sending the BYE incorrectly. > > > > > > > > Dale > > > > > > > @Mey.Yu: yes i agree with that, but Asterisk should know the new > > > location of mobile host so he can sends SIP signaling correctly to new > > > address of mobile host. I was only commenting at the question of how RTP being send to the correct address. > > > > > > @Dale: yes it's true, Asterisk should send BYE and all the signaling to > > > new address of mobile host, but mobile host should send new INVITE > > > message to fixed host when he change subnet, so it's true that there is > > > a problem in Asterisk but there is a problem in the softphone clients > > > too, in my opinion. > > > > > > I tried the same scenario with OpenSER that not act as RTP/SIP B2BUA and > > > it happens the same things: when the mobile moves, the clients sends RTP > > > correctly to eachother, but no new INVITE from mobile host in new > > > subnet, and still wrong signaling from OpenSER (when the fixed host > > > hangup the call, BYE is sended to old address of mobile) > > > If u want me to post wireshark trace for OpenSER scenario, just let me > know. In the posted trace, REGISTER for the new contact address only happened after RTP has been redirected. New contact only taking effect for the next incoming new termination request. The BYE was send to the old A contact from Asterisk is correct as everyone says, because A never did update Asterisk with the new sip contact for the on going session.
No free lunch rule still stands I think. If you can send your trace, it should give some clue as how the RTP is being redirected without a RTP B2B and no SDP update. YuMei > > > Regards, > > > > > > Treuz > > > > > > _______________________________________________ > > > Sip-implementors mailing list > > > [email protected] > > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > > _______________________________________________ > > Sip-implementors mailing list > > [email protected] > > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors > > > > _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
