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.
@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. Regards, Treuz _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
