From: Treuz <[EMAIL PROTECTED]> hello all, for first sorry for my english. I have an issue with SIP "mid-call mobility" scenario. I'm using Asterisk (that act as sip server) and 2 softphone clients, all in the same subnetwork. I make a call from softphone A to B and in the middle of the call the client B moves to another subnetwork. The client B then obtain a new IP address via DHCP in the new subnetwork and take back the call with client A. The problem is that i got no SIP signaling nor any new INVITE message (as stated in this document: http://www.cs.columbia.edu/~hgs/papers/Wedl9908_Mobility.pdf in section 3.2) from client B to inform A about it's new IP address, but the RTP flow is directed to the correct host so the call can continue. In addition, after B moves and A hangup the call, the "BYE" message is directed to the old IP address of B. I tried several softphone client, but no luck to see that new INVITE message. How is possible that A sends RTP correctly to new-B if he don't know yet the new IP address? Don't know where is the problem
Have you traced the SIP messages between A and Asterisk, and between Asterisk and B? Asterisk functions as a B2BUA, and all the signaling passes through it. So if the signalling is not working, it is likely that Asterisk is involved. Dale _______________________________________________ Sip-implementors mailing list [email protected] https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
