[EMAIL PROTECTED] wrote:
>    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
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>
>   
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

Treuz
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