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.
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Date: Wed, 09 Apr 2008 00:42:03 +0200
From: Andrea <[EMAIL PROTECTED]>
Subject: Re: [Sip-implementors] issue with mid-call mobility, no new
INVITE message
To: [email protected]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
[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
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