Hi Hitesh,

It appears that the far end GW detects media inactivity and drops the
call. In case you know the new IP address simply send a re-INVITE with
the information. This would keep the call going. 

regards
Rayees
 

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hitesh Tewari
Sent: Tuesday, September 04, 2007 3:49 PM
To: [email protected]
Subject: [Sip-implementors] reINVITE

Hi,

I have a query on a very specific problem. We have a scenario where a
SIP phone is located behind a NAT and wishes to communicate with a PSTN
phone. We have a client on the phone that opens up the network for a
specific amount of time (e.g. 3 minutes) after which the client has to
re-login for an additional three minute period and so on. If a user has
initiated a SIP call and a re-login occurs  we are assigned new SIP and
RTP ports. This has the effect of breaking the call.

My question is whether there is any mechanism in SIP to convey these new
settings to the PSTN gateway in order to preserve the call. I have seen
messages on this forum indicating that the reINVITE message allows one
to change the IP and Port  numbers, but not sure if this is enough for
our specific requirements.

Your comments and suggestion would be highly appreciated. Thanks.

Hitesh
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