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