Vivek -- You're talking about symmetrical RTP. There's no fundamental requirement for RTP to go to and from the same set of IPs; there could be four different IP addresses involved in a single ordinary two-way media session, and that's perfectly valid.
However, in practice, many devices DO use the same IP address and port for both sending and receiving media. And some designs require this; for example, it's typical for hosted-IP-PBX service providers to require supported customer-premise device endpoints to send and receive on the same IP address and port. That limitation is useful for implementing security policies (e.g., using "latching" on the RTP streams in their SBC). The SIPConnect folks like symmetric signaling and media for this reason. There are common cases where the IP addresses may not match; for example, if a media server is used to play ringback music until the actual PSTN called party is connected. You may receive RTP from the media server for a while, and then start receiving RTP from the PSTN gateway. If you're building a new application, it would be good if you don't require symmetric media since the standards also don't require it. If you build in a restriction, you'll also be restricting the cases where your application can be used. On Jul 30, 2009, at 8:11 AM, Vivek Batra wrote: > > During SIP signaling, when SDP offer/ answer completes, calling SIP > device > starts sending RTP to the IP:Port recieved in SDP answer from called > party. > Calling SIP device opens its UDP port (for RTP) sent in the SDP offer. > Now, should calling SIP device listens for RTP packets on its UDP port > (which its advertise in SDP offer) from any IP Address, OR > Should calling SIP device listens for RTP packets on its UDP port > only from > IP Address recieved in SDP answer from called party? > > Does this makes sense to restrict SIP device to listen for RTP > packets on > its UDP port only from IP Address recieved in SDP answer from called > party > (not assumed any NAT scenerio :)) > > Actually what happens in absence of this restriction, my SIP client > start > mixing the RTP packets recieved from any IP address! > > Any suggestion?? > > Best Regards, > Vivek Batra > _______________________________________________ > Sip-implementors mailing list > Sip-implementors@lists.cs.columbia.edu > https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors Mark R Lindsey lind...@e-c-group.com http://e-c-group.com/~lindsey +12293160013 _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
