Hello I think I finally understood the issue/solution, but I'd like to make sure I'm correct:
- In Diana Cionoiu's famous article on Freshmeat (http://freshmeat.net/articles/nat-traversal-for-the-sip-protocol), regardless of whether SIP end-points use a public IP or are behind a NAT, RTP packets flow directly between the two SIP end-points because the SIP server only acts... as an SIP server, meaning it only acts as a registrar (for SIP end-points to make themselves know with an IP + RTP ports), and then as a Central office (to ring the other SIP end-point, and close the connection when an SIP end-point decides to hangup) - OTOH, for IP PBX's like Asterisk to provide PBX services (eg. call transfer, call parking, etc.), it must remain in the loop, and hence, by default (canreinvite=no), all RTP packets always go through Asterisk, even if both SIP end-points live in the same network as the Asterisk server (and hence, since NAT is not involved, there's no need for any kung-fu with rewriting information in SDP packets and asking the NAT box to open the relevant ports for RTP) Is this correct? Thank you. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
