rting the extension would send RTCP packets to this IP address and port number instead of using the RTP IP address (connection line) and port number plus one. If just the port number needs to be changed, the address information can be omitted: a=rtcp: 53022 Note that without this SDP extension, the SIP and RTP sessions will still work, but no RTCP would be exchanged. References Rosenberg, J., et al., "SIP: Session Initiation Protocol," RFC 3261, 2002. Rosenberg, J., "A Presence Event Package for the Session Initiation Protocol (SIP)," IETF Internet-Draft, Work in Progress, January 2003. Roach, A., "Session Initiation Protocol (SIP)-Specific Event Notification," RFC 3265, 2002. Campbell, B., et al., "SIMPLE Presence Publication Mechanism," IETF Internet-Draft, Work in Progress, February 2003. Schulzrinne, H., and C. Agboh, "Session Initiation Protocol (SIP)-H.323 Interworking Requirements," IETF Internet-Draft, Work in Progress, February 2003. Rosenberg, J., H. Salama, and M. Squire, "Telephony Routing over IP (TRIP)," RFC 3219, 2002. Bangalore, M., et al., "A Telephony Gateway REgistration Protocol (TGREP)," IETF Internet-Draft, February 2003. SIP Clients and Servers [8] Donovan, S., and J. Rosenberg, "The SIP Session Timer," IETF Internet-Draft, Work in Progress. [9] Franks, J., et al., "HTTP Authentication: Basic and Digest Access Authentication," RFC 2617, 1999. [10] Ramsdell, B., "S/MIME Version 3 Message Specification," RFC 2633, 1999. [11] Rosenberg, J., R. Mahy, and S. Sen, "NAT and Firewall Scenarios and Solutions for SIP," IETF Internet-Draft, Work in Progress, March 2003. [12] Rosenberg, J., et al., "STUN-Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)," RFC 3489, 2003. [13] Rosenberg, J., et al., "Traversal Using Relay NAT (TURN)," IETF Internet-Draft, Work in Progress, March 2003. [14] Yon, D., "Connection-Oriented Media Transport in SDP," IETF Internet-Draft, Work in Progress, May 2002. Rosenberg, J., J. Weinberger, and H. Schulzrinne, "An Extension to the 