ption of the 183, the caller's UAC begins receiving the RTP packets sent from the gateway and presents the audio to the caller SO they know that the call is progressing in the PSTN. The call completes when the called party answers the telephone, which causes the telephone switch to send an Answer Message (ANM) to the gateway. The gateway then cuts the PSTN audio connection through in both directions and sends a 200 OK response to the caller. Because the RTP media path is already established, the gateway echoes the SDP in the 183 but causes no changes to the RTP connection. The UAC sends an ACK to complete the SIP signaling exchange. Because there is no equivalent message in ISUP, the gateway absorbs the ACK. The call terminates when the caller sends the BYE to the gateway. The gateway maps the BYE to the ISUP Release message or REL. The gateway sends the 200 OK to the BYE and receives a RLC from the PSTN. These two mes- sages have no dependency on each other; if, for some reason, either the SIP or PSTN network does not respond properly, one does not want resources held in the other network as a result. INVITE sip: :+12025551313@gw.carrier.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 8.19.19.06:5060;branch=z9hG4bK4545 Max-Forwards: 70 From: <sip:filo.farnsworth@television.tv>;ta To: sip:+12025551313@gw.carrier.com;user=phone> Call-ID: 49235243082018498@television.tv CSeq: 1 INVITE Supported: 100rel Contact: sip:filo.farnsworth@studio.television.tv Content-Type: application/sdp Content-Length: 154 o=FF 2890844535 2890844535 IN IP4 8.19.19.06 t=0 0 C=IN IP4 8.19.19.06 m=audio 5004 RTP/AVP 0 8 <Two alternative codecs, a=rtpmap: 0 PCMU/8000 PCM u-Law or a=rtpmap 8 PCMA/8000 PCM A-Law SIP: Understanding the Session Initiation Protocol CdPN=202-555-1313, NPI=E.164, NOA=National <Gateway maps telephone into called party number SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 8.19.19.06:5060;branch=z9hG4bK4545 From: <sip:filo.farnsworth@television.tv>;tag=12 To: +12025551313@gw.carrier.com;user=phone>;tag=37 =Tag and brackets Cal