org CSeq: 1 ACK Content-Length: 0 ACK sip:schockley@transistor.com: SIP/2.0 Via: SIP/2.0/UDP 9.8.7.6:5060;branch=z9hG4bk5. Via: SIP/2.0/UDP discrete.sampling.org:5060;branch=z9hG4bK5654 Max-Forwards 69 From: Shannon <sip:shannon@sampling.org>;tag=cgdf4 To: Schockley <sip:shockley@transistor.com>;tag=1 Call-ID: adf8gasdd7f1d@discrete.sampling.org SIP: Understanding the Session Initiation Protocol CSeq: 1 ACK Content-Length: : 0 10.3 SIP to PSTN Call Through Gateway In the example shown in Figure 10.3, the calling SIP phone places a telephone call to the PSTN through a PSTN gateway. The SIP phone collects the dialed digits and puts them into a SIP URI used in the Request-URI and the To header. The caller may have dialed either the globalized phone number 1-202-555-1313 or they may have just dialed a local number 555-1313, and the SIP phone added the assumed country code and area code to produce the globalized URI. The SIP phone has been preconfigured with the IP address of caller PSTN gateway telephone Telephone 50.60.70.80 8.19.19.06 switch +1-202-555-1313 INVITE sip:+12025551313@gw.carrier.com M1 IAM M2 Ringing voltage ACM M3 183 M4 PRACK M5 200 M6 RTP packets Ring tone Answer ANM M7 200 M8 ACK M9 RTP packets PCM speech Analog speech BYE M10 200 M12 REL M11 RLC M13 Hangup Figure 10.3 SIP to PSTN call through gateway. Call Flow Examples the PSTN gateway, SO it is able to send the INVITE directly to gw. car- rier.com. The gateway initiates the call into the PSTN by selecting an SS7 ISUP trunk to the next telephone switch in the PSTN. The dialed digits from the INVITE are mapped into the ISUP IAM. The ISUP Address Complete Message (ACM) is sent back by the PSTN to indicate that the trunk has been seized. Progress tones are generated in the one-way audio path established in the PSTN. In this example, ring tone is generated by the far end telephone switch. The gateway maps the ACM to the 183 Session Progress response containing SDP indicating the RTP port that the gateway will bridge the audio from the PSTN. Upon rece