television.tv CSeq: 3 BYE M13 RLC SIP: Understanding the Session Initiation Protocol 10.4 PSTN to SIP Call Through Gateway Figure 10.4 shows a call originating from a telephone in the PSTN that termi- nates on a SIP phone in the Internet. The compact form of SIP is used through- out the example. Note that there is no compact form for CSeq or Max-Forwards. M1 Setup CdPN=6512345, NPI=E.164, NOA=International <Dialed telephone number CgPN=4567890, NPI=E.164, NOA=International <PSTN caller's number INVITE sip:+6512345@incoming.com;user=phone SIP/2.0 V: SIP/2.0/UDP 65.3.4.1:5060;branch=z9hG4bK343 <Compact form of headers Max-Forwards: 70 f: <sip:+45.67890@incoming.com;user=phone>;tag=6a589b1 <includes t : <sip:+65.12345@incoming.com;user=phone> i: a3-65-99-1d@65.3.4.1 CSeq: 1 INVITE m: sip gw3.incoming.com C: application/sdp l: 126 Proxy telephone Database gateway server phone switch Setup CdPn=+6512345 M1 INVITEsip:+6512345@gw.carrier.com M2 +6512345? M4 100 M3 sip-phone@home.com M5 INVITE sipphone@home.com M6 180 M7 180 M8 Alterting M9 200 M11 200 M10 Connect M12 ACK M13 PCM speech RTP packets Figure 10.4 PSTN to SIP phone through gateway. Call Flow Examples O=- 2890844535 2890844535 IN IP4 65.3.4.1 C=IN IP4 65.3.4.1 m=audio 62432 RTP/AVP 0 a=rtpmap: 0 PCMU/8000 SIP/2.0 100 Trying V : SIP/2.0/UDP 65.3.4.1:5060branch=z9hG4bk343 f: sip:+45.67890@incoming.com;user=phone>;tag=6a589b1 t : sip:+65.12345@incoming.com;user=phone i: a3-65-99-1d@65.3.4.1 CSeq: 1 INVITE Service Query: +65-12345 M5 Location Service Response: sip:user@home.com <Number maps to SIP URI INVITE sip:user@home.com SIP/2.0 V : SIP/2.0/UDP 176.5.8.2:5060;branch=z9hG4bK942834822.1 . V : SIP/2.0/UDP 5.3.4.1:5060;branch=z9hG4bK343 Max-Forwards: 69 f: ksip:+45.67890@incoming.com;user=phone>;tag=6a589b1 t : sip:+65.12345@incoming.com;user=phone i: a3-65-99-1d@65.3.4.1 CSeq: 1 INVITE m: sip:gw3.incoming.com C : application/sdp 1:126 O=- 2890844535 2890844535 IN IP4 65.3.4.1 t=0 0 C=IN IP4 65.3.4.1 m=audio 62432 RTP/AVP 0 a=rtpmap: 0 PCMU/8000 SIP/2.0 180 Ring