The following documents contain provisions which, through reference in this text, constitute provisions of the present document.
[1]
TS 23.231: "SIP-I based Circuit Switched Core Network ; Stage 2".
[2]
TS 23.153: "Out of Band Transcoder Control; Stage 2".
[3]
TS 29.235: "Interworking between the 3GPP CS domain with SIP-I as signalling protocol and other networks".
[4]
ITU-T Recommendation Q.1912.5: "Interworking between Session Initiation Protocol (SIP) and Bearer Independent Call Control protocol or ISDN User Part".
[5]
RFC 2046: "Multipurpose Internet Mail Extensions (MIME) Part Two: Media Types".
[6]
RFC 3966: "The tel URI for Telephone Numbers".
[7]
[8]
RFC 3204: "MIME media types for ISUP and QSIG Objects".
[9]
RFC 3261: "SIP: Session Initiation Protocol".
[10]
RFC 3262: "Reliability of Provisional Responses in the Session Initiation Protocol (SIP)".
[11]
RFC 3264: "An Offer/Answer Model with the Session Description Protocol (SDP)".
[12]
RFC 3311: "The Session Initiation Protocol (SIP) UPDATE Method".
[13]
RFC 3312: "Integration of Resource Management and Session Initiation Protocol (SIP)".
[14]
RFC 3323: "A Privacy Mechanism for the Session Initiation Protocol (SIP)".
[15]
RFC 3325: "Private Extensions to the Session Initiation Protocol (SIP) for Network Asserted Identity within Trusted Networks".
[16]
RFC 3326: "The Reason Header Field for the Session Initiation Protocol (SIP)".
[17]
RFC 4566: "SDP: Session Description Protocol".
[18]
TS 29.232: "Media Gateway Controller (MGC) - Media Gateway (MGW) interface; Stage 3".
[19]
TS 29.415: "Core network Nb data transport and transport signalling".
[20]
TS 29.414: "Core Network Nb data transport and transport signalling".
[21]
TR 21.905: "Vocabulary for 3GPP Specifications".
[22]
RFC 3389: "Real-time Transport Protocol (RTP) Payload for Comfort Noise".
[23]
RFC 4733: "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals".
[24]
RFC 4028: "Session Timers in the Session Initiation Protocol (SIP)".
[25]
RFC 4960: "Stream Control Transmission Protocol".
[26] Void
[27]
RFC 4168: "The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP)".
[28]
RFC 5407: "Example Call Flows of Race Conditions in the Session Initiation Protocol (SIP)".
[29]
[30]
RFC 2460: "Internet Protocol, Version 6 (IPv6)".
[31]
RFC 3550: "RTP: A Transport Protocol for Real-Time Applications".
[31a]
TS 26.102: "Mandatory speech codec; Adaptive Multi-Rate (AMR) speech codec;Interface to Iu, Uu and Nb".
[32]
TS 26.103: "Speech codec list for GSM and UMTS".
[33]
TS 23.003: "Numbering, addressing and identification".
[34]
RFC 4715: " The Integrated Services Digital Network (ISDN) Subaddress Encoding Type for tel URI ".
[35]
RFC 4320: "Actions Addressing Identified Issues with the Session Initiation Protocol's (SIP) Non-INVITE Transaction".
[36]
RFC 5079: "Rejecting Anonymous Requests in the Session Initiation Protocol (SIP)".