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The Session Initiation Protocol (SIP) is the signaling protocol selected by the 3rd Generation Partnership Project (3GPP)〔〔 to create and control multimedia sessions with two or more participants in the IP Multimedia Subsystem (IMS), and therefore is a key element in the IMS framework. SIP was developed by the Internet Engineering Task Force (IETF) to become a standard for controlling multimedia sessions in Internet Protocol (IP) networks, working in the application layer. Several SIP extensions〔〔〔 have been added to the basic protocol specification〔 in order to extend its functionality. These extensions are based on Request for Comments (RFC) protocol recommendations by the IETF. The 3GPP, which is a collaboration between groups of telecommunications associations aimed at developing and maintaining the IMS, stated a series of requirements for SIP〔 to be successfully used in the IMS. Some of them could be addressed by using existing capabilities and extensions in SIP while, in other cases, the 3GPP had to collaborate with the IETF to standardize new SIP extensions〔 to meet the new requirements. In any case, the IETF evolves SIP in a generic basis, so that the use of its extensions is not restricted to the IMS framework. == 3GPP requirements on SIP == There are some general requirements stated by the 3GPP for the IMS to work successfully. These include an efficient use of the radio interface by minimizing the exchange of signaling messages between the mobile terminal and the network, a minimum session setup time by performing tasks prior to session establishment instead of during session establishment, a minimum support required in the terminal, the support for roaming and non-roaming scenarios with terminal mobility management (supported by the access network, not SIP), and support for IPv6 addresses. Other requirements are more specific and involve SIP modifications, such as new header fields to exchange user or server's information, and new SIP methods to support new network functionality: requirement for registration, re-registration, de-registration, subscription to events and notifications, instant messaging or call control primitives with additional capabilities such as call transference. Some of these specific requirements〔 are: *Quality of service support with policy and charging control, as well as resource negotiation and allocation before alerting the destination user. *Identification of users for authentication, authorization and accounting purposes. Security between users and the network and among network nodes is a major issue to be addressed by using mutual authentication mechanisms such as private and public keys and digests, as well as media authorization extensions. It must be also possible to present both the caller and the called party the identities of their counterparts, with the ability to hide this information if required. Anonymity in session establishment and privacy are also important. *Protection of SIP signaling with integrity and confidentiality support based on initial authentication and symmetric cryptographic keys; error recovery and verification are also needed. *Session release initiated by the network (e.g. in case the user terminal leaves coverage or runs out of credit). *Source-routing mechanisms. The routing of SIP messages has its own requirements in the IMS as all terminal originated session setup attempts must transit both the P-CSCF and the S-CSCF so that these call session control functions (CSCFs) servers can properly provide their services. There can be special path requirements for certain messages as well. *Interoperation between IMS and the public switched telephone network (PSTN). Finally, it is also necessary that other protocols and network services such as DHCP or DNS〔 are adapted to work with SIP, for instance for outbound proxy (P-CSCF) location and SIP Uniform Resource Identifier (URI) to IP address resolution, respectively. 抄文引用元・出典: フリー百科事典『 ウィキペディア（Wikipedia）』 ■ウィキペディアで「Extensions to the SIP for the IP multimedia subsystem」の詳細全文を読む スポンサード リンク
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