![]() ![]() If secure transmission is required, the scheme sips is used. The URI scheme used for SIP is sip and a typical SIP URI has the form or where domainname requires DNS SRV records to locate the servers for SIP domain while hostport can be an IP address or a fully qualified domain name of the host and port. The syntax of the URI follows the general standard syntax also used in Web services and e-mail. Įvery resource of a SIP network, such as user agents, call routers, and voicemail boxes, are identified by a Uniform Resource Identifier (URI). Voice and video media streams are typically carried between the terminals using the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP). For call setup, the body of a SIP message contains a Session Description Protocol (SDP) data unit, which specifies the media format, codec and media communication protocol. SIP works in conjunction with several other protocols that specify the media format and coding and that carry the media once the call is set up. SIP has also found applications in messaging applications, such as instant messaging, and event subscription and notification. The modification can involve changing addresses or ports, inviting more participants, and adding or deleting media streams. It also allows modification of existing calls. SIP can be used to establish two-party ( unicast) or multiparty ( multicast) sessions. SIP is only involved in the signaling operations of a media communication session and is primarily used to set up and terminate voice or video calls. Protocol operation An example of a SIP message exchange between two users, Alice and Bob, to establish and end a direct media session. SIP has been standardized primarily by the Internet Engineering Task Force (IETF), while other protocols, such as H.323, have traditionally been associated with the International Telecommunication Union (ITU). SIP is distinguished by its proponents for having roots in the Internet community rather than in the telecommunications industry. It has been extended for video conferencing, streaming media distribution, instant messaging, presence information, file transfer, Internet fax and online games. SIP was designed to provide a signaling and call setup protocol for IP-based communications supporting the call processing functions and features present in the public switched telephone network (PSTN) with a vision of supporting new multimedia applications. In June 2002 the specification was revised in RFC 3261 and various extensions and clarifications have been published since. In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular networks. The protocol was standardized as RFC 2543 in 1999. SIP was originally designed by Mark Handley, Henning Schulzrinne, Eve Schooler and Jonathan Rosenberg in 1996 to facilitate establishing multicast multimedia sessions on the Mbone. ![]() ![]() For the transmission of media streams (voice, video) the SDP payload carried in SIP messages typically employs the Real-time Transport Protocol (RTP) or the Secure Real-time Transport Protocol (SRTP). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). SIP is designed to be independent of the underlying transport layer protocol and can be used with the User Datagram Protocol (UDP), the Transmission Control Protocol (TCP), and the Stream Control Transmission Protocol (SCTP). Most commonly, media type and parameter negotiation and media setup are performed with the Session Description Protocol (SDP), which is carried as payload in SIP messages. SIP works in conjunction with several other protocols that specify and carry the session media. A call established with SIP may consist of multiple media streams, but no separate streams are required for applications, such as text messaging, that exchange data as payload in the SIP message. ![]() SIP is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). The protocol defines the specific format of messages exchanged and the sequence of communications for cooperation of the participants. SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE ( VoLTE). The Session Initiation Protocol ( SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications. ![]()
0 Comments
Leave a Reply. |
AuthorWrite something about yourself. No need to be fancy, just an overview. ArchivesCategories |