Understanding Session Initiation Protocol Pdf Download
SIP is a signalling protocol designed to create, modify, and terminate a multimedia session over the Internet Protocol. It is an application layer protocol that incorporates many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP). This tutorial covers most of the topics required for a basic understanding of SIP and to get a feel of how it works.
understanding session initiation protocol pdf download
SIP (Session Initiation Protocol) is a signaling protocol, widely used for setting up, connecting and disconnecting communication sessions, typically voice or video calls over the Internet. SIP is a standardized protocol with its basis coming from the IP community and in most cases uses UDP or TCP. The protocol can be used for setting up, modifying and terminating two-party (unicast), or multiparty (multicast) sessions consisting of one or more media streams. Modifications can include changing IP addresses or/or ports, inviting more participants, and adding or deleting the media streams.
As shown above certain information is sent along with an Invite which starts the process to establish a call session. That call session is typically voice sent via RTP (Realtime Transport Protocol). The Real-Time Transport Protocol (RTP) is an Internet protocol standard that specifies a way for programs to manage real time transmission of multimedia data, with VoIP is usually voice, but could be video, as well. For a list of SIP response codes and their corresponding meanings we have provided a list to the left, along with a PDF download for reference.
The Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating communication sessions that include voice, video and messaging applications.[1] SIP is used in Internet telephony, in private IP telephone systems, as well as mobile phone calling over LTE (VoLTE).
SIP works in conjunction with several other protocols that specify and carry the session media. 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 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). For secure transmissions of SIP messages over insecure network links, the protocol may be encrypted with Transport Layer Security (TLS). 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).
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. The protocol was standardized as .mw-parser-output cite.citationfont-style:inherit;word-wrap:break-word.mw-parser-output .citation qquotes:"\"""\"""'""'".mw-parser-output .citation:targetbackground-color:rgba(0,127,255,0.133).mw-parser-output .id-lock-free a,.mw-parser-output .citation .cs1-lock-free abackground:url("//upload.wikimedia.org/wikipedia/commons/6/65/Lock-green.svg")right 0.1em center/9px no-repeat.mw-parser-output .id-lock-limited a,.mw-parser-output .id-lock-registration a,.mw-parser-output .citation .cs1-lock-limited a,.mw-parser-output .citation .cs1-lock-registration abackground:url("//upload.wikimedia.org/wikipedia/commons/d/d6/Lock-gray-alt-2.svg")right 0.1em center/9px no-repeat.mw-parser-output .id-lock-subscription a,.mw-parser-output .citation .cs1-lock-subscription abackground:url("//upload.wikimedia.org/wikipedia/commons/a/aa/Lock-red-alt-2.svg")right 0.1em center/9px no-repeat.mw-parser-output .cs1-ws-icon abackground:url("//upload.wikimedia.org/wikipedia/commons/4/4c/Wikisource-logo.svg")right 0.1em center/12px no-repeat.mw-parser-output .cs1-codecolor:inherit;background:inherit;border:none;padding:inherit.mw-parser-output .cs1-hidden-errordisplay:none;color:#d33.mw-parser-output .cs1-visible-errorcolor:#d33.mw-parser-output .cs1-maintdisplay:none;color:#3a3;margin-left:0.3em.mw-parser-output .cs1-formatfont-size:95%.mw-parser-output .cs1-kern-leftpadding-left:0.2em.mw-parser-output .cs1-kern-rightpadding-right:0.2em.mw-parser-output .citation .mw-selflinkfont-weight:inheritRFC 2543 in 1999. 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. In June 2002 the specification was revised in RFC 3261[3] and various extensions and clarifications have been published since.[4]
A user agent is a logical network endpoint that sends or receives SIP messages and manages SIP sessions. User agents have client and server components. The user agent client (UAC) sends SIP requests. The user agent server (UAS) receives requests and returns a SIP response. Unlike other network protocols that fix the roles of client and server, e.g., in HTTP, in which a web browser only acts as a client, and never as a server, SIP requires both peers to implement both roles. The roles of UAC and UAS only last for the duration of a SIP transaction.[5]
SIP-I, Session Initiation Protocol with encapsulated ISUP, is a protocol used to create, modify, and terminate communication sessions based on ISUP using SIP and IP networks. Services using SIP-I include voice, video telephony, fax and data. SIP-I and SIP-T[28] are two protocols with similar features, notably to allow ISUP messages to be transported over SIP networks. This preserves all of the detail available in the ISUP header.[a] SIP-I was defined by the ITU-T, whereas SIP-T was defined by the IETF.[29] 350c69d7ab