Editing Session Initiation Protocol (section)

==Protocol operation==
[[File:SIP session setup example.svg|300px|thumb|An example of a SIP message exchange between two users, Alice and Bob, to establish and end a direct media session.]]
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. SIP can be used to establish two-party ([[unicast]]) or multiparty ([[multicast]]) sessions. It also allows modification of existing calls. The modification can involve changing addresses or [[Computer port (software)|ports]], inviting more participants, and adding or deleting media streams. SIP has also found applications in messaging applications, such as instant messaging, and event subscription and notification.

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. 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. 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).<ref name="Johnston"/><ref>{{Cite book |title=Telecom 101 |last=Coll |first=Eric |publisher=Teracom Training Institute |year=2016 |isbn=9781894887038 |pages=77–79}}</ref>

Every resource of a SIP network, such as user agents, call routers, and voicemail boxes, are identified by a [[Uniform Resource Identifier]] (URI). The syntax of the URI follows the general standard syntax also used in [[Web service]]s and e-mail.<ref name="RFC 3986">{{cite IETF |rfc=3986 |title=Uniform Resource Identifiers (URI): Generic Syntax |date=2005}}</ref> The URI scheme used for SIP is ''sip'' and a typical SIP URI has the form ''<nowiki>sip:username@domainname</nowiki>'' or ''<nowiki>sip:username@hostport</nowiki>'', where ''domainname'' requires DNS [[SRV record]]s 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. If [[secure transmission]] is required, the scheme ''sips'' is used.{{sfn|Miikka Poikselkä|Georg Mayer|Hisham Khartabil|Aki Niemi|2004}}{{sfn|Brian Reid|Steve Goodman|2015}}

SIP employs design elements similar to the HTTP request and response transaction model.<ref>{{cite web|title=SIP: Session Initiation Protocol|url=https://www.ietf.org/rfc/rfc3261|website=IETF}}</ref> Each transaction consists of a client request that invokes a particular method or function on the server and at least one response. SIP reuses most of the header fields, encoding rules and status codes of HTTP, providing a readable text-based format.

SIP can be carried by several [[transport layer]] protocols including [[Transmission Control Protocol]] (TCP), [[User Datagram Protocol]] (UDP), and [[Stream Control Transmission Protocol]] (SCTP).<ref name="RFC 4168">{{cite IETF |rfc=4168 |title=The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP) |date=2005}}</ref><ref>{{Cite journal|last1=Montazerolghaem|first1=Ahmadreza|last2=Hosseini Seno|first2=Seyed Amin|last3=Yaghmaee|first3=Mohammad Hossein|last4=Tashtarian|first4=Farzad|date=2016-06-01|title=Overload mitigation mechanism for VoIP networks: a transport layer approach based on resource management|journal=Transactions on Emerging Telecommunications Technologies|volume=27|issue=6|pages=857–873|doi=10.1002/ett.3038|s2cid=27215205 |issn=2161-3915}}</ref>  SIP clients typically use TCP or UDP on [[port number]]s 5060 or 5061 for SIP traffic to servers and other endpoints. Port 5060 is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used for traffic encrypted with [[Transport Layer Security]] (TLS).

SIP-based telephony networks often implement call processing features of [[Signaling System 7]] (SS7), for which special SIP protocol extensions exist, although the two protocols themselves are very different. SS7 is a centralized protocol, characterized by a complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a [[client-server]] protocol of equipotent peers. SIP features are implemented in the communicating endpoints, while the traditional SS7 architecture is in use only between switching centers.