Editing Session Initiation Protocol (section)

==Network elements==
The network elements that use the Session Initiation Protocol for communication are called ''SIP user agents''. Each ''user agent'' (UA) performs the function of a ''user agent client'' (UAC) when it is requesting a service function, and that of a ''user agent server'' (UAS) when responding to a request. Thus, any two SIP endpoints may in principle operate without any intervening SIP infrastructure. However, for network operational reasons, for provisioning public services to users, and for directory services, SIP defines several specific types of network server elements. Each of these service elements also communicates within the client-server model implemented in user agent clients and servers.<ref>{{Cite journal|last1=Montazerolghaem|first1=A.|last2=Moghaddam|first2=M. H. Y.|last3=Leon-Garcia|first3=A.|date=March 2018|title=OpenSIP: Toward Software-Defined SIP Networking|journal=IEEE Transactions on Network and Service Management|volume=15|issue=1|pages=184–199|doi=10.1109/TNSM.2017.2741258|issn=1932-4537|arxiv=1709.01320|s2cid=3873601}}</ref>

===User agent===
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.<ref name="RFC 3261"/>

A SIP phone is an [[IP phone]] that implements client and server functions of a SIP user agent and provides the traditional call functions of a telephone, such as dial, answer, reject, call hold, and call transfer.<ref name="Azzedine">{{cite book|last=Azzedine|title=Handbook of algorithms for wireless networking and mobile computing|publisher=CRC Press|year=2006|page=774|url=https://books.google.com/books?id=b8oisvv6fDAC&pg=PT774 | isbn=978-1-58488-465-1}}</ref><ref>{{cite book|last=Porter|first=Thomas |author2=Andy Zmolek |author3=Jan Kanclirz |author4=Antonio Rosela |title=Practical VoIP Security|publisher=Syngress |year=2006|pages=76–77|url=https://books.google.com/books?id=BYxdyekyRlwC&pg=PA76 | isbn=978-1-59749-060-3}}</ref> SIP phones may be implemented as a hardware device or as a [[softphone]]. As vendors increasingly implement SIP as a standard telephony platform, the distinction between hardware-based and software-based SIP phones is blurred and SIP elements are implemented in the basic firmware functions of many IP-capable communications devices such as [[smartphone]]s.

In SIP, as in HTTP, the [[user agent]] may identify itself using a message header field (''User-Agent''), containing a text description of the software, hardware, or the product name. The user agent field is sent in request messages, which means that the receiving SIP server can evaluate this information to perform device-specific configuration or feature activation. Operators of SIP network elements sometimes store this information in customer account portals,<ref>{{cite web |url=http://www.voipuser.org/forum_topic_14998.html |archive-date=2011-07-16 |archive-url=https://web.archive.org/web/20110716170218/http://www.voipuser.org/forum_topic_14998.html |title=User-Agents We Have Known |publisher=VoIP User}}</ref> where it can be useful in diagnosing SIP compatibility problems or in the display of service status.

===Proxy server===
A proxy server is a network server with UAC and UAS components that functions as an intermediary entity for the purpose of performing requests on behalf of other network elements. A proxy server primarily plays the role of call routing; it sends SIP requests to another entity closer to the destination. Proxies are also useful for enforcing policy, such as for determining whether a user is allowed to make a call. A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it.

SIP proxy servers that route messages to more than one destination are called forking proxies. The forking of a SIP request establishes multiple dialogs from the single request. Thus, a call may be answered from one of multiple SIP endpoints. For identification of multiple dialogs, each dialog has an identifier with contributions from both endpoints.

===Redirect server===
A redirect server is a user agent server that generates [[List of HTTP status codes#3xx redirection|3xx (redirection) responses]] to requests it receives, directing the client to contact an alternate set of URIs. A redirect server allows proxy servers to direct SIP session invitations to external domains.

===Registrar===
[[File:SIP-registration-flow.png|thumb|280px|SIP user agent registration to SIP registrar with authentication.]]

A registrar is a SIP endpoint that provides a location service. It accepts REGISTER requests, recording the address and other parameters from the user agent. For subsequent requests, it provides an essential means to locate possible communication peers on the network. The location service links one or more IP addresses to the SIP URI of the registering agent. Multiple user agents may register for the same URI, with the result that all registered user agents receive the calls to the URI.

SIP registrars are logical elements and are often co-located with SIP proxies. To improve network scalability, location services may instead be located with a redirect server.

===Session border controller===
[[File:SIP-B2BUA-call-flow.png|thumb|280px|Establishment of a session through a [[back-to-back user agent]].]]

[[Session border controller]]s (SBCs) serve as [[middlebox]]es between user agents and SIP servers for various types of functions, including network topology hiding and assistance in [[NAT traversal]]. SBCs are an independently engineered solution and are not mentioned in the SIP RFC.

===Gateway===
[[Gateway (telecommunications)|Gateways]] can be used to interconnect a SIP network to other networks, such as the PSTN, which use different protocols or technologies.