Session Initiation Protocol

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).[2]

The protocol defines the specific format of messages Session Initiation Protocol
exchanged and the sequence of communications for
Communication protocol
cooperation of the participants. SIP is a text-based
protocol, incorporating many elements of the Hypertext Abbreviation SIP
Transfer Protocol (HTTP) and the Simple Mail Transfer
Purpose Internet
Protocol (SMTP).[3] A call established with SIP may telephony
consist of multiple media streams, but no separate
streams are required for applications, such as text Introduction March 1999

messaging, that exchange data as payload in the SIP OSI layer Application layer
message. (Layer 7)

SIP works in conjunction with several other protocols that Port(s) 5060, 5061
specify and carry the session media. Most commonly,
RFC(s) 2543, 3261
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).

History

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 RFC 2543 (https://datatracker.ietf.org/doc/html/rfc2543) 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 (https://datatracker.ietf.org/doc/
[4]
html/rfc3261) and various extensions and clarifications have been published since.[5]

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. It has been extended for
video conferencing, streaming media distribution, instant messaging, presence information, file
transfer, Internet fax and online games.[1][6][7]

SIP is distinguished by its proponents for having roots in the Internet community rather than in the
telecommunications industry. 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).

Protocol operation

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 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).[3][8]

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 services and e-mail.[9] The URI scheme used for SIP is sip and a typical SIP
URI has the form sip:username@domainname or sip:username@hostport, 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. If secure transmission is required, the scheme
sips is used.[10][11]

SIP employs design elements similar to the HTTP request and response transaction model.[12] 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).[13][14] SIP
clients typically use TCP or UDP on port numbers 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.
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.[15]

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.[6]

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.[16][17] 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 smartphones.

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,[18] 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 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

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

Establishment of a session through a back-to-

back user agent.

Session border controllers (SBCs) serve as middleboxes 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

Gateways can be used to interconnect a SIP network to other networks, such as the PSTN, which
use different protocols or technologies.

SIP messages

SIP is a text-based protocol with syntax similar to that of HTTP. There are two different types of SIP
messages: requests and responses. The first line of a request has a method, defining the nature of
the request, and a Request-URI, indicating where the request should be sent.[19] The first line of a
response has a response code.

Requests

Requests initiate a functionality of the protocol. They are sent by a user agent client to the server
and are answered with one or more SIP responses, which return a result code of the transaction,
and generally indicate the success, failure, or other state of the transaction.
SIP requests

Request
Description Notes RFC references
name

Register the URI listed in the To-header field

RFC 3261 (https://dat
with a location server and associates it with The command implements a
REGISTER atracker.ietf.org/doc/
the network address given in a Contact header location service.
html/rfc3261)
field.

When sent during an established

Initiate a dialog for establishing a call. The RFC 3261 (https://dat
dialog (reinvite) it modifies the
INVITE request is sent by a user agent client to a user atracker.ietf.org/doc/
sessions, for example placing a
agent server. html/rfc3261)
call on hold.

RFC 3261 (https://dat

Confirm that an entity has received a final
ACK atracker.ietf.org/doc/
response to an INVITE request.
html/rfc3261)

RFC 3261 (https://dat

This message may be sent by
BYE Signal termination of a dialog and end a call. atracker.ietf.org/doc/
either endpoint of a dialog.
html/rfc3261)

Usually means terminating a call RFC 3261 (https://dat

CANCEL Cancel any pending request. while it is still ringing, before atracker.ietf.org/doc/
answer. html/rfc3261)

RFC 3311 (https://dat

Modify the state of a session without
UPDATE atracker.ietf.org/doc/
changing the state of the dialog.
html/rfc3311)

RFC 3515 (https://dat

Ask recipient to issue a request for the
REFER atracker.ietf.org/doc/
purpose of call transfer.
html/rfc3515)

RFC 3262 (https://dat

PRACK is sent in response to
PRACK Provisional acknowledgement. atracker.ietf.org/doc/
provisional response (1xx).
html/rfc3262)

RFC 6665 (https://dat

Initiates a subscription for notification of
SUBSCRIBE atracker.ietf.org/doc/
events from a notifier.
html/rfc6665)

RFC 6665 (https://dat

Inform a subscriber of notifications of a new
NOTIFY atracker.ietf.org/doc/
event.
html/rfc6665)

RFC 3903 (https://dat

PUBLISH Publish an event to a notification server. atracker.ietf.org/doc/
html/rfc3903)

MESSAGE Deliver a text message. Used in instant messaging RFC 3428 (https://dat
applications. atracker.ietf.org/doc/
 html/rfc3428)

RFC 6086 (https://dat

Send mid-session information that does not This method is often used for
INFO atracker.ietf.org/doc/
modify the session state. DTMF relay.
html/rfc6086)

RFC 3261 (https://dat

It is often used for NAT keepalive
OPTIONS Query the capabilities of an endpoint. atracker.ietf.org/doc/
purposes.
html/rfc3261)

Responses

Responses are sent by the user agent server indicating the result of a received request. Several
classes of responses are recognized, determined by the numerical range of result codes:[20]

1xx: Provisional responses to requests indicate the request was valid and is being processed.

2xx: Successful completion of the request. As a response to an INVITE, it indicates a call is

established. The most common code is 200, which is an unqualified success report.

3xx: Call redirection is needed for completion of the request. The request must be completed with
a new destination.

4xx: The request cannot be completed at the server for a variety of reasons, including bad request
syntax (code 400).

5xx: The server failed to fulfill an apparently valid request, including server internal errors (code
500).

6xx: The request cannot be fulfilled at any server. It indicates a global failure, including call
rejection by the destination.
Transactions

Example: User1's UAC uses an invite

client transaction to send the initial
INVITE (1) message. If no response is
received after a timer-controlled wait
period the UAC may choose to
terminate the transaction or
retransmit the INVITE. Once a
response is received, User1 is
confident the INVITE was delivered
reliably. User1's UAC must then
acknowledge the response. On
delivery of the ACK (2), both sides of
the transaction are complete. In this
case, a dialog may have been
established.[21]

SIP defines a transaction mechanism to control the exchanges between participants and deliver
messages reliably. A transaction is a state of a session, which is controlled by various timers. Client
transactions send requests and server transactions respond to those requests with one or more
responses. The responses may include provisional responses with a response code in the form 1xx,
and one or multiple final responses (2xx – 6xx).

Transactions are further categorized as either type invite or type non-invite. Invite transactions differ
in that they can establish a long-running conversation, referred to as a dialog in SIP, and so include
an acknowledgment (ACK) of any non-failing final response, e.g., 200 OK.

Instant messaging and presence

The Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions
(SIMPLE) is the SIP-based suite of standards for instant messaging and presence information.
Message Session Relay Protocol (MSRP) allows instant message sessions and file transfer.
Conformance testing

The SIP developer community meets regularly at conferences organized by SIP Forum to test
interoperability of SIP implementations.[22] The TTCN-3 test specification language, developed by a
task force at ETSI (STF 196), is used for specifying conformance tests for SIP implementations.[23]

Performance testing

When developing SIP software or deploying a new SIP infrastructure, it is important to test the
capability of servers and IP networks to handle certain call load: number of concurrent calls and
number of calls per second. SIP performance tester software is used to simulate SIP and RTP traffic
to see if the server and IP network are stable under the call load.[24] The software measures
performance indicators like answer delay, answer/seizure ratio, RTP jitter and packet loss, round-trip
delay time.

Applications

SIP connection is a marketing term for voice over Internet Protocol (VoIP) services offered by many
Internet telephony service providers (ITSPs). The service provides routing of telephone calls from a
client's private branch exchange (PBX) telephone system to the PSTN. Such services may simplify
corporate information system infrastructure by sharing Internet access for voice and data, and
removing the cost for Basic Rate Interface (BRI) or Primary Rate Interface (PRI) telephone circuits.

SIP trunking is a similar marketing term preferred for when the service is used to simplify a telecom
infrastructure by sharing the carrier access circuit for voice, data, and Internet traffic while removing
the need for PRI circuits.[25][26]

SIP-enabled video surveillance cameras can initiate calls to alert the operator of events, such as the
motion of objects in a protected area.

SIP is used in audio over IP for broadcasting applications where it provides an interoperable means
for audio interfaces from different manufacturers to make connections with one another.[27]

Implementations

The U.S. National Institute of Standards and Technology (NIST), Advanced Networking Technologies
Division provides a public-domain Java implementation[28] that serves as a reference
implementation for the standard. The implementation can work in proxy server or user agent
scenarios and has been used in numerous commercial and research projects. It supports RFC 3261
(https://datatracker.ietf.org/doc/html/rfc3261) in full and a number of extension RFCs including
RFC 6665 (https://datatracker.ietf.org/doc/html/rfc6665) (event notification) and RFC 3262 (http
s://datatracker.ietf.org/doc/html/rfc3262) (reliable provisional responses).

Numerous other commercial and open-source SIP implementations exist. See List of SIP software.

SIP-ISUP interworking

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[29] 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.[30]

Encryption

Concerns about the security of calls via the public Internet have been addressed by encryption of
the SIP protocol for secure transmission. The URI scheme SIPS is used to mandate that SIP
communication be secured with Transport Layer Security (TLS). SIPS URIs take the form
sips:user@example.com .

End-to-end encryption of SIP is only possible if there is a direct connection between communication
endpoints. While a direct connection can be made via Peer-to-peer SIP or via a VPN between the
endpoints, most SIP communication involves multiple hops, with the first hop being from a user
agent to the user agent's ITSP. For the multiple-hop case, SIPS will only secure the first hop; the
remaining hops will normally not be secured with TLS and the SIP communication will be insecure.
In contrast, the HTTPS protocol provides end-to-end security as it is done with a direct connection
and does not involve the notion of hops.

The media streams (audio and video), which are separate connections from the SIPS signaling
stream, may be encrypted using SRTP. The key exchange for SRTP is performed with SDES
(RFC 4568 (https://datatracker.ietf.org/doc/html/rfc4568) ), or with ZRTP (RFC 6189 (https://datatr
acker.ietf.org/doc/html/rfc6189) ). When SDES is used, the keys will be transmitted via insecure
SIP unless SIPS is used. One may also add a MIKEY (RFC 3830 (https://datatracker.ietf.org/doc/htm
l/rfc3830) ) exchange to SIP to determine session keys for use with SRTP.
See also

Computer telephony integration (CTI)

Computer-supported telecommunications applications (CSTA)

H.323 protocols H.225.0 and H.245

IP Multimedia Subsystem (IMS)

Media Gateway Control Protocol (MGCP)

Mobile VoIP

MSCML (Media Server Control Markup Language)

Network convergence

Rendezvous protocol

RTP payload formats

SIGTRAN (Signaling Transport)

SIP extensions for the IP Multimedia Subsystem

SIP provider

Skinny Client Control Protocol (SCCP)

T.38

XIMSS (XML Interface to Messaging, Scheduling, and Signaling)

Notes

a. ISUP detail is important as there are many country-specific variants of ISUP that have been implemented
over the last 30 years, and it is not always possible to express all of the same detail using a native SIP
message.

References

1. "What is SIP?" (https://web.archive.org/web/20140624044447/http://www.networkworld.com/article/233

2980/lan-wan/what-is-sip-.html) . Network World. May 11, 2004. Archived from the original (http://www.n
etworkworld.com/article/2332980/lan-wan/what-is-sip-.html) on June 24, 2014.
 2. "4G | ShareTechnote" (https://www.sharetechnote.com/html/Handbook_LTE_VoLTE.html) .
www.sharetechnote.com. Retrieved 2023-03-09.

3. Johnston, Alan B. (2004). SIP: Understanding the Session Initiation Protocol (Second ed.). Artech House.
ISBN 9781580531689.

4. "SIP core working group charter" (http://www.ietf.org/dyn/wg/charter/sipcore-charter.html) . Internet

Engineering Task Force. 2010-12-07. Retrieved 2011-01-11.

5. "Search Internet-Drafts and RFCs" (https://datatracker.ietf.org/doc/search/?name=SIP&rfcs=on&sort=dat

e) . Internet Engineering Task Force.

6. SIP: Session Initiation Protocol (https://datatracker.ietf.org/doc/html/rfc3261) . 2002.

doi:10.17487/RFC3261 (https://doi.org/10.17487%2FRFC3261) . RFC 3261 (https://datatracker.ietf.org/
doc/html/rfc3261) .

7. Rouse, Margaret. "Session Initiation Protocol (SIP)" (http://searchunifiedcommunications.techtarget.com/

definition/Session-Initiation-Protocol) . TechTarget.

8. Coll, Eric (2016). Telecom 101. Teracom Training Institute. pp. 77–79. ISBN 9781894887038.

9. Uniform Resource Identifiers (URI): Generic Syntax (https://datatracker.ietf.org/doc/html/rfc3986) . 2005.

doi:10.17487/RFC3986 (https://doi.org/10.17487%2FRFC3986) . RFC 3986 (https://datatracker.ietf.org/
doc/html/rfc3986) .

10. Miikka Poikselkä et al. 2004.

11. Brian Reid & Steve Goodman 2015.

12. "SIP: Session Initiation Protocol" (https://www.ietf.org/rfc/rfc3261) . IETF.

13. The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP) (htt
ps://datatracker.ietf.org/doc/html/rfc4168) . 2005. doi:10.17487/RFC4168 (https://doi.org/10.17487%2
FRFC4168) . RFC 4168 (https://datatracker.ietf.org/doc/html/rfc4168) .

14. Montazerolghaem, Ahmadreza; Hosseini Seno, Seyed Amin; Yaghmaee, Mohammad Hossein; Tashtarian,
Farzad (2016-06-01). "Overload mitigation mechanism for VoIP networks: a transport layer approach
based on resource management". Transactions on Emerging Telecommunications Technologies. 27 (6):
857–873. doi:10.1002/ett.3038 (https://doi.org/10.1002%2Fett.3038) . ISSN 2161-3915 (https://www.wo
rldcat.org/issn/2161-3915) . S2CID 27215205 (https://api.semanticscholar.org/CorpusID:27215205) .

15. Montazerolghaem, A.; Moghaddam, M. H. Y.; Leon-Garcia, A. (March 2018). "OpenSIP: Toward Software-
Defined SIP Networking". IEEE Transactions on Network and Service Management. 15 (1): 184–199.
arXiv:1709.01320 (https://arxiv.org/abs/1709.01320) . doi:10.1109/TNSM.2017.2741258 (https://doi.or
g/10.1109%2FTNSM.2017.2741258) . ISSN 1932-4537 (https://www.worldcat.org/issn/1932-4537) .
S2CID 3873601 (https://api.semanticscholar.org/CorpusID:3873601) .

16. Azzedine (2006). Handbook of algorithms for wireless networking and mobile computing (https://books.goo
gle.com/books?id=b8oisvv6fDAC&pg=PT774) . CRC Press. p. 774. ISBN 978-1-58488-465-1.
17. Porter, Thomas; Andy Zmolek; Jan Kanclirz; Antonio Rosela (2006). Practical VoIP Security (https://books.g
oogle.com/books?id=BYxdyekyRlwC&pg=PA76) . Syngress. pp. 76–77. ISBN 978-1-59749-060-3.

18. "User-Agents We Have Known" (https://web.archive.org/web/20110716170218/http://www.voipuser.org/f

orum_topic_14998.html) . VoIP User. Archived from the original (http://www.voipuser.org/forum_topic_1
4998.html) on 2011-07-16.

19. Stallings, p.214

20. Stallings, pp.216-217

21. Wright, James. "SIP - An Introduction" (http://www.konnetic.com/Documents/KonneticSIPIntroduction.pd

f) (PDF). Konnetic. Retrieved 2011-01-11.

22. "SIPit Wiki" (http://www.sipit.net/) . Retrieved 2017-10-07.

23. Experiences of Using TTCN-3 for Testing SIP and also OSP (https://web.archive.org/web/20140330061038/
http://portal.etsi.org/ptcc/downloads/TTCN3SIPOSP.pdf) (PDF), archived from the original (http://portal.
etsi.org/ptcc/downloads/TTCN3SIPOSP.pdf) (PDF) on March 30, 2014

24. "Performance and Stress Testing of SIP Servers, Clients and IP Networks" (http://startrinity.com/VoIP/Tes
tingSipPbxSoftswitchServer.aspx) . StarTrinity. 2016-08-13.

25. "AT&T Discusses Its SIP Peering Architecture" (http://sip-trunking.tmcnet.com/topics/enterprise-voip/artic

les/109840-att-discusses-its-sip-peering-architecture.htm) . sip-trunking.tmcnet.com. Retrieved
2017-03-20.

26. "From IIT VoIP Conference & Expo: AT&T SIP transport PowerPoint slides" (http://hdvoicenews.com/2010/
10/18/from-iit-voip-conference-expo-att-sip-transport-powerpoint-slides/) . HD Voice News. 2010-10-19.
Retrieved 2017-03-20.

27. Jonsson, Lars; Mathias Coinchon (2008). "Streaming audio contributions over IP" (http://tech.ebu.ch/web
dav/site/tech/shared/techreview/trev_2008-Q1_coinchon.pdf) (PDF). EBU Technical Review. Retrieved
2010-12-27.

28. "JAIN SIP project" (https://github.com/usnistgov/jsip) . Retrieved 2024-06-24.

29. SIP-T Context and Architectures (https://datatracker.ietf.org/doc/html/rfc3372) . September 2002.

doi:10.17487/RFC3372 (https://doi.org/10.17487%2FRFC3372) . RFC 3372 (https://datatracker.ietf.org/
doc/html/rfc3372) .

30. "Why SIP-I? A Switching Core Protocol Recommendation" (https://web.archive.org/web/2012031709160

3/http://www.4gamericas.org/documents/3G_Americas_SIP-I_White_Paper_August_2007-FINAL.pdf)
(PDF). Archived from the original (http://www.4gamericas.org/documents/3G_Americas_SIP-I_White_Pap
er_August_2007-FINAL.pdf) (PDF) on 2012-03-17.

Brian Reid; Steve Goodman (22 January 2015), Exam Ref 70-342 Advanced Solutions of Microsoft Exchange
Server 2013 (MCSE), Microsoft Press, p. 24, ISBN 9780735697904
 Miikka Poikselkä; Georg Mayer; Hisham Khartabil; Aki Niemi (19 November 2004), The IMS: IP Multimedia
Concepts and Services in the Mobile Domain, John Wiley & Sons, p. 268, ISBN 978047087114-0

External links

IANA: SIP Parameters (https://www.iana.org/assignments/sip-parameters)

IANA: SIP Event Types Namespace (https://www.iana.org/assignments/sip-events/sip-events.xht

ml)