Session Initiation Protocol (SIP)
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Session Initiation Protocol (SIP) is a protocol developed by IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality.
SIP clients traditionally use TCP and UDP port 5060 to connect to SIP servers and other SIP endpoints. SIP is primarily used in setting up and tearing down voice or video calls. However, it can be used in any application where session initiation is a requirement. These include, Event Subscription and Notification, Terminal mobility and so on. There are a large number of SIP-related RFCs that define behavior for such applications. All voice/video communications are done over RTP.
A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the public switched telephone network (PSTN).
SIP enabled telephony networks can also implement many of the more advanced call processing features present in Signalling System 7 (SS7), though the two protocols themselves are very different. SS7 is a highly centralized protocol, characterized by highly complex central network architecture and dumb endpoints (traditional telephone handsets). SIP is a peer-to-peer protocol.
SIP network elements
Hardware endpoints, devices with the look, feel, and shape of a traditional telephone, but that use SIP and RTP for communication, are commercially available from several vendors. Some of these can use Electronic Numbering (ENUM) or DUNDi to translate existing phone numbers to SIP addresses using DNS, so calls to other SIP users can bypass the telephone network, even though your service provider might normally act as a gateway to the PSTN network for traditional phone numbers (and charge you for it).
SIP makes use of elements called proxy servers to help route requests to the user s current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users.
SIP also provides a registration function that allows users to upload their current locations for use by proxy servers.
Since registrations play an important role in SIP, a User Agent Server that handles a REGISTER is given the special name registrar.
It is an important concept that the distinction between types of SIP servers is logical, not physical. ..
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Ramesh G. Shihora

.doc   SIP - Session Initiation Protocol.doc (Size: 133.5 KB / Downloads: 107)
Session Initiation Protocol (SIP) is quickly gaining popularly with application service providers (ASPs), communication service providers(CSPs), and network service providers (NSPs) focused on offering their customers innovative, new IP - based services . Adopted in 1999 by the Internet Engineering Task Force (IETF) , SIP provides for the seamless transmission of voice, fax ,and data across IP and traditional telephone networks. The IETF defines SIP as “a text- based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users. Such sessions include voice, video, chat, interactive games, and virtual reality.” SIP is used to establish peer-to-peer media sessions on an IP network including Internet telephony, conferencing ,instant messaging, and unified messaging. SIP has become a standard IETF protocol for signaling in third-generation mobile networking because it is able to initiate, modify “terminal independent.” This seminar and presentation report deals with basic overview of SIP: Session Initiation Protocol.
SIP – A protocol that allows voice, data, fax, video, instant messaging and even online gaming to be integrated with web-based applications. The session initiation protocol (SIP) is emerging as the favored standard for setting up, modifying and terminating telephone calls over the internet. Its main things on the fly-in the hands of the user.
1.1 Definition
The Session Initiation Protocol (SIP) is an application –layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants.
These sessions include internet multimedia conferences, Internet telephone calls and multimedia distribution. A multimedia session is a set of multimedia senders and receivers and the data streams flowing from senders to receivers. Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these.
1.2 History
SIP has its origins in late 1996 as a component of the “Mbones” set of utilities and protocols. The Mbone or multicast backbone was an experimental multicast network overplayed on top of the public internet. It was used for distribution of multimedia content, including talks and seminar and presentations, broadcasts of space shuttle launches and IETF meetings. One of its essential component was a mechanism for inviting users to listen an ongoing or future multimedia session on the internet. Basically – a session ignition protocol. Thus SIP was born.
As an Mbone tool, SIP was designed with certain assumptions in mind.
First was scalability ► Since users could reside anyware on the internet, the protocol needed to work wide-area from day one. User could be invited to lots of sessions, so the protocol needed to scale in both directions.
A second assumption was component reuse ► Rather than inventing new protocol tools those already developed within the IETF would be used. That includes things like MIME, URLS and SDP. This resulted in a protocol that integrated well with other IP applications(such as web and e-mail).
Despite its historical strengths SIP saw relatively slow progress thought 1999. That’s about when interest in internet telephony began to take off. People began to see SIP as a technology that would also work for VoIP, not just Mbone sessions. The result was an intensified effort towards completing the specification in late 1998, and completion by the end of the year. It received official approval as RFC (Request for Comments, the official term for an IETF standard) in February and issuance of an RFC number, 2543, in March.
Form there, industry acceptance of SIP grew exponentially. Its scalability, extensibility, and most important flexibility appealed to service providers and vendors who had needs that a vertically integrated protocol, such as H.323, could not address. Among services MIC ( particularly MIC’s Henry Sinnreich, regarded as the “Pope” of SIP ) led the evangelical charge. Throughout 1999 and 2000, it saw adoption by most major vendors, and announcements of networks by service providers. Interoperability bake offs were held thought 1999, attendance doubling at each successive event. Tremendous success was achieved in interoperability among vendors. Other standard bodies began to look at SIP as well, including ITU and ETSI, TIPHON, IMTC, soft switch consortium, and JAIN. Looking forward, 2000 will be a year in which real SIP networks are deployed, SIP vendors step forward to announce real products, and applications and services began to appear.
An Internet Engineering Task Force (IETF) standard, SIP is an open, internet genuine protocol for establishing and managing multi party, mixed media sessions over converged networks. SIP enables the creation and deployment of feature rich services that go far beyond simple VoIP calls.
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Session Initiation Protocol (SIP) is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.

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SIP-Session Initiation Protocol.
Allows two end points to establish media sessions with one another.
Developed by IETF (Internet Engineering Task Force)
Incorporates elements of two widely used protocols i.e. HTTP and SMTP.
It takes the client server design from HTTP and text encoding scheme from SMTP.
Introduction (Contd)
SIP supports five facets of establishing and terminating a call:-
User location
User availability
User capabilities
Session set-up
Session handling, including transfer and termination of sessions
History of SIP
Developed by IETF Multi Party Multimedia Control Working Group (MMUSIC).
Achieved Proposed Standard status in March 1999 and was published as RFC (Request for Comments) 2543 in April 1999.
Internet-Draft (I-D’s) containing bug fixes and clarifications to SIP was submitted beginning in July 2000.
This document was then published in RFC 3261.
SIP Entities
There are many SIP entities but the ones in use are-
User agents
Proxy Server
Redirect Server
Registrar or Registration Server
Others include B2BUA’s, Presence Agents, SIP Gateways.
SIP Entities (Contd)
It is a SIP enabled end-device.
Takes input from the user and acts on behalf of the user.
Basically contains two sub-entities:-
User agent Client (UAC)
User agent Server (UAS)
User agent client generates requests and the User agent server generates responses.
SIP Entities (Contd)
It is used to receive requests from a UA or another proxy and acts on behalf of the UA forwarding the request.
It cannot make alterations in the message to be forwarded.
Doesn’t need to understand the request to forward it.
Two types of proxies:-
SIP Entities (Contd)
SIP server that responds to the request but does not forward requests.
Sends back the location information to the caller.
Minimizes the work of forwarding the messages each time by reducing the number of forwards.
SIP Entities (Contd)
Accepts SIP register requests.
Contact information is made available to other SIP servers in the domain.
SIP Entities (Contd)
Other important SIP entities include
Presence Agents-
Used to detect the presence of a user.
Back to Back User Agents(B2BUA)-
They act like proxy servers with the difference that they can modify the SIP messages.
SIP Gateways
They are basically used to connect IP based network to other networks like PSTN, H.323 etc
SIP messages
SIP communication between two entities is through messages.
SIP messages are text-encoded and use SDP (Session Description Protocol).
These messages contain header fields which give the required information about the sender’s address, receiver’s address, maximum forwards possible etc
The two basic types of SIP messages are:-
SIP requests or methods
SIP responses
SIP Requests
INVITE Initiates a call
ACK Confirms a final response from invite
BYE Terminates a call
CANCEL Cancels a searches
OPTIONS Queries the capabilities of the other side
REGISTER Registers with the location service
INFO sends mid session information that does not modify the session state
Other requests include REFER, SUBSCRIBE, NOTIFY, PRACK etc
SIP Responses
Different Classes of SIP responses are shown below:-
1xx – provisional, searching, ringing etc.
2xx – success
3xx – redirection, forwarding
4xx – request failure
5xx – server failures
6xx – global failure
SIP Call Flow Example-1
SIP Call Flow Example-2
VoIP Network Diagram
Advantages of SIP
Significant amount of bandwidth available
Hardware is less expensive as compared to Circuit switched technology
Lower calling costs
Single unified network
New integrated applications
SIP Trunking is a cost effective way to increase productivity while reducing costs, offering the highest quality communication experience.
Complex structure due to various services implemented including voice, video , IM etc.
Generally uses UDP (User Datagram Protocol) for transport which is susceptible to packet loss.
Faces the problem of power interruptions.
Traffic congestion control is also one of the major challenges that SIP faces.
The presence of NAT (Network Address Translator) in a network will result in a nonroutable private IP address in the Via header field due to which the packet will not reach the proper destination.
An uninvited third party, knowing all the SDP information could guess the RTP SSRC number and send unwanted media to either party, so called media spamming.
Comparison with H.323
Future of SIP
SIP has been called the integrated services user part (ISUP) of next-generation networks.
SIP plays a central role in the IP Multimedia Subsystem (IMS), a family of protocols which is supposed to define the architecture of next-generation mobile networks capable of streaming various kinds of text, voice and video data to mobile phone subscribers even as they roam between networks.
SIP Trunking is one of the emerging technologies in the future.
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.pdf   RADVISIONSIPProtocolOverview.pdf (Size: 112.03 KB / Downloads: 24)


The Session Initiation Protocol (SIP) is a signaling protocol for initiating,
managing and terminating voice and video sessions across packet networks. SIP
sessions involve one or more participants and can use unicast or multicast
communication. Borrowing from ubiquitous Internet protocols, such as HTTP
and SMTP, SIP is text-encoded and highly extensible. SIP may be extended to
accommodate features and services such as call control services, mobility,
interoperability with existing telephony systems, and more.
SIP is being developed by the SIP Working Group, within the Internet
Engineering Task Force (IETF). The protocol is published as IETF RFC 2543
and currently has the status of a proposed standard.
This section describes the key constituents of SIP.
SIP ENTITIES A SIP network is composed of four types of logical SIP entities. Each entity has
specific functions and participates in SIP communication as a client (initiates
requests), as a server (responds to requests), or as both. One “physical device”
can have the functionality of more than one logical SIP entity. For example, a
network server working as a Proxy server can also function as a Registrar at the
same time.

Following are the four types of logical SIP entities:


In SIP, a User Agent (UA) is the endpoint entity. User Agents initiate and
terminate sessions by exchanging requests and responses. RFC 2543 defines the
User Agent as an application, which contains both a User Agent client and User

Agent server, as follows:

 User Agent Client (UAC)—a client application that initiates
SIP requests.
 User Agent Server (UAS)—a server application that contacts
the user when a SIP request is received and that returns a
response on behalf of the user.
Some of the devices that can have a UA function in a SIP network are:
workstations, IP-phones, telephony gateways, call agents, automated answering


A Proxy Server is an intermediary entity that acts as both a server and a client
for the purpose of making requests on behalf of other clients. Requests are
serviced either internally or by passing them on, possibly after translation, to
other servers. A Proxy interprets, and, if necessary, rewrites a request message
before forwarding it.

A Redirect Server is a server that accepts a SIP request, maps the SIP address
of the called party into zero (if there is no known address) or more new
addresses and returns them to the client. Unlike Proxy servers, Redirect Servers
do not pass the request on to other servers.
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