Session Protocol Initiation

Technical presentation

Session Initiation Protocol (of which the initials is SIP ) is a protocol standardized and standardized by the IETF (described by the RFC 3261 which makes obsolete the RFC 2543, and supplemented by the RFC 3265) which was conceived to establish, modify and finish sessions multi-media. It takes care of the Authentification and the localization of the multiple participants. It also takes care of the negotiation on the types of Média usable by the various participants in encapsulating messages SDP (Session Description Protocol). SIP does not transport the data exchanged during the session like the video voice or it. SIP being independent of the transmission of the data, any type of data and protocols can be used for this exchange. However protocol RTP (Real-time Transport Protocol) generally ensures the audio and video sessions. SIP replaces H.323 gradually.

Context

Per hour of the revolution of information by the use of the mother of all networks, Internet, one attends the convergence of telephony and data processing (Couplage telephony-data processing).

The STN and ISDN (ISDN, “Numéris” being the trade name given by France Telecom, then operator of state) and by extension the telephones of first and second generation (satellite and GSM) made their time as networks supporting the voice only, with band-widths and very low flows.

H.323 was a protocol pioneer of telephony on IP, emanating from the authorities of the world of telecommunications (UIT or ITU). SIP, from design a little more recent, comes from the world of the Internet (IETF) and is undoubtedly integrated a little better on the networks IP.

The editors, service providers and operators putting of time to offer software and services around H323 and SIP, the Skype firm made its hole with several million users by proposing the software customers, the service and the footbridge towards the entering and outgoing public networks.

Currently, to fill the weaknesses in term of presence and networks (NAT) of protocol SIP, of the editors and service providers planchent on the integration of SIP with Jabber, protocol reliable, standard, opened and tested presence and of Instant messaging.

Operation

Bases

SIP shares many similarities with the protocol HTTP like coding in ASCII and the codes of answer.

The customer sends requests to the waiter, which returns an answer to him. The basic methods are:

*INVITE makes it possible a customer to require a new session

*ACK confirms the establishment of the session

*CANCEL cancels one INVITES

*BYE outstanding finishes a session in progress

The codes of answer are similar to HTTP.

*100 Trying

*200 OK

*404 Not Found

The codes equal to or higher than x80 are specific to SIP.

*180 Ringing

*486 Busy

*etc.

On the other hand, SIP differs from HTTP owing to the fact that an agent SIP (To use Agent, UA) plays usually at the same time the parts of customer and from waiter. I.e. it can as well send requests, that to answer those which it receives.

In practice, the installation of SIP rests on three elements: To use Agent, registrar and proxy.

To use Agent

The User Agents indicate the agents which one finds in the telephones SIP, the softphones (telephone software on IP) of the computers and PDA or footbridges SIP. In theory, one can establish sessions directly between two User Agents , two telephones for example. But that requires to know the Adresse IP of the recipient. That is not the ideal because an address IP can not be public (behind a NAT) or change and it is much more complicated to retain than a URI. User Agents can thus be recorded near Registrars to announce their site running, i.e. them Adresse IP.

Registrar

The Registrar is a waiter which manages requests REGISTER sent by the Users Agents to announce their current site. These requests thus contain a Adresse IP, associated with a URI, which will be stored in a Database.

URI SIP are very similar in their form with addresses email: sip: utilisateur@domaine.com

Generally, of the mechanisms of Authentification allow to prevent that whoever can be recorded with any URI.

Proxy

A Proxy SIP is used as intermediary between two User Agents which does not know their respective sites (Adresse IP). Indeed, association URI-Addresses IP was stored beforehand in a Database by a Registrar . Proxy can thus question this Database to direct the messages towards the recipient.

Proxy is satisfied to relay only messages SIP to establish, control and finish the session. Once the established session, the data, for example a flow RTP for the VoIP, does not forward by the Proxy waiter. They are exchanged directly between the User Agents .

Induced characteristics of the SIP

One finds in the argumentations for or against the SIP the following elements:

Open: the protocols and official documents are detailed and accessible to all in remote loading
Standard: the IETF standardized the protocol and its continuous evolution by the creation or the evolution of other protocols which function with SIP
Basé on address IP: that induces that the SIP does not cross the NAT (but that can be solved by deploying additional customer-server mechanisms like STUN or by coupling SIP with Jabber)
logical Intégration has of another standard: Integration has jabber makes it possible for example to circumvent the problems related to addressing IP.
Very similar to HTTP
Compatible P2P: on a LAN, SIP functions completely in P2P (even more easily with the integration of Zeroconf), it is not the case for the establishment of sessions between two pars separated by a NAT
Flexible: SIP is also used for any multi-media type of sessions (voice, video, but also music, virtual reality, etc)
Téléphonie on public networks: there exist many footbridges (subscription-based services) towards the public network of telephony (STN, GSM, etc) making it possible to emit or receive vocal calls
Common points with H323: the use of protocol RTP and some codecs its and video are joint
Mauvaise implementation: a bad implementation or an incomplete implementation of protocol SIP in the User Agents can disturb operation or generate superfluous traffic on the network.
incompatibilities: H323 (standard and open) and Skype (owner) profit from their respective network effect, although H323 tends to disappear with the profit from SIP
Présence and instant messaging: SIP shows a certain number of weaknesses in the management of the presence and the instant messaging, but the integration of the open standard specialized Jabber solves the majority of these problems
Faible many users: SIP is known still little and used by the general public, not having reached a critical mass, it does not profit from the effect protected network
Non: login/password in light.

Aspects of the market

The SIP was adopted now by all the large actors of industry as a protocol of choice for the evolution of the networks. Tendency clearly established on a worldwide scale since ~2003-2004.

Retained by AOL, Yahoo, Microsoft, Wanadoo, Orange, Iliad, etc
Retained by Microsoft for MSN and Microsoft Office Live Communication Server (LCS/OCS)
Retained by the 3GPP Multi-media IP Subsystem - TISPAN for its ecosystem of services and indication
Retained by all the large traditional suppliers of PABX: Alcatel-Lucent, Nortel, Cisco, Avaya, Ericsson, Siemens, etc
Retained by all the suppliers of PABX open-source: Asterisk, IPTel, etc
Retenu in Voix architectures on IP of convergence fixes mobile where the services of presence are a strong requirement.

External bonds

Information on protocol SIP
work of the IETF on SIP
the page of voip-info.org devoted to the SIP

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