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Session the Initiation Protocol (SIP) is a network protokol for the setting up a communication meeting between two and more participants. Minutes are specified in the RFC 3261 (in former times RFC 2543). In the IP-Telefonie is the SIP frequently used minutes.

Function mode

Contrary to H.323, which comes from the ITU-T, SIP with view of Internet was developed of the IETF and orients themselves at the architecture of usual applications of Internet. To easy capable of being implementedness, scaling barness, expandabilities and flexibility one paid attention from the beginning.

To be used SIP can, in order to administer arbitrary sessions with one or more participants. It is not limited to Internet Telefonie, but sessions can be arbitrary conferences, computer games etc.

Since by an SIP address the current IP address participants can be determined, also the possibility is offered that one will be attainable over an address in the future, which can be used then both for E-Mail and Telefonie.

In order to lead however an Internet telephone call, one needs more than only SIP. SIP serves only to make communication possible - the actual data for communication must be exchanged over other, minutes suitable for it. For this session the Description Protocol (SDP, RFC 2327) is used and real-time transport Protocol (RTP, RFC 3550). SDP serves for it, those between the terminator points for using codecs to negotiate transportation minutes etc. Task of RTP is it to transport the Multimedia data stream (audio, video, text etc.) i.e. of the codecs and compressed data coded to package and over UDP dispatch.
SIP is based among other things on HTTP minutes - it uses a similar header structure and is likewise text-based minutes. For the way of writing of the participant addresses the URI format well-known of E-Mail is used: "“sip: user@domain"”. A further address mechanism is tel the URI in RFC the 2806 is described. Bsp: "“tel: +49 69 1234567"”. This can be changed if necessary into a SIP URI Bsp: "“sip: +49-69-1234567@domain"”.
Support finds SIP already in many devices of various manufacturers and it seems to standard minutes for Voice of over IP (VoIP) to develop. SIP was selected also by the 3rd generation Partnership Project (3GPP) as minutes for  in the 3G-Mobilfunk (UMTS). Also the specification NEXT generation network (NGN) with the European Telecommunications standard of institutes (ETSI) project group Telecommunications and Internet converged services and Protocols for Advanced Networking (TISPAN) is based upon SIP.
Pro and cons
To the advantages of SIP it belongs that it concerns here an open standard, which found meanwhile very far spreading contrary to Skype, which represents a closed and  system. With Skype there is Ltd. with the central announcing server of the Skyper. "“single POINT OF failures"”. If a DDoS attack against this server should take place, no more connections can be developed in the Skype net. Since SIP servers are distributed, such an attack concerns only the respective offerer and not the entire Telefonie obtained over SIP. A further advantage of SIP is the possibility of being able to modify a meeting already established. In addition simply within the meeting a further INVITE Message with the new SDP Sitzungseigenschaften is sent to the opposite side. Thus a new medium can be modified and/or removed be added or an existing medium. The appropriate message is called also rh-INVITE Request.
A disadvantage from SIP is that it falls back for the transmission of the language data to RTP. For it used UDP haven are assigned dynamically, which makes the use of SIP in connection with Firewalls or network ADDRESS translation (NAT, RFC 2663) difficult, since the most Firewalls and/or NAT routing cannot assign dynamically assigned haven to the signaling connection. Remedy for this problem creates the employment of STUN (simple Traversal OF UDP of over NATs), which NAT routing, in addition, other minutes such as IAX (InterAsterisk eXchange) recognizes and penetriert. IAX combines signaling and medium data on a UDP connection. Like H.323 IAX is binary minutes, why the elimination of errors is more difficult than with SIP. In addition IAX is not standardized and only badly documented.
Newer minutes of the IETF for the solution of the NAT Traversal problem represent Interactive Connectivity establishment (ICE), which is already supported by some SIP Clients, and usually by firmware Upgrade to be installed can. 
So-called Application Layer gateway (ALG) represents a further solution for the NAT Traversal problem. These are inserted SIP Proxys, which installs on NAT routing and/or a Firewall, for smooth transfer of SIP signaling and - medium stream ensure. A ALG can provide with SIP telephone calls automatically for the opening the necessary haven on Firewall and mark RTP  with DiffServ bits, whereby these can be transported with higher priority over Internet. 
Further many services, which admits from the classical Telefonie is, can be illustrated not directly with SIP minutes. They are defined as extensions to SIP minutes. Manufacturers of SIP solutions (hard and Softphones, exchange technique and SIP Firewalls) implement usually only one part of all of these extensions, so that always all services with each terminal cannot be supported.
One of these extensions is SIP for Business SIP-B, which makes many switching-oriented capability characteristics for the classical branch technology possible with SIP. Even services like the chief secretary function are possible with SIP-B. With classical branches these services are implemented over  minutes, so that one must buy special system telephones, which match perfectly the used branch. With the change to SIP based branch needs one no more such system telephones, must however instead whereupon it respects that the used SIP telephones support SIP-B, which at present only with very few models the case is.
The Early dial functionality of SIP, over which message 484 - Incomplete ADDRESS implements, sends with each depressing the key a INVITE Message to the server, as long as to the entered address and/or number by this can be dissolved. Still not all SIP Clients support and - servers this function.
See also
  • IP-Telefonie: Current solutions for a list of existing implementations
  • SIP telephone
  • Telephones NUMBER Mapping (ENUM)
  • IP increase in value service
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Articles in category "Session Initiation Protocol"
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