Voice over IP has a structured network that can lay its foundation on, but it lacks many other important features. One of them is a reliable protocol. But this problem is on the way to being solved. SIP (Session Initiation Protocol) is the name of the protocol that seems to be on the lips of every VoIP products manufacturer.
SIP (Session Initiation Protocol)
Feature of SIP VoIP is that it offers the possibility for users to start and receive communications and services from any location and for networks to identify the users wherever they may be.
SIP analyzes requests from clients and retrieves responses from servers. Interlocutors are identified by SIP URLs. SIP decides the end system to be used for the session, the communication media and media parameters, as well as the called party’s desire to take part of the communication. SIP then sets call parameters at either end of the communication, handles call transfer and termination. A great feature of SIP is that it offers the possibility for users to start and receive communications and services from any location and for networks to identify the users wherever they may be.
SIP is an alternative protocol to H.323
SIP is an alternative protocol to H.323 – the IP videoconference transmissions protocol that’s been in use since its approval in 1996. Up until now, H.323 has been the VoIP industry standard, but its leadership has been questioned by the wide implementation of SIP in the past year(s).
Multicast conference and media integration
– SIP also provides you with the possibility of adding participants to an already existing session, such as a multicast conference. Also, media can be added to (and removed from) an existing session.
– SIP supports personal mobility
– SIP supports name mapping and redirection services, which supports personal mobility – users can maintain a single externally visible identifier regardless of their network location.
In order to safely and reliably establishing and terminating multimedia communications,
SIP supports five facets:
* user location (for deciding which end system is to be used for the communication);
* user availability (for determining whether the called party is willing to engage in communication);
* user capabilities (for determining the media and media parameters that are to be used in the communication);
* session setup (“ringing”, for establishing the communication session parameters at both ends) and session management (that includes data packets transfer and termination of sessions,
* modifying session parameters, and invoking services).
Complete multimedia architecture
SIP can be looked at as a component that can be used with other IETF protocols to build a complete multimedia architecture, such as the Real-time Transport Protocol (RTP) for transporting real-time data and providing Quality of Service feedback or the Real-Time streaming protocol (RTSP) for controlling delivery of streaming media. That is why SIP should be used in conjunction with other protocols in order to provide best services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols.