What is SIP

SIP (Session Initiation Protocol) :-

It is one of the signaling protocol as SS7 & H.248,

It is an application layer protocol that can extablish, modify & terminate sessions or calls. These sessions include multimedia conference, internet telephony, & similiar applications.

SIP is one of the key protocol that implements voice-over IP (VOIP).

It is a signalling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session. The ability to establish these sessions means that a host of innovative services become possible, such as voice-enriched e-commerce, web page click-to-dial, Instant Messaging with buddy lists, and IP Centrex services.

SIP supported services -

• Name Mapping
• Redirection
• ISDN Services
• Intelligent Network (IN) services.
• User location
• User capabilities
• User availability
• Call set-up
• Call handling
• Call forwarding
• Call-forwarding no answer
• Call-forwarding busy
• Call-forwarding unconditional
• Other address-translation services
• Callee and calling "number" delivery, where numbers can be any (preferably unique) naming scheme
• Personal mobility, i.e., the ability to reach a called party under a single, location-independent address even when the user changes terminals
• Terminal-type negotiation and selection: a caller can be given a choice how to reach the party, e.g. via Internet telephony, mobile phone, an answering service, etc.
• Terminal capability negotiation
• Caller and callee authentication
• Blind and supervised call transfer
• Invitations to multicast conference

It support 5 facets of extablishing & terminating multimedia communications :-

1. User Location - Detemining end system to be used for communication.
2. User Capabilities - Determining the media & media parameters tobe used.
3. User Availability - Determining the willingness of called party to engange in communication.
4. Call Setup - Sending ringback tone to the called party & estabilishing call parameters at both end called & calling party.   
5. Call Holding & Control - Includes redirection, transfer & termination of calls.

SIP can also initiate multi-party calls using multipoint control unit (MCU) or fully meshed interconnection.
Internet Telephony gateway that connects PSTN parties can also use SIP to setup calls between them.
SIP can use User Datagram Protocol (UDP) & Transmision Control Protocol (TCP) as transport protocol, UDP is preferred.

Architecture
There are two basic components within SIP:

• SIP user agent.
• SIP network server.

The User Agent is the end system component for the call. The user agent itself has a client element, the User Agent Client (UAC) and a server element, the User Agent Server (UAS). The client element initiates the calls and the server element answers the calls. This allows peer-to-peer calls to be made using a client-server protocol.

SIP user agents can be lightweight clients suitable for embedding in end-user devices such as mobile handsets or PDAs. Alternatively, they can be desktop applications that bind with other software applications such as contact managers.


The SIP server is the network device that handles the signalling associated with multiple calls. The main function of the SIP servers is to provide name resolution and user location, since the caller is unlikely to know the IP address or host name of the called party, and to pass on messages to other servers using next hop routing protocols.

SIP servers can operate in two different modes: stateful and stateless. The difference between these modes is that a server in a stateful mode remembers the incoming requests it receives, along with the responses it sends back and the outgoing requests it sends on.

A server acting in a stateless mode forgets all information once it has sent a request. These stateless servers are likely to be the backbone of the SIP infrastructure while stateful-mode servers are likely to be the local devices close to the user agents, controlling domains of users.

 
SIP Addressing :-
Uniform Resource Locator (URL) are used within SIP messages to indicate the originator (FROM), current destination (requested URL), final destination (TO) of a SIP request & to specify redirection address (Contact).

SIP URL has a Syntax :-

SIP:User:password@host:port;transport-param|user-param|method-param|ttl-param|maddr-param|other-param

Their meaning -

• SIP - indicates SIP is used for communication with a specified end system.
• User - Consists of any characters in the form of email address or telephone number.
• Password - can be included but not recommended because of security risk.
• Host - can be host(other user) domain name or IP address.
• Port - indicates port number to which request is sent, default is 5060, a public SIP port number.
• Transport-Param - Indicates which transport protocol to be used, TCP or UDP, default is UDP.
• User-Param - can be a telephone number, 2 values are available for this field, IP & Phone number, when field is set to "phone" username is telephone number & corresponding end system is an IP Telephony Gateway. 
• Method-Param - Specifies method or operation to be used.
• TTL-Param - Designates the Time-To-Live (TTL) of UDP multicast data packet. It is valid only when transport parameter is UDP & Maddr parameter is "Multicast Address".
• Maddr-Param - Provides the server address to be contacted for a user, overriding the address supplied in the host field. This address is typically a multicast address.

NOTE - The following parameters are optional
Transport-Param, User-Param, Method-Param, TTL-Param, Maddr-Param, Other-Param.


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