VOIP Service SIP Protocol

What is SIP?

SIP stands for Session Initiation Protocol. It is an internet protocol which has been developed and designed within the Internet Engineering Task Force (IETF).

The protocol is used for establishing, and controlling sessions with one or more devices (for example PCs, Telephones or PDA). Examples of a session can include Voice Over IP (VoIP) calls, Instant Messaging, distributed computer games, etc.

SIP should not be confused with H.323. Though similar in concept, and both have been used in initiating and managing communications, the H.323 protocol was actually developed as part of the International Telecommunication Union (ITU) and targeted at videoconferencing over ISDN lines. Because of this limitation in scale, the H.323 protocol is not used widely for Voice Over IP services.

SIP is not the only protocol that the communicating devices will require in order to successfully route a VoIP call. The purpose of SIP is just to initiate and control the communication, the communication itself occurs via other protocols. Two of the most common protocols that are used for VoIP along with SIP are Real Time Protocol (RTP) and Session Description Protocol (SDP)

• RTP: In association with several codecs that can convert a user’s voice into computer data, the RTP protocol is used to carry the real-time multimedia data.
• SDP: This is the protocol by which the capabilities and available codecs upon the participant’s devices are exchanged.

A SIP session is initiated by a client (sometimes called a User Agent Client – or UAC), and is routed by entities in the network to the receiving client. The client can take on the form of a piece of software on a Personal Computer, or hardware such as an IP phone, or even a traditional phone with an Analog Telephone Adapter (ATA).

SIP has three major goals that are essential in ensuring that communications can occur in a consistent and simple manner. These are as follows:

Name resolution and routing
SIP clients are identified by a unique address, commonly known as a Uniform Resource Identifier (URI). This URI can take many forms, but in practice it is typically similar to a phone number or an email address. Name resolution takes sessions targeted for a URI, and maps that to a specific UAC, and routes the request via network servers that understand how to route SIP.

Capability negotiation
UAC’s will have different capabilities, such as the ability to have high or low quality audio, or the ability to guarantee a certain quality of service. As part of the session initiation, these capabilities can be negotiated between the UACs, such that a consistent set of capabilities are agreed upon.

Participant management
Within a session there are typically two participants (or two independent UACs). However there are cases whereby additional participants may be required to be part of the session. An example of this would be establishing a conference session and adding additional users. SIP provides mechanisms to control who can include additional participants, and how they can be added or removed.

SIP makes VoIP possible, not only technically but also making it usable and cost effective. The use of SIP allows your voice and data to travel on the same internet connection, and therefore not require a dedicated phone line. This is the main reason why VoIP can save you $500 a year on your home bill.