SIP, the session initiation protocol, is the IETF protocol for VOIP and different textual content and multimedia sessions, like instantaneous messaging, video, on-line video games and different services.

Abstract from the RFC 3261 (formatted_and_explained version) – SIP: Session Initiation Protocol

This report describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions encompass Internet cellphone calls, multimedia distribution, and multimedia conferences.

SIP invites used to create classes elevate session descriptions that permit members to agree on a set of compatible media types. SIP makes use of elements referred to as proxy servers to assist route requests to the user’s modern location, authenticate and authorize customers for services, put in force provider call-routing policies, and grant aspects to users. SIP additionally offers a registration characteristic that allows customers to upload their modern areas for use through proxy servers. SIP runs on top of various one-of-a-kind transport protocols.


SIP is very plenty like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for telephone calls are described in SDP -the session description protocol.

SIP is a text-based protocol that uses UTF-8 encoding
SIP makes use of port 5060 each for UDP and TCP. SIP may additionally use other transports
SIP presents all potentialities of the common Internet Telephony facets like:

call or media transfer
call conference
call hold
Since SIP is a bendy protocol, it is viable to add more elements and hold downward interchangeability.

SIP also does go through from NAT or firewall restrictions. (Refer to NAT and VOIP)

SIP can be considered as the enabler protocol for telephony and voice over IP (VoIP) services. The following points of SIP play a predominant role in the enablement of IP telephony and VoIP:

Name Translation and User Location: Ensuring that the name reaches the referred to as celebration at any place they are located. Carrying out any mapping of descriptive information to region information. Ensuring that small print of the nature of the call (Session) are supported.
Feature Negotiation: This lets in the team concerned in a call (this may also be a multi-party call) to agree on the features supported recognizing that no longer all the events can support the equal degree of features. For instance video may or may not be supported; as any shape of MIME kind is supported via SIP, there is plenty of scope for negotiation.

Call Participant Management: During a call a participant can convey different users onto the name or cancel connections to different users. In addition, customers could be transferred or positioned on hold.
Call function changes: A user should be able to change the call characteristics throughout the direction of the call. For example, a name can also have been set up as ‘voice-only’, but in the path of the call, the customers might also want to enable a video function. A 1/3 party joining a name may additionally require special points to be enabled in order to participate in the call
Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable choice of the splendid codec for establishing a name between the quite a number devices. This way, much less advanced devices can take part in the call, supplied the suitable codec is selected.

The SIP protocol

The SIP protocol defines countless methods.