Basic explanation: What is the SIP protocol?

In the process of establishing a session, we must apply it to the SIP protocol. Some friends may not understand this agreement. It doesn't matter. Here we will explain the content of this Protocol. You may be familiar with the SIP protocol. The Session Initialization Protocol is an application layer control protocol. So what are more specific functions and applications? It can be used to create, modify, or terminate multimedia sessions, such as Internet phone calls. SIP can invite participants to an existing session, such as a multicast meeting. You can add or delete media in an existing session. The SIP protocol supports name ing and redirection services, and supports user mobility. No matter where your network is located, you only need to maintain a single external visual identifier.

SIP supports the creation and termination of multimedia communication in five aspects:

User positioning: Determine the terminal system used for communication;

User availability: determines whether the called party is willing to join the communication;

User capability: determines media and media parameters;

Call Establishment: the establishment of the session parameters of the caller and the called party;

Call Management: includes transferring and terminating sessions, modifying call parameters, and calling services.

The SIP protocol can be combined with other IETF protocols to establish a sound multimedia structure, such as providing real-time data transmission and service quality QOS) Feedback of real-time transmission protocol RTP) real-time stream protocol RTSP that provides streaming media transmission control, and provides gateway-controlled media gateway control protocol MEGACO for Public Switched Telephone Network PSTN ), and the Session Description Protocol SDP that describes multimedia sessions ). Therefore, SIP must work together with other protocols to provide complete services for users. However, the basic functions and operations of SIP do not depend on these protocols.

SIP provides a set of security services, including anti-denial-of-service attacks, user authentication for users and proxy for users), Integrity Protection and encryption and privacy services.

The SIP protocol supports both IPv4 and IPv6. For Internet phone conversations, SIP does the following:

Identify the caller and callee through the SIP address. When a SIP call is established, the caller first locates the appropriate server and then sends a SIP request. The most common SIP behavior is invitation. The SIP request is redirected by the proxy or triggered by a series of new SIP requests, rather than directly reaching the target called party. Users can register their location through the SIP server. Sip url) can be embedded into the web page, so you can click it to establish a call session with the other party.

Protocol Structure

SIP information can be transmitted over TCP or UDP. SIP information is text-based and uses the ISO 10646 character set in UTF-8 encoding. Each line of information must be terminated through CRLF. Most information methods and header fields are similar to HTTP. The SIP information can be either request information or response information.

A request has the following format:

Method

Request URI

SIP version

Method: The Method executed on the resource. Possible methods include: Invite, Ack, Options, Bye, Cancel, and Register.

Request-URI: a sip url or a universal URI. A Request is to be addressed to a user or service.

SIP version-the version of the active SIP.

For the Response Header Format, see:

SIP version

Status code

Reason phrase

SIP version-the version of the active SIP protocol.

Status-code-three-digit integer result code, used to try to understand and meet the request requirements.

The original description of Reason-phrase-status code.