What is the SIP protocol? What is the working mechanism?

With the development of the network, VoIP services and unified communication platforms become more and more important. The SIP protocol is an exciting focus. Some may ask, what is the SIP protocol? What's better about it? Here we will go into more details.

The Chinese meaning of the SIP protocol is the process initialization protocol. The existence of the SIP protocol is mainly to promote more personalized user communication systems, more choices, better connections, and more open and compatible, A communication protocol. The SIP protocol is developed by the international Internet standard-making organization, a non-profit global Organization. Currently, the SIP protocol has been adopted by major telecom service providers around the world. SIP is also used for call control of the next-generation 3GPP mobile communication network. Microsoft uses it for Xbox and Windows Messenger. AOL and many other companies use it in instant messaging systems.

Since the birth of the SIP protocol, it adopts a similar pattern as HTTP. Like HTTP, it is also based on an IP network. One of the most influential advantages of the Internet is that various applications can be deployed between network servers and browsers, instead of relying on IP networks. This is also the case for processes based on the SIP protocol. When the user starts to contact other Parties, the process starts, the contact ends, and the process ends ). SIP protocol servers, such as mindi's 3050 integrated communication platform) and users such as mindi's 5055 SIP phones, can fully control the voice, video, conference, and instant messaging of the entire process ). This is in stark contrast to the service mode provided by the traditional circuit switching mode. In the traditional mode, terminal products such as phones do not have call control functions, and all service functions are controlled in the central switching unit. The SIP Protocol provides the maximum flexibility for enterprise management personnel and system support.

Compared with the previous VoIP protocol, the SIP protocol increases the scalability and collaboration of communication systems to a new level, on which many new services can be established more easily.

Now we understand what the SIP protocol is, but to gain a deeper understanding of the protocol, we also need to look at its working principles.

Implementation Mechanism of the SIP protocol

SIP is a layered protocol, which means that its behavior is described based on a set of equal and independent processing stages. Each stage is loosely coupled. the hierarchical description of the protocol is used to express the function, so that the description of the function can span several elements in one part. it does not specify any implementation method. when we say that an element contains a layer, we mean that it complies with the Rule Set defined by this layer.

Not every element specified by the Protocol contains layers. moreover, the elements specified by SIP are logical elements, not physical elements. A physical implementation can be selected as different logical elements, or even based on transactions.

The bottom layer of SIP is syntax and encoding. Its Encoding is defined by enhanced Backus-Nayr formal syntax BNF.

The second layer is the transport layer. It defines how a client on the network sends requests and receives responses, and how a server receives and sends responses. All the SIP elements include the transport layer.

The third layer is the Transaction layer. transactions are the basic elements of SIP. A transaction is composed of a request sent by a client transaction to a server transaction using the transport layer) and all the responses sent to the client from the server transaction that should be requested. the transaction layer processes Application Layer retransmission, matches the response to the request, and the application layer times out. any user agent client UAC) to complete the task using a set of transactions generated. the user proxy contains a Transaction layer, and stateful proxies also have. the stateless proxy does not include the Transaction layer. the transaction layer has a client component called a client transaction) and a server component is called a server transaction. Each component represents a finite state machine and is constructed to process specific requests.

The layer above the Transaction layer is called the Transaction user TU ). every SIP entity, except the stateless proxy, is a Transaction user. when an TU wants to send a request, it generates a client transaction instance and transmits the request and IP address, port, and transmission mechanism used to send the request to it. A client transaction generated by a TU can also be deleted. when the client cancels a transaction, it requests the server to stop further processing, restore the State to before the transaction initialization, and generate a specific error response to the transaction. this is completed by the CANCEL request, which constitutes its own transaction, but involves the transaction to be canceled.

SIP uses an EMAIL address to indicate the user address. each user is identified by a hierarchical URL, which constructs elements such as a user's phone number or host name for example: SIP: user@company.com ). because of its similarity with the EMAIL address, SIPURLs is easy to associate with the user's EMAIL address.

SIP provides its own reliability mechanism to be independent from the grouping layer, and only requires unreliable data packet service. SIP can be typically used over UDP or TCP.

SIP provides necessary protocol mechanisms to ensure that the terminal system and proxy server provide the following services:

◆ User Positioning

◆ User capability

◆ User availability

◆ Call Creation

◆ Call handling

◆ Call forwarding, including: 1) equivalent to 800 calls; 2) call forwarding without response; 3) call forwarding before busy calls; 4) Unconditional call forwarding

◆ Call number transfer, which can be any naming mechanism.

◆ Individual mobility, such as reaching the called party through a single location-independent address, even if the called party changes the terminal.

◆ Negotiation and selection of terminal types: the caller can give a selection of how to reach the other party, such as through Internet phone, mobile phone, or response service.

◆ Terminal capability negotiation

◆ Caller and authorized caller Authentication

◆ Uninformed and guided call transfer

◆ Multicast meeting invitation

When a user wishes to call another user, the caller uses INVITE to request an initial call. The request contains sufficient information for the caller to participate in the session. if the client knows the location of the other party, it can directly send the request to the IP address of the other party. if you do not know, the client sends the request to the locally configured SIP network server. if the server is a proxy server, it will parse the location of the called USER and send the request to them. there are many ways to complete the previous step, such as searching DNS or accessing the database. the server can also be a redirection server, which can return the location of the called user to the called client for direct contact with the user. in the process of locating the user, the SIP network server can, of course, proxy or redirect the call to another server until it reaches a server that explicitly knows the IP address of the called user.

Once a user address is found, the request is sent to the user. Several options are generated. in the simplest case, the user's phone client receives the request-that is, the user's phone rings. if the user accepts the call, the customer uses the specified capability of the client software to respond to the request and establish a connection. if the user rejects the call, the session will be redirected to the voice mail server or another user. for "specify capabilities", refer to the features you want to enable. for example, the client software can support video conferencing, but the user only wants to use audio conferencing, then only the audio function is enabled.

SIP also has two important features. the first is that the stateful SIP proxy server can split incoming calls or replicate incoming calls, so that several extension branches can run simultaneously. the branch of the first response receives the call. this feature is very convenient when a user is working in two locations, such as a lab or an office) or at the same time, the manager and his/her secretary are ringed.

The second feature is the unique ability of SIP to return different media types. for example, you can contact the company. when the SIP server receives a connection request from the client, it can return to the client through the WEB interactive voice response page, this page has available department branches or users on the list. after clicking the appropriate link, a request is sent to the selected user to create a call.

I believe that after reading the article, we have understood what the SIP protocol is and have mastered the working mechanism of the SIP protocol. We hope to provide a valuable reference for your future work and study.