What is the SIP protocol?

SIP (Session Initiation Protocol) is a multimedia communication protocol formulated by IETF (Internet Engineering Task Force). It is a text-based application layer control protocol, which is used to create, modify and release sessions of one or more participants. It is widely used in CS (Circuit Switched), NGN (Next Generation Network) and IMS(IP Multimedia Subsystem) networks, and can support and apply to multimedia services such as voice, video and data, as well as special services such as presentation and instant messaging. It can be said that where there is an IP network, there is a SIP protocol. [1-2] SIP is similar to HTTP. SIP can reduce the development time of applications, especially advanced applications. Because IP-based s IP uses IP network, fixed network operators will gradually realize the far-reaching significance of SIP technology to them. Chinese name session initiation protocol mbth session initiation protocol appeared in the mid-1990s, and IETF published it in 1999 directory 1 session protocol.

Compression mechanism? App application

2 development process

3 communication requirements

4 Conversational composition

User agent? Register the server? Proxy server? Redirect server

5 common messages

6 protocol comparison

Standard application target? Standard architecture? System composition structure? How easy is it to achieve? summary

7 related technologies? Open source project? 5Java 1 Session Protocol SIPSIP (Session Initiation Protocol) is an application layer signaling control protocol. Used to create, modify and release sessions for one or more participants. These sessions can be Internet multimedia conferences [3], IP telephony or multimedia distribution. Participants in a conversation can communicate through multicast, unicast or a mixture of the two. SIP interoperates with Resource Reservation Protocol (RSVP) which is responsible for voice quality. It also cooperates with several other protocols, including Lightweight Directory Access Protocol (LDAP) for location, Remote Authentication Dial-in User Service (RADIUS) for authentication and RTP for real-time transmission. An important feature of SIP is that it does not define the type of session to be established, but only defines how the session should be managed. This flexibility means that SIP can be used in many applications and services, including interactive games, music and video on demand, as well as voice, video and web conferencing. SIP messages are text-based, so they are easy to read and debug. For designers, the programming of new services is simpler and more intuitive. SIP reuses MIME type descriptions like e-mail clients, so session-related applications can start automatically. SIP reuses several existing mature Internet services and protocols, such as DNS, RTP, RSVP, etc. There is no need to introduce new services to support the SIP infrastructure, because many parts of the infrastructure are already in place or ready-made. The extension of SIP is easy to define and can be added to new applications by service providers without damaging the network. The old devices based on SIP in the network will not hinder the new services based on SIP. For example, if the old SIP implementation does not support the method/header used by the new SIP application, it will be ignored. SIP is independent of the transport layer. Therefore, the underlying transport can be IP using ATM. SIP uses User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) to flexibly connect users independent of the underlying infrastructure. SIP supports multi-device function adjustment and negotiation. If a service or session starts video and voice, you can still transmit voice to devices that don't support video, or you can use other device features, such as one-way video streaming. Communication providers, their partners and users are increasingly eager for a new generation of IP-based services. Now with SIP (Session Initiation Protocol), the urgent need is solved. SIP is an idea born in the computer science laboratory less than ten years ago. It is the first protocol suitable for multi-user sessions of various media contents, and has now become the specification of the Internet Engineering Task Force (IETF). Today, more and more operators, CLEC (competing local operators) and ITSP(IP telephone service provider) are providing SIP-based services, such as local and long-distance telephone technology, online information and instant messaging, IP Centrex/Hosted PBX, voice messaging, push-to-talk, multimedia conferencing and so on. Independent software vendors (ISV) are developing new development tools to build SIP-based applications and SIP software for carrier networks. Network equipment vendors (NEV) are developing hardware to support SIP signaling and services. Now many IP phones, user agents, network proxy servers, VOIP gateways, media servers and application servers are using SIP. SIP evolved from similar authoritative protocols, such as Web Hypertext Transfer Protocol (HTTP) formatting protocol and Simple Mail Transfer Protocol (SMTP) e-mail protocol, and developed into a powerful new standard. However, although SIP uses its own unique user agent and server, it does not work in isolation. SIP supports the provision of converged multimedia services and cooperates with many existing protocols responsible for authentication, location information, voice quality, etc. SIP is flexible, extensible and open. It has inspired the Internet, fixed and mobile IP networks to launch a new generation of services. SIP can complete network messages on multiple PCs and telephones, and simulate the Internet to establish a session. Unlike the long-standing international telecommunication union (ITU) SS7 standard (used for call setup) and ITU H.323 video protocol combination standard, SIP works independently in the underlying network transmission protocol and media. It specifies how terminal devices of one or more participants can establish, modify and disconnect connections, regardless of voice, video, data or network-based content. SIP is much better than some existing protocols, such as Media Gateway Control Protocol (MGCP) which converts PSTN audio signals into IP packets. Because MGCP is a closed pure voice standard, it is complicated to enhance it through signaling function, which sometimes leads to the destruction or discarding of messages, thus preventing providers from adding new services. Using SIP, programmers can add a small amount of new information to the message without affecting the connection. For example, SIP service providers can create new media including voice, video and chat content. If MGCP, H.323 or SS7 standards are used, the provider must wait for a new version of the protocol that can support this new media. If SIP is used, companies with branches in two continents can realize media transmission, although gateways and devices may not recognize media. Moreover, because the message structure of SIP is similar to that of HTTP, developers can use a common programming language (such as Java) to create applications more conveniently. For operators who have been waiting for several years and want to deploy services such as call waiting and calling number identification using SS7 and AIN, if SIP[4] is used now, it only takes a few months to deploy advanced communication services. This scalability has achieved great success in more and more SIP-based services. Vonage is a service provider for users and small business users. It uses SIP to provide users with more than 20,000 digital local calls, long-distance calls and voice mail lines. Deltathree provides Internet telephony products, services and infrastructure for service providers. It provides a PC-to-phone solution based on SIP, which enables PC users to call any phone in the world. Denwa Communications Company wholesales voice services around the world. It uses SIP to provide calling number identification, PC-to-PC and telephone-to-PC voicemail, as well as Web-based teleconference, unified communication, customer management, self-configuration and personalized services. Some authorities predict that the relationship between Sip and IP will develop into the relationship between SMTP and HTTP and the Internet, but others say that this may mark the end of AIN. So far, the 3G community has chosen SIP as the session control mechanism of the next generation mobile network. Microsoft chose SIP as its real-time communication strategy and deployed it in Microsoft XP, Pocket PC and MSN Messenger. Microsoft also announced that the next version of CE dot net will use the VoIP application interface layer based on SIP, and promised to provide voice and video calls based on SIP to users' PCs. In addition, MCI is using SIP to deploy advanced telephone technology services to IP communication users. Users will be able to inform the calling party whether they are available and the preferred communication mode, such as email, telephone or instant message. With online information, users can also immediately establish chat sessions and hold audio conferences. Using SIP will continuously realize various functions. Compression mechanism SIP compression mechanism mainly reduces the time delay by changing the length of SIP messages. Typical SIP messages range in size from hundreds to thousands of bytes. In order to be suitable for transmission over narrowband wireless channels, IMS extends SIP to support the compression of SIP messages. When the wireless channel is fixed, the number of frames k in the SIP message only depends on the message size. From the delay model, we can see that it not only affects the transmission delay of SIP messages, but also affects the probability of SIP retransmission. For adaptive timer, k also becomes a key factor affecting the initial value of timer. [5] Apply for google to release the world's first open source Html5 sip client. The HTML5 SIP client is an open source, completely written by JavaScript, and integrates social networks (FaceBook, Twitter, Google+), online games, e-commerce and other applications. Without extensions, plug-ins or necessary gateways, video stack technology relies on WebRTC. Like the home page

At present, SIP is a text-based protocol similar to HTTP. SIP can reduce the development time of applications, especially advanced applications. Because IP-based s IP uses IP network, fixed network operators will gradually realize the far-reaching significance of SIP technology to them.

Almost all IP voice-related products in the market follow the H.323 protocol issued by ITU-T organization. Although the development and production of these products are based on the H.323 standard, which mainly describes the data transmission in the local area network, there are few descriptions about the design of IP phones, and the contents of the H.323 protocol selected by various manufacturers in the actual development and implementation process are also different, that is to say, although all major manufacturers follow the H.323 protocol, they follow different protocols. Therefore, IP phones between major manufacturers cannot communicate with each other. Therefore, the IP voice communication system in the enterprise must choose the gateway and other equipment produced by the same manufacturer when designing. This greatly restricts the development of IP voice communication system. At present, everyone is aware of this problem and hopes to have a truly unified standard. Moreover, major manufacturers have made a certain degree of alliance to study the formulation of the real standard of IP voice communication system.