[Encyclopedia]-SIP (Session Initiation Protocol)

SIP (Session Initiation Protocol)
SIP is a text-based protocol similar to HTTP. SIP can reduce the development time of applications, especially advanced applications. As the IP protocol-based sip utilizes the IP network, fixed network operators will gradually realize the profound significance of the SIP Technology for them.

SIP
Session Initiation Protocol is a signaling control protocol at the application layer. Creates, modifies, and releases sessions for one or more participants. These sessions can be Internet Multimedia Conferencing [1], IP phone or multimedia distribution. Session participants can communicate with each other through multicast, unicast, or a mixture of the two.
SIP and Resource Reservation Protocol (RSVP) responsible for voice quality interoperability. It also cooperates with several other protocols, including Lightweight Directory Access Protocol (LDAP) for locating and remote identity authentication for Identity Authentication Dial-In User Service (RADIUS) and RTP and other Protocols responsible 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 to manage sessions. With this flexibility, it means that sip can be used in a wide range of applications and services, including interactive games, music and video on demand, as well as voice, video, and web conferencing. SIP messages are text-based and therefore easy to read and debug. New Service programming is simpler and more intuitive for designers. The MIME type description is reused as the e-mail client, so session-related applications can be started automatically. SIP reuse several existing mature Internet services and protocols, such as DNS, RTP, And rsvp. There is no need to introduce new services to support the SIP infrastructure, because many of the infrastructure is in place or ready to use.
The expansion of SIP is easy to define. It can be added by the service provider to a new application without damaging the network. The old SIP-based devices in the network will not impede new services based on the 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 transmission layer. Therefore, the underlying transmission can use an atm ip address. 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 the service or session starts the video and voice, you can still transmit the voice to a device that does not support the video, or use other device functions, such as one-way video stream transmission.
Communication providers and their partners and users are increasingly eager for a new generation of IP-based services. Now we have the Session Initiation Protocol (SIP), which is urgently needed. SIP was born less than a decade ago in the computer science laboratory. It is the first protocol suitable for multi-user sessions in various media content. Now it has become the specification of the Internet Engineering Task Group (IETF.
Today, more and more carriers, CLEC (competing with local carriers), and ITSP (IP phone service providers) 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 (key-press call), and Multimedia conferences. Independent software vendors (ISVs) are developing new development tools to build SIP-based applications and SIP software for carrier networks. Network equipment suppliers (Nev) are developing hardware that supports sip signaling and services. Nowadays, many IP phones, user agents, network proxy servers, VOIP gateways, media servers, and application servers are using SIP.
SIP uses similar authoritative protocols, such as web Hypertext Transfer Protocol (HTTP) formatting and Simple Mail Transfer Protocol (SMTP) email protocols-evolved and evolved into a new and powerful standard. However, although sip uses its own unique user proxy and server, it does not work in an integrated manner. SIP supports integrated multimedia services and works with a large number of existing protocols for identity authentication, location information, and voice quality.
SIP is flexible, scalable, and open. It inspires the power of a new generation of Services launched by the Internet and fixed and mobile IP networks. SIP can complete network messages on multiple PCs and phones and simulate Internet sessions.
Unlike the long-standing International Telecommunication Union (ITU) SS7 standard (for call establishment) and ITU H.323 Video protocol combination standard, SIP works independently of the underlying network transmission protocol and media. It specifies how terminal devices of one or more participants can establish, modify, and interrupt connections, regardless of voice, video, data, or web-based content.
SIP is much better than some existing protocols, such as the Media Gateway Control Protocol (MGCP) that converts the PSTN audio signal into IP packets ). Because MGCP is a closed voice standard, it is complicated to enhance it through the signaling function. Sometimes, the message is damaged or discarded, which prevents the provider from adding new services. With sip, programmers can add a small amount of new information to messages without affecting connections.
For example, a sip service provider can create a new media that contains voice, video, and chat content. If the MGCP, H.323, or SS7 standard is used, the provider must wait for a new version of the protocol that supports this new media. If you use sip, although the gateway and device may not be able to recognize the media, companies with branches on two continents can achieve media transmission.
In addition, because the message Construction Method of SIP is similar to HTTP, developers can easily use common programming languages (such as Java) to create applications. For carriers that want to use SS7 and advanced intelligent network (AIN) to deploy call wait, caller ID Recognition, and other services after several years, if we use sip [2], deployment of advanced communication services takes only a few months.
This scalability has already achieved significant 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 over 20,000 digital city calls, long calls, and voice mail lines. Deltathree provides Internet telephone technology products, services and infrastructure for service providers. It provides a SIP-based PC-to-phone solution that enables PC users to call any phone number in the world. Denwa communications wholesale voice services worldwide. It uses SIP to provide caller identification, voice mail, teleconference, Unified Communication, customer management, self-configuration, and web-based personalized services from PC to PC and telephone to PC.
Some authority predict that the relationship between SIP and IP will develop into a relationship similar to SMTP and HTTP and Internet, but some people say that it may mark the end of ain. So far, the 3G community has selected sip as the next-generation mobile network session control mechanism. Microsoft has selected sip as its real-time communication policy 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 SIP-based VoIP application interface layer and promise to provide SIP-based voice and video calls to users' PCs.
In addition, MCI is using SIP to deploy advanced telephone technical services to IP users. The user will be able to notify the caller whether they are free and the preferred communication method, such as email, phone or instant message. With online information, users can also instantly establish chat sessions and hold audio meetings. Using SIP will continuously implement various functions.

Compression Mechanism
The SIP compression mechanism mainly reduces the Latency by changing the length of the SIP message. The size of a typical SIP Message ranges from several hundred to several thousand bytes. To facilitate transmission over a narrowband wireless channel, IMS expands the SIP message and supports the compression of the SIP message. When a wireless channel is specified, the number of frames per sip message depends on the message size. From the latency model, we can see that not only the delay of SIP message transmission, but also the probability of SIP retransmission are affected. For the adaptive timer, K is also a key factor affecting the initial value of the timer. [3]

Application of SIP
Google released the world's first open-source HTML5 SIP client
The HTML5 SIP client is an open-source client that fully utilizes JavaScript to integrate social networking (Facebook, Twitter, Google +), online games, and e-commerce applications. No extensions, no plug-ins, or necessary gateways. The video stack technology relies on WebRTC. Like Demo Video demo on the home page, you can easily implement real-time video/audio calls between Chrome and IOS/Android mobile devices.
This client is a technology that can be used in a browser to connect to any SIP or IMS network to call and receive audio/video calls and instant information. This Protocol Parser (SIP, SDP...) is highly optimized by using the Ragel lookup table and is suitable for embedded systems with limited hardware (memory and computing power.
New features of the HTML5 SIP client include:
Supports audio/video calls;
Supports instant information;
Presence;
Call persistence/recovery;
Show call transfer;
Supports multiple accounts;
Dual-Tone Multi-frequency signal (DTMF) using siip info

2. Historical Review

Time of occurrence
SIP appeared in the middle of 1990s and originated from research by Henning schulzrinne, associate professor of computer science at Columbia University, and its research team. Professor schulzrinne not only proposed a real-time data transmission protocol (RTP) over the Internet, but also developed a standard proposal for real-time stream transmission protocol (RTSP, it is used to control the stream transmission of audio and video content on the web.
Schulzrinne originally planned to write multi-party multimedia session control (mmusic) standards. In October 1996, he submitted a draft to IETF, which included the important content of SIP. Shulzrinne deleted irrelevant content related to media content in the new standards submitted on April 9, 1999. Subsequently, IETF released the first sip specification, RFC 2543. Although some vendors have expressed concern that the H.323 and MGCP protocols may greatly compromise their investment in the SIP service, IETF continued to do so and released the SIP specification RFC 2001 in 3261.
The release of RFC 3261 marks the establishment of the SIP Foundation. Since then, several additional versions of RFC have been released to enrich content in security, identity authentication, and other fields. For example, RFC 3262 specifies the reliability of the temporary response. RFC 3263 establishes the rules for locating the SIP proxy server. RFC 3264 provides a proposal/response model, and RFC 3265 identifies specific event notifications.
As early as 2001, suppliers began to launch SIP-based services. Today, people are enthusiastic about the agreement. Organizations such as Sun Microsystems's Java Community process are using a common Java programming language to define application programming interfaces (APIS) so that developers can build sip components and applications for service providers and enterprises. Most importantly, more and more competitors are using promising new services to enter the SIP market. SIP is becoming one of the most important protocols since HTTP and SMTP.
Advantages of SIP: Web-like scalable open communication
With sip, service providers can select standard components at will to quickly control new technologies. Regardless of the number of media content and participants, users can find and contact each other. SIP negotiates sessions so that all participants can agree on and modify session functions. It can even add, delete, or transfer users.
However, SIP is not omnipotent. It is neither a Session Description Protocol nor a conference control function. To describe the load conditions and features of message content, SIP uses the Internet Session Description Protocol (SDP) to describe the characteristics of terminal devices. SIP itself does not provide quality of service (QoS). It is interoperable with the resource retention setting Protocol (RSVP) responsible for voice quality. It also cooperates with several other protocols, including Lightweight Directory Access Protocol (LDAP) for locating and remote identity authentication for Identity Authentication Dial-In User Service (RADIUS) and RTP and other Protocols responsible for real-time transmission.

Communication requirements
1. User positioning service
2. Session Creation
3. Session participant Management
4. Limited identification of features

3 session Composition
SIP sessions use up to four main components: SIP User Agent, SIP registration server, SIP proxy server, and SIP redirection server. These systems complete the SIP session by transmitting messages that include the SDP protocol (used to define the content and features of the message. The following describes the various sip components and their functions in this process.
User Agent
The SIP User Agent (UA) is an end user device, such as a mobile phone, multimedia handheld device, PC, and PDA used to create and manage sip sessions. The user agent client sends a message. The user proxy server responds to messages.
Registration Server
The SIP registration server is a database that contains all user proxies in the domain. In SIP Communication, these servers retrieve the IP address and other information of the other party and send the information to the SIP proxy server.
Proxy Server
The SIP Proxy Server accepts the sip ua Session Request and queries the SIP registration server to obtain the UA address information of the recipient. It then forwards the session invitation information directly to the receiver UA (if it is in the same domain) or the proxy server (if the UA is in another domain ).
Redirect Server
The SIP redirection server allows the SIP proxy server to direct the SIP session invitation information to an external domain. The SIP redirection server can be on the same hardware as the SIP registration server and the SIP proxy server.
The following scenarios describe how to coordinate sip components to establish a sip session between UA in the same domain and different domains:
Establish a sip session in the same domain

Future of SIP
SIP can connect to any IP Network (wired LAN and WAN, public Internet backbone network, mobile 2.5g, 3G and Wi-Fi) and any IP device (phone, PC, PDA, mobile handheld device) users, and thus a large number of lucrative new business opportunities, improved the communication between enterprises and users. SIP-based applications (such as VoIP, Multimedia conferences, push-to-talk (key-call), location services, online information, and Im) are used independently, it also provides many new business opportunities for service providers, ISVs, network equipment providers, and developers. However, the fundamental value of SIP is that it can combine these functions to form a variety of large-scale seamless communication services.
With sip, service providers and their partners can customize and provide SIP-based combined services so that users can use conference, web control, online information, IM and other services in a single communication session. In fact, a service provider can create a flexible combination of applications that meet the needs of multiple end users, rather than installing and supporting a single distributed application that relies on the limited functionality or types of terminal devices.
By combining IP-based communication services under a single and open standard sip application architecture, service providers can greatly reduce the cost of designing and deploying new IP-based hosting services for users. It is the powerful motive force for SIP scalability to promote the development of the industry and the market, and the hope of all of us.

4 Protocol Comparison
Comparison between H.323 protocol and SIP protocol

Introduction
H.323 and SIP are the protocols launched by the communication and Internet camps. H.323 attempts to regard IP phones as well-known traditional phones, but the transmission mode has changed from circuit switching to group switching. The SIP Protocol focuses on using IP phones as an application on the Internet, which requires both signaling and QoS compared with other applications (such as FTP and E-mail, they support basically the same business, and also use RTP as the media transmission protocol. But H.323 is a relatively complex protocol.
H.323 uses a binary method based on ASN.1 and compression encoding rules to represent its messages. ASN.1 special code generators are usually required for lexical and syntax analysis. The text-based protocol of SIP is similar to HTTP. Text-based encoding means that the meaning of the header domain is clear at a glance, such as from, to, subject and other domain names. This distributed standard style that requires almost no complex documentation, its superiority has been proven in the past (SMTP, a popular mail protocol, is an example ). The message body of the SIP is described using SDP. The format of each item in SDP is =, which is also relatively simple.
In terms of support for conference calls, as the multi-point control unit (MCU) Centrally implements the conference control function, all the participating Conference terminals send control messages to the MCU, And the MCU may become the neck, especially for large-scale conferences with additional features, and H.323 does not support the multicast function of signaling, its single function limits scalability and reduces reliability. The Design of SIP is a distributed call model with the distributed multicast function. The multicast function not only facilitates conference control, but also simplifies user positioning and group invitation, and can save bandwidth. However, the centralized control of H.323 is easy to charge, and bandwidth management is relatively simple and effective.
Special protocols are defined in H.264 to supplement services, such as h.2.1, h.2.2, and H.264. SIP does not specifically define a protocol for this purpose, but it easily supports supplementary or intelligent services. You only need to make full use of the header domain defined by the SIP (such as the Contact Header domain) and make simple extensions to the SIP (such as adding several domains) to implement these services. For example, for call transfer, you only need to add the Contact Header domain to the bye request message and add the third-party address to be transferred. For some smart services that are difficult to implement by extending the header domain, you can add business agents in the architecture to provide some supplementary services or interfaces with intelligent network devices.
In H.323, the call establishment process involves the third signaling to: Ras signaling channel, call signaling channel, and H.245 control channel. Through the coordination of these three channels, the H.323 call can be carried out, and the call establishment takes a long time. In sip, Session Request and media negotiation are performed together. Although h.323v2 has improved the call establishment process, it still cannot be compared to that of SIP, which only requires 1.5 loop latency to establish a call. H.323 call signaling channels and H.245 control channels require reliable transmission protocols. The SIP protocol is independent of the Low-layer protocol. Generally, the protocol such as UDP cannot be connected. The reliability mechanism of the credit layer is used to ensure reliable message transmission.
In short, H.323 follows the traditional telephone signaling mode, which is relatively mature and has already seen many H.323 products. In line with the traditional design concept in the communication field, H.323 implements centralized and hierarchical control and uses the H.323 Protocol to facilitate connection with traditional telephone networks. The SIP protocol draws on the design ideas of other Internet standards and protocols and follows the principles of simplicity, openness, compatibility, and scalability that the Internet has always adhered.
The following is a simple analysis of their application goals, standard structure, system composition, and the difficulty of system implementation.
Standard Application goals

The H.323 standard was established by the ITU-T organization on the basis of H.320/H.324 in 1996. Its application goal is to: in the network environment of the base IP address, provides reliable real-time applications for audio and video and data. After years of technological development and standard improvement, H.323 has become a mature standard family accepted by many ITU members and customers.
The SIP standard was proposed by itef in 1999. Its application goal is to achieve real-time communication of data and audio and video in an Internet-based environment, especially to popularize the application of video communication to thousands of households. Because the SIP protocol is relatively simple and free compared with H.323, the vendor can construct a system that meets the needs of applications at a relatively low cost. For example, you can simply use Microsoft's SIP-based MSN and RTC to construct a simple, Internet-based video communication environment. In this way, network operators can use existing network resources to expand their video and audio communication services at minimal cost.
Standard Architecture

H.323 is a single standard, not a complete standard family of real-time multimedia applications in an IP environment, there are comprehensive and strict regulations on the establishment, management, and transmission of call media formats. A multimedia system that complies with H.323 Standards can ensure stable and complete multimedia communication applications.
In a strict sense, the SIP standard is a signaling standard that implements real-time multimedia applications. Because it adopts a text-based encoding method, it is used in applications, especially in point-to-point application environments, with great flexibility, scalability, and cross-platform compatibility, operators can easily use the existing network environment to implement large-scale promotion and application.
However, the SIP protocol itself does not support multi-point conference functions and management and control functions. Instead, it depends on the implementation of other protocols, affecting the system completeness, especially for the requirements for multi-point communication, it is difficult to implement a simple SIP System. In response to these shortcomings, the ITU-T sg16 team headed by radvison proposed the use specification of SIP and achieved interconnection between SIP and H.323, and successfully solved the application problem of SIP in Multi-Point environment.
System Composition

First, it is analogous to the functional aspects of the main components of the system. The UA of SIP is equivalent to an H.323 terminal, which can initiate and receive calls and complete the encoding and decoding applications of the transmitted media; the functions of the SIP proxy server, redirection server, and registration server are equivalent to the H.323 keeper, which implements terminal registration, call Address Resolution, and routing.
Secondly, although the implementation of call signaling and control is different, a SIP-based call process is similar to that of H.323 q931. The Session Description Protocol (SDP) used by SIP) it is similar to the call control protocol h.245.
Implementation difficulty

The H.323 standard signaling information adopts the binary code that complies with ASN.1 per and must be strictly defined throughout the connection implementation process. The degree of freedom of the system is small. For example, to implement large-scale application, you need to plan all aspects of the network.
The SIP standard signaling information is text-based and adopts UTF-8 encoding that complies with iso000046. The structure of the entire system is relatively flexible, and the implementation of terminals and servers is relatively easy and cost-effective, from the perspective of network operators, to construct a large-scale video communication network, the cost of using the SIP System is much cheaper and more feasible.

Summary
By comparing the protocols of SIP and H.323, we can easily see that the relationship between H.323 and SIP is not opposite, but complementary in different application environments. As a communication standard based on Internet applications, SIP is an effective and feasible means to popularize video communication and introduce it to thousands of households. The combination of the H.323 and SIP systems ensures that users can achieve diversified functions such as multi-party conferences on the basis of constructing a relatively inexpensive and flexible sip video system, it also reliably implements intercommunication between the SIP System and the H.323 system to meet users' requirements for real-time multimedia communication in the future to the maximum extent.