SIP protocol and video communication ----------------------------------------------------------------------------------------------------------------------------------- ----------------------------------- Keywords: SIP Video Communication H.323 Summary: Dark Overview of Articles There is video communication technology, including H.320 and H.323 applications. Then describe the IETF can be used for protocols for video communications: SIP. The history of the SIP protocol is first described in the SIP introduction, then describes the components of the SIP. Decreasing the components illustrate the flow of SIP call creation. The SIP is used for the advantages and disadvantages of video communication by comparison with the H.323 protocol family. Finally, it indicates the prospect of the SIP protocol for video communication. Introduction Communication is the basic needs of human survival, and communication has become an essential content in modern life. At any time, any location and people communicate with people and people are the goal of telecom development. Communication technology has developed to today, and the telephone network is almost covered around the world. Voice communication (telephone) seems to have basically reached the above objectives. However, with the development of technology, people are not satisfied with only voice communication. Large-scale video communications has become the development direction of the next phase of information industry. Although there have been more than 20 years, it is not only a unified standard and has mature products; however, because of various reasons, there is no universal application like the phone. Video communication seems to have always been a gold mine that is not exploited by a sufficient amount. With the development of transmission technology, bandwidth resources are not bottleneck; with a SARS's raging, video communications has become hot. With the emergence of the SIP protocol, video communications has developed technology in technology. Video Communication Protocol Overview The H.320-based video application conventional conference television utilizes the transmission of real-time audio, video and data information using telephone network 2M or 1.544M DIP digital line connection terminal conference television device. By using a multi-point controller, you can have all the operating switching functions of all the main venue in a control board. The initial conference television manufacturers produced their own dedicated compression and communication algorithms, and various conference television manufacturers products cannot interconnect. As ITU-T launched the H.320 protocol, the above problem has been solved with a large extent. H.320 is a standard for frequent transmission of synchronous circuit switches (such as ISDN). Circuit switched networks are suitable for real-time applications, such as long and audio and video signals with determining delay. The establishment of the circuit depends on the external signaling, the concentrated routing control and expensive switching equipment. Using the H.320 protocol, the ideal circuit for commercial conference TV on the telephone online is 384 kbps. Circuits using 384 kbps provide high quality audio and video signals in a reasonable cost. Direct connection with 2M or 1.544m is certainly easy to meet the above bandwidth requirements, but is equal to establishing a private network, the price will make the user unbearable. Because the telephone network relay price continues to decline, the H.320-based H.320-based television conference application is large, although H.320 communication cost is slightly expensive than existing methods, its market will continue to continue in the next few years. Growth - although its growth is slow. Video Application Based on the H.323 protocol H.323 is a multimedia communication series standard for the local area network developed by the International Telecommunications Union. The protocol is specifically developed for local area network technology that does not provide service quality (QoS), such as TCP / IP and IPX running on the Ethernet, fast Ethernet and token Ring. Although the H.323 protocol is particularly developed, as long as the bandwidth delay meets the requirements, it can also be applied in a larger range, such as metro network and wide area network. In May 1997, the International Telecommunications League 15th studies redefined H.323, which became the standard of multimedia communication system that does not guarantee the quality of service quality. H.323 is established on the basis of H.320. Some of the functions are brought about by group switched networks instead of circuit switched networks, and others have brought by the development of compression algorithms and signaling technology. The H.323 protocol has stipulated that the same video and audio compression algorithm with H.320 has complement some new algorithms. H.323 is a huge protocol, four parts: terminal, gateway, gatekeeper and multi-point control unit: L Terminal: On IP-based network is a client endpoint.
It needs to support the following three features: support signaling and control; support real-time communication; support coding, second compression, and decompression. L Gateway: Provides a connection between the package switching network and the SWIRCH STWORK. L Netlear: It is optional in the H.323 system, but if they appear, they have some mandatory functions, gatekeepers complete address translation, accept control, bandwidth control, and domain management. The gatekeepers also support the call control signaling, call authentication, bandwidth management, and call management of 4 optional features. L Multi-Point Control Unit (MCU): The multi-point control unit supports more than 3 end users for sessions. A typical MCU includes a multi-point controller (MC) and several (or no) multi-point processors (MP). The MC provides control functions such as negotiation between terminals. MP completes the processing of media streams in the session, such as hybrid, voice / video exchange. H.323 is a complex and huge protocol, which is currently the mainstream technology of video applications. However, due to its too complex, it is now being challenged based on SIP protocol video applications. SIP Profile SIP Overview SIP (SESSION Initiation Protocal) is called a session initial protocol, which is organized by IETF (Internet Engineering Task Force) in 1999, in an IP network, especially in the Internet, particularly in Internet. In real, a signaling protocol for real-time communication applications. The so-called session is referring to data exchange between users. In applications based on the SIP protocol, each session can be a variety of different types of content, which may be ordinary text data, or digitally processed audio, video data, but also data such as games, etc. Have huge flexibility. As a standard proposed by IETF, the SIP protocol has been largely drawing on other widely existing Internet protocols, such as HTTP (hypertext transport protocol), SMTP (Simple mail transport protocol), etc., as these protocols are also used. Based on text-based encoding, this is also one of the largest features compared to other existing standards in video communications. The proposal and development of the SIP protocol is accompanied by the development of the Internet. He has passed a few phases so far: the first time the SIP concept is, the main application of SIP is for the Internet. Text applications, such as email, text chat, etc .; March 1999, ITEF's Multimedia Session Control (MMUSIC) Working Group proposed RFC2543 suggestions for various manufacturers and agencies; N 1999, SIP Working Group from MMusic The SIP Working Group was established, and the SIP Working Group was established and the draft of SIP in July 2000; N 2002, ITEF's SIP Working Group issued RFC3261 proposal to replace RFC2543. Due to the shortcomings of the network environment and related multimedia technology, only the application of the SIP protocol, only for various text applications, with the development of technology, and pass and IP network in IP Phone Working Group (IPTEL), IP networks (TRIP) Working Group and other brothers working groups, greatly strengthen support for multimedia communication in the SIP protocol. Due to the rapid development of the Internet, SIP has begun accepted by ITU-T SG16, ETSI TIPON (European Standardization), IMTE and other standardized organizations, and established in these organizations related to SIP-related Working group.
Especially as the main members of ITU-T SG16, based on the development of H323 applications for many years, the SIP application guidance is proposed, and the SIP application guidance is proposed, and the corresponding SIP protocol stack has enabled ITU members to achieve. The interoperability between these two protocols. The basic composition of the SIP system is distinguished according to the logical function. The SIP system consists of four elements: user agent, SIP proxy server, redirect server, and SIP registration server. LSIP User Agent: Also known as the SIP terminal, is an end user in the SIP system, which defines them as an application in RFC3261. Based on the roles they play in the session, it can be divided into two types from the User Agent Client (UAC) and User Agent Server (UAS). The former is used to initiate a call request, which is used to respond to a call request. LSIP Proxy Server: It is an intermediate element that is both a client and a server. It has the ability to parse the name. It can make a call request to the next hop server in front of the user. The server then determines the address of the next hop. l Redirect Server: is a server planning the SIP call path. After obtaining the address of the next hop, tell the front user immediately, let the user send a request directly to the next hop address and exit itself. This call is controlled. LSIP Registration Server: Used to complete the login of UAS, in the NE of the SIP system, all UAS must log in in a login server so that the UAC can find them through the server. Below is a schematic diagram of a SIP call setup process: Figure 1 SIP call setup process
1) SIP User Agent Sends a call setup request to the SIP proxy server; 2) SIP proxy server sends a call setup request to the redirect server; 3) Redirect the server Returns a redirect message; 4) SIP proxy server to the redirect server The specified SIP proxy server sends a call setup request; 5) The requested SIP proxy server uses non-SIP protocol, such as domain name query or LDAP, etc. to the locator server query called called called location; 6) Location server Returns the called location (called SIP proxy server) 7) The requested SIP proxy server sends a call setup request to the called SIP proxy server; 8) The called SIP proxy server sends a call to the SIP user agent (called ringing or display); 9) The called user agent agreed or rejected to the called SIP user proxy server; 10) The called user proxy server agreed or rejected to the proxy server requested by the calling proxy server; 11) The proxy server requested by the calling proxy server The bishop proxy server has agreed or rejected; 12) The calling proxy server indicates whether the called SIP user agent indicates whether the called call request is agreed. After the call is established, the two parties communicate with information such as the media and compression algorithms obtained by negotiation. The call demolition process is similar to the establishment process, which is not described here. SIP Advantages and Problems for Video Communication Since the SIP protocol and the H.323 protocol family are based on a packet switched network, the most mature video communication system currently packet switched online is based on the H.323 protocol. So the video communication system using the SIP protocol is inevitably needs to be compared with the H.323 system to obtain advantages and insufficient. Although the SIP protocol and the H.323 protocol are not who replaces who is competitive relationship, it can help us make more appropriate choices under different conditions by comparion. L The protocol function module compares the user agent equivalent to the terminal (or gateway on the packet switched network side), the SIP server equivalent to the H.323. In addition, SIP is similar to the RAS and Q.931 protocols in H.323, while SDP is equivalent to H.245. In the IETF's SIP architecture, the carrier of the media stream uses the RTP protocol, which is the same as H.323. Therefore, the main difference between H.323 and IETF's SIP is how call signaling and control are implemented. L Establishment and demolition of the Basic Call is based on a reliable transport protocol -TCP protocol, so call setup requires two connection stages: TCP connection establishment and call connection establishment. In the third edition of H.323, support TCP and UDP, thereby simplifying the call setup process. SIP's call establishes a process similar to the H.323 third edition, using the INVITE packet. The process of call demolition is opposite to the call, the call and the called are called, and the H.323 protocol uses the Release Complete, the SIP protocol adopts BYE. L Call Control Service SIP and H.323 support call hold, call forwarding, call forward, call waiting, conference calls, and other supplementary services. Keep a call as an example: H.323 defines a near-point call hold and a far-call to maintain two scenarios that hold the business. The gatekeepers are only transparently transmitted. SS-HOLD. The SIP implements the same function, as long as the INCVITE command that changes the SDP described is sent to the party that needs to be called. Changed SDP Description Segments only turn the destination address sent by the media to <0.0.0.0>, while other content is unchanged. Receive the user's UA, keep the call until there is a new invite arrival. LSIP's third party control third-party control means that the third party who does not participate in the session has the ability to establish a call, and this business feature is currently only SIP has. H.323 is also working in trying to add the same business function. Third-party control has many applications, including the Secretary for manager dialing, automatic dialing, participant call forwarding, and call center business. Third-party control is a business characteristic of SIP worthy of good use.
Due to this feature of SIP, ITU-T and IETF use SIP protocols when implementing PINT (In and Internet Interoperability) services. L The ability exchange capacity exchange is to communicate with each other's processing capabilities of the media stream, determine the ability of the two parties, thereby ensuring that the multimedia signal is accepted by both parties. H.323 is capable of exchange with H.245 protocol. All capabilities of the terminal are described in a set of Capability Descriptor structures, each of them is a Simultaneous Capabilities structure and a Capability Decriptor Number. With this configuration, accurate information of each terminal ability is represented in the related tightening structure. SIP uses SDP to perform capacity exchange, the calling party uses an Option demand to identify the called, current, SIP is not as complete and flexible with H.245, because the expression of SDP, such as SIP does not support asymmetry Ability exchange (only or only) and the concurrent ability of audio and video encoding. L Quality Quality Services contains many different indicators, one and multimedia stream related QoS parameters include bandwidth, maximum delay, time delay jitter, and packet loss rate. In addition, there is also a QoS that the call sets the delay affects the feeling of the sensation, which depends largely on the signaling protocol. Call delay also relies on the transmission protocol of the bearer signaling information used, especially when signaling information is lost. So, for the media stream, we first consider the support protocol to QoS support, then examine the call setup delay, because the impact of the error detection and error correction mechanism is delayed. l The QoS support of the media stream is in H.323, and the gatekeeper provides a set of rich control and management functions, including address translation, accept control, bandwidth control, and geographical management; the gatekeeper also provides call control signaling, call snap , Bandwidth management and call management. SIP itself does not support management and control functions, but relies on other protocols. In recent years, the new classified service architecture begins to focus on the eye, H.323 third edition can provide some hierarchical services based on QoS negotiation parameters (bit flow rate, time delay, jitter), and the terminal can apply for a guarantee when call initialization. A similar service is supported by the service, controlled services and non-specified services, and similar services are supported by SIP and H.323. L Call establishment time H.323 The first version is high at the call setup. The second version is greatly improved, and the third version is better. SIP is very similar to the H.323 third edition when the call is settled. However, if the UDP call sets failed, the third version of H.323 is better than SIP, H.323, almost, also established a UDP connection and a TCP Connection, it provides an effective mechanism that closes the TCP connection if the UDP connection is successful; otherwise, TCP is enabled immediately. SIP is sequentially operated UDP and TCP, if the UDP fails, the call setup delay is increased. L loop detection is to prevent loops, H.323 define the PathValue domain to indicate the maximum number of signaling information that can be achieved before discarding. The problem is to define a very critical value. This value should be changed accordingly after the network changes. The SIP uses the VIA header field to check its content. If the new endpoint has appeared in the VIA list, it means there is a loop. The method of SIP is better than H.323. L Interoperability between Interoperability: The complete backward compatibility of H.323 makes all different H.323 versions to achieve seamless integration. In terms of SIP, new versions may make some old features no longer be implemented.
