The multimedia communication system on the H.323, SIP, MGCP, H.248 and softswitch packets is only two types of H.323 and SIP (SESSION Intiation Protocol). These two framework protocols are flat. The H.323 uses centralized management and processing methods to push the complexity of the network device to the network. MGCP (Media Gateway Control Protocol) Media Gateway Control Protocol can be applied to the H.323 system can also be applied to the SIP system, and MGCP only involves the problem of gateway decomposition. The gateway is decomposed into MGC (Media Gateway) and MG (Media Gateway ). The gateway here is a gateway between the packet switched network and the traditional circuit switched network. H.248 is a gateway control protocol. It is a formal standard for telecommunications standardization research teams in IETF (Internet Engineering Task Force "Task Force and ITU-T (International Telecommunication Union-Telecom) International Telecommunications Alliance. Thus the MGCP has greatly advanced. It should be said that H.248 has replaced MGCP and becomes the protocol standard between MGC and MG. Softswate is proposed with a switching matrix in the traditional circuit switched network and is exchanged by hardware, in fact, the exchange of IP phones is soft exchange. Softswitch is just a means of exchange implementation, that is, it can be applied to the H.323 system can also be applied to the SIP system. The media gateway controller device (MGC) decomposed after the gateway is a softswitch device. The above is a personal summary after reading: "Overall Technical Requirements for IP Phone / Fax Service" (below) ---- About H. 323, SIP, MGCP, H. 248 and soft exchange
CWW.NET.CN JANUARY 1 July 2003 13:57 Communication World Network For IP phones standard issues, due to problems with different international organizations and commercial interests of different equipment manufacturers, they have said. This makes it difficult for domestic operators and equipment manufacturing companies to set up their ideas, and it is necessary to make some instructions on the most aspects of the current statement. The current statement is generally involved in the following four questions: First, about H. 323 Recommendation H. 323 It is recommended to be a proposal proposed for the ITU-T 16 (SG16) for multimedia conference systems, not specifically proposed for IP phones, so it involves far more than IP phone. As long as it is an IP phone, especially the phone to the phone from the IP phone of the gateway, H. 323 It is recommended to complete its requirements, so H. 323 It is recommended to be "borrowed" to come as a standard for IP phones. Since the current IP phone has developed very quickly, and the multimedia conference system of IP network has developed relatively slow, so in order to adapt to the application of IP phones, h. 323 is also independent of the IP phone adds some new content (such as the fast establishment of the call). For IP phones, it does not use H. 323 Recommendation, and use a series of recommendations, including H. 225, H. 245, H. 235, H. 450 and H. 341, etc. Only H. 323 It is recommended to "overall technical requirements", so I usually refer to the IP phone of this approach to H. 323IP phone. H. 323 Recommendation is a more complete recommendation that provides a working mode in centralized processing and management, which matches the management method of the telecommunications network, which is why IP phones used in the telecommunications network have almost no exceptions. The land is based on H. 323 IP phone working mode.
Currently H. 323IP phones are most controversial. One is: someone thinks H. 323 mode is only available for small nets, not suitable for big nets. Its reason is H. 323 single domain structure is unable to do big net, and other address parsing commands LRQ uses broadcast mode, further limits its operating range. We have introduced multi-stage multi-domain IP telephone architecture and multi-layer non-broadcast mode address analysis mechanism in detail. Obviously, this problem has been overcome. From now on, theory and practice have shown H. 323 Beneficial to make an IP telephone system that is any scale. The other is: think H. 323 It is recommended to be more complicated, resulting in H. 323 The IP telephone system recommended is too complicated. At this point, some manufacturers' propaganda is misleading. Any communication system including IP phone, after its demand, from the system perspective, its system complexity has been determined, and the remaining is just a function allocation problem. Is the function set in the center, or the function is allocated to the edge, which is based on the specific environment and management system, rather than the forehead. For IP phones from the phone to the phone, the function cannot be assigned to the edge (no smartphone), and it is only possible to use the centralized management and processing method, so H. 323 mode is actually a good working mode. Second, the other regular agreement that SIP often talks is SIP, the SIP protocol proposes another set of IP phones, which is one and H. 323 SAIP protocol. In order to understand the SIP protocol, the SIP protocol is described in detail below. 1. The composition of the system is a SIP system mainly consisting of two parts: user agent and web server. User Agent has a User Agent Client (UAC) and User Proxy Server (UAS), where the user agent client is used to initiate a call, and the user agent server is used to respond to calls. The User Agent Client and User Agent Server constitutes the user-ended application, which is initiated and received by these two applications. There are two types of web servers that are proxy servers (Proxy) and Relocation Servers (REDIRECT).
The proxy server is similar to the MTA (MessageTransfer Agent) of HTTP's Proxy and SMTP. It is a bit like a repeater. It does not respond to the user request, just forwarding the user's repeater, then add the address of the message to the path header of the message. To ensure that the occurrence of the loop will be returned to the response to return and prevent the occurrence of the loop. The relocation server is very similar to DNA. After it receives the user's request, if it is determined that it is not the destination address, respond to the user to the next address to access the server, not forward request messages. 2. The starting point of the SIP network structure SIP protocol is to learn from the success of the Web business, and the IP telephone service network is based on the existing Internet, so the SIP has H. 323 completely different design ideas. It is a decentralized protocol that pushes the complexity of the network device to the network, so that the core network is still a "BESTeffort" transfer channel, which is the reason why the core network server in the SIP system can not reserve the state (STATELESS). The SIP message itself contains all information about a call). Because the core network server needs to handle a large number of calls, the status of each call will greatly improve the system's processing power, lay the foundation for the establishment of large-scale IP telephone traffic network, while the edge network server can be status (stateful ). This STATELESS and STATEFUL combined mode can give full play to the characteristics of SIP (such as user positioning and lookup) and retain the design idea of Internet unable to connect data transfer.
With H. 323 Agreement is based on IP phones, we believe that the SIP protocol requires relatively smart terminals, that is, terminals need to include both user proxy clients and user proxy servers, and implements call requests, call promises, and some users of these two parts. It is precisely because the SIP system has a relatively smart terminal system, so it is likely to realize user personalization. The universal use of the SIP protocol is to be comparable to the number of Internet users and the number of online PCs is comparable to the number of telephone online telephones. The IP network contains the various web servers necessary for the SIP system. A normal connection process is: 1) UAC issues a call request to the web server (Proxy or Redirect); User request (rejection or accept request), the response returns along the original road; 4) After the calling UA receives the response, turn on the called or terminate this call request; for the user terminal, the user terminal is not intelligent, you can also use SIP. As a call signaling, this will greatly weaken the unique advantage of the SIP protocol, such as supporting the user's mobility, the selectivity of the user dialogue, and some applications combined with the web. The gateway device should have the functions of UAC and UAS, which is equivalent to enhancing the smart user terminal to the network, and implements the function of the intelligent terminal by the gateway. But in this case, the user's personalized characteristics supported by the SIP protocol will be greatly weakened because the gateway needs to be multiple users instead of serving a single user. Therefore, we believe that the SIP protocol is more suitable for smart user terminals, based on existing loose Internet, and closely combined with protocols used on existing Internet.
Also consider replacing the gateway device before the user's telephone, which enables the gateway device, implements the function of UAC and UAS, which will increase the cost of purchasing the front machine. 3. SIP Messages SIP protocols have six types of messages, which are INVITE, BYE, OPTION, ACK, REGISTER, and CANCEL messages. INVITE: INVITE message is used to initiate a call request. The INVITE message includes two parts of the message header and the data area. The INVITE message header contains information such as the main, the address information, call topic, and call priority; the data area is information about session media, which can be implemented by session description protocol SDP, SDP and H. The 245 protocol has a similar function, mainly used to describe the media processing power of the terminal. Since the data area is invisible for the SIP protocol, the SIP protocol can also be used and h. The 245 protocol is used in combination. BYE: When a user decides to abort the session, you can use BYE to indicate the end of the session. Options: The ability information used to ask the called end, but Options itself cannot initiate a call. ACK: Confirmed the received message. Register: Used to transmit location information or address information to the SIP server. Cancel: Cancel the current request, but it does not stop the established connection. 4. The SIP features SIP protocols have simple, extended, and close to existing Internet applications.
Simple refers to only three messages INVITE, BYE, and ACK and four heads (TO, FORM, CALL-ID, CSEQ) can implement simple Internet phones. Scalability refers to the characteristics of the network server combined with Stateful and STATELESS. The features closely combined with the existing Internet are mainly to refer to the SIP protocol and the Web and E-mail business. Click-to-Dial and click Fax (Click-to "and click Fax (Click-To" and Click-to-Dial -Fax) The protocol is based on SIP. Third, about MGCP, H. 248 About MGCP has a lot of confusion, which has also caused a considerable degree of confusion. The first thing to clearly point out that MGCP and H. 323 and SIP are not the same level. H. 323 Recommendations and SIP protocols have proposed two sets of IP phone architectures. The two are completely independent, they cannot be compatible with each other, and it is impossible to include another, both of which are only interoperable. MGCP does not involve the architecture of IP phones, only involving gateway decomposition, so it can not only be used in H. The 323IP phone system can also be used for the IP telephone system for SIP.
Second, in terms of gateway decomposition, IETF and ITU-T are well matched. Under the direct management of the two major organizations, a formal standard has been completed, this is H. 248 Recommendations / Macogo. H. The Context concept is introduced in 248 (the earliest introduced by Lucent), which introduces this concept in MDCP, and adds a number of Package definitions to greatly advance the MGCP. It should be said H. 248 It is recommended that MGCP has been replaced, which has become the protocol standard between MGC and MG. The gateway decomposes into a media gateway (MG) and the Media Gateway Controller (MGC), which is the need to develop large-scale IP telephone gateways. From logically, it is very simple, the gateway can be composed of the media gateway and the media gateway controller. It is not that simple from a physically, so far, the decomposition of the gateway does not have a certain way, but is different from different needs. Generally, the functionality of the media gateway controller is: - Handling H with a gatekeeper. 225 RAS message. - Treatment No. 7 signaling (optional). - Handling H. 323 Signaling (optional). The function of the media gateway is: the endpoint interface of the -IP network. - Circuit switched network endpoint interface. - can also process H. 323 signaling (in a certain type of decomposition). - You can also process circuit exchange signaling with RAS functions (in a certain type of decomposition). - Treat media stream.
The protocol of the media gateway controller is H. 248 Suggested that there must be two terms before discussing the A Interface Agreement. , Simulation lines and RTP streams, etc. ● Context-associated communication passenger is a collection of communications in a call or a session, an associated communication object represents a call or a media type in a session. For example, from SCN to IP calls, an associated communication portfolio includes TDM audio communication termination object and RTP audio streamline termination object. The Context is first proposed in the MDCP protocol, which makes the agreement flexibility and scalability, H. 248 It is recommended to extend this concept. Fourth, soft exchange (Softswitch) is currently sailed. It seems that soft exchange is a new architecture and a new concept that produces a new generation of IP telephone equipment. In essence, soft exchange did not propose a new architecture. At present, there are only two architectures of IP phones, one is based on H. 323 IP phone architecture, the other is based on SIP-based IP phone architecture, there is currently no third architecture. It should be said that soft exchange is just a means of implementation. It can be used to h. 323 The IP phone system can also be used within the SIP-based IP telephone system. The exchange of IP phones is special compared to traditional telephone equipment. In the traditional telephone system, the exchange matrix is exchanged by the exchange matrix in the exchange device, which may be time division, or may be space.
