With the increasingly fierce global market opening and competition, traditional telecommunications network technology is undergoing profound changes, and the competition in the communication market has also become more intense. The business-exchanged service based on the original circuit exchange will gradually transfer to a mechanism based on packet exchange and data communication, and IP will account for the main position, and VoIP technology has become one of the most hot focus of the communication industry.
At present, the standard protocol of ITU-T and the SIP protocol proposed by ITU-T are included in the international standard protocol.
1, H.323 protocol
The H.323 protocol is currently used in the VoIP network, which is used in the range of scope as shown in Figure 1. This architecture includes H.323 terminals, gateways, gates, and multi-point control units (MCUs). The overall target of H.323 realizes media stream exchange between H.323 endpoints.
Figure 1H.323 range and H.323 terminal interaction
Among them, the H.323 terminal is an endpoint that performs real-time communication with other H.323 endpoints; the gateway is a H.323 endpoint, gateway, both sides of the gateway, both sides of the gateway, between the H.323 network and other types of networks, and the media format. The conversion between the gateway is carried out inside the gateway; it is in the H.323 network, it is an optional entity, which can be controlled (refer to the authorization of access from one or more endpoints, and allows or rejects the end point Any call of calls) Many H.323 terminals, gateways, and multi-point controllers; multi-point controller (MC) is a H.323 endpoint that manages multi-point meetings between multiple terminals and / or gateways. The MC pointed out that the media shared by each entity can also change the configuration of the resource. The position of the MC can be placed in a separate MCU or in combination with entities such as gateways, gates or h.323 terminals.
The H.323 protocol is a huge protocol, including many related protocols, forming a protocol stack, as shown in Figure 2. Media exchange is implemented by RTP running on UDP, as long as RTP is inexplicab. The RTP protocol provides end-to-end delivery services for audio, video and other real-time data, which can transmit the timing and sequential information necessary to recover real-time signals to the receiving endpoint, and the RTCP protocol can provide QoS monitoring means to the transmission and reception of both sides and network operators.
Figure 2H.323 Protocol Stack
The message exchanged between H.323 endpoints is defined by the two protocols of H.225.0 and H.245. The H.225.0 protocol includes two parts, part of the IUT-T q.931 proposed variants for establishing and removing connections between H.323 endpoints, referred to as call signaling or q.931 signaling. Another part is called login, licensing, and status (RSA) signaling, between endpoints and watches, so that you will use this signaling to license or reject access to network resources. H.245 is a control protocol for two or more endpoints that ensure that one entity transmits only a medium that can be received and understood by another entity, which runs on one or more logical channels between endpoints. .
These three signaling protocols - RAS, Q.931, and H.245 can be used to establish calls, maintain calls, and dismantle calls. The transmission of different protocol messages is performed alternately, and FIG. 3 briefly describes a call setup process and control of all relevant protocols during the call. It can be understood that when an end point wants to establish a call with another endpoint, first, the source endpoint uses the RAS signaling to obtain license from a warehouse; then, the source endpoint uses the Q.931 signaling to establish and destination endpoints. Communication; Finally, source endpoints use H.245 to control signaling and destination endpoint negotiation media parameters and establish media transfer.
Figure 3 Call process and protocol control
2, SIP protocol
SIP is a signaling protocol developed by IETF to handle the establishment, changes, disassembly of multimedia sessions, compared to other protocols, and SIP describes the session features of potential session participants. Two basic network entities, clients, and servers are defined in SIP. The client refers to an application that establishes a connection with the server in order to send a request to the server, the server is a application that provides services to the client and returns the response. There are four different types of servers:
(1) User proxy server: It contacts the user when receiving the SIP request and returns responders on behalf of the user.
(2) Proxy Server: Represents other clients to initiate a request, which acts in both the server that acts as a client. Before forwarding the request, it can rewrite the contents of the original request message.
(3) Redirect the server: it receives the SIP request, and map the original address in the request into zero or more new addresses, returns to the client.
(4) Register the server: it receives the client's registration request to complete the registration of the user address.
Communication between the SIP network entity is done by the SIP message, and its syntax constitution is based on the text, which can be divided into two types of message types, that is, request messages and response messages.
Each message is composed of one starting line, zero or more headers and any message. For the request message, the starting behavior requests line, specifies the type of submitted request, the syntax structure is: request-line = method sp request URI SP SIP-VERSION CRLF, which defines 7 methods, names, and descriptions in Table 1 Show. For the response message, the start behavior status line indicates that a request is successful or failed, the syntax structure is status-line = SIP VERSION SP Status Code DP Reason-Phrase CRLF, the state code is defined, and its value is at 100 and Between 699, the first number indicates the level of response, and the description of the different levels is shown in Table 2. The header provides more information on request or response and how to carry additional information. Messages typically describe the type of session to be established, but SIP does not define the structure or content of the message body, which is described by another different protocol, the most common for SDP (session description protocol).
Table 1SIP request method description
Method - Description
INVITE - a request for responding to calling users
ACK - used to initialize a session
BYE - End a connected call
CANCLE - Used to cancel a call that has been issued but unconnected
Register - Information for registering clients to registration server
Opti0N - Information and function for querying the server
INF0 - Used to send information in communication but does not change any communication
Table 2SIP request method description
Status code - description
1xx - notice
2xx - success
3xx - redirection
4XX - request failure
5xx - server error
6xx - global error
The SIP protocol supports three call mode. The user agent client calls to the user agent server, and the user agent client redirects the call, and the proxy server represents the user client to the called initiator. Taking the application proxy server as an example, a simple call is established, as shown in Figure 4.
Figure 4 Processing server establishing a call process
3. Comparison between H.323 protocol and SIP protocol
(1) The encoding mechanism of the two protocols is different, and H.323 is used in binary coding, and the implementation is more complicated, and SIP is a text-based protocol, which is simple.
(2) The SIP session request process and the media negotiation process are carried out together, so the call setup time is short, and the signaling control process for negotiations for the call establishment process in H.323 and media parameters is separate.
(3) Defining a special protocol is defined in H.323 to supplement the business, while SIP can easily support the head domain to make simple extensions, it is convenient to support the head domain to make it easy to use the defined head domain. powerful.
(4) H.323 is concentrated, and SIP is similar to other Internet protocols, which is designed to have distributed multicast functions for distributed call model services.
In summary, H.323 is used in traditional telephone signaling patterns, in line with traditional design ideas in communication, has been widely accepted, and the application is mature. The SIP protocol draws on the Internet's standards and protocols design ideas, simple, flexible, etc. It is attracting more and more equipment manufacturers to pay attention and support, and gradually become the direction of future development, but it is not mature enough. At present, these two protocols are actively seeking improvements, and they will coexist in VoIP services for a long time.
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