"[Abstract] this paper introduces IP telephony gateway, SIP protocol and H.248 protocol, and puts forward an IP telephony gateway solution supporting H.248 and SIP protocol based on H.323 protocol stack. Key words: H.323 SIP gateway gatekeeper 1 introduction according to the general technical requirements for IP telephony / fax service, considering the compliance with international standardization and interoperability, the construction of IP telephony / fax network in China should take itu-th.323 protocol as the standard. Therefore, we have developed an IP telephony / fax gateway system supporting H.323 protocol stack. However, with the continuous development of network and VoIP (voice over Internet Protocol) technology, gateway systems adapted to different networks must be developed to support the different needs of operators. H. 248 and SIP (session initialization protocol) protocols are relatively popular and fast-growing network protocols recently. This paper will propose an effective solution based on the combination of these two protocols and the current H.323 protocol stack. 2 Introduction to the developed IP gateway the developed gateway sp30itg of carrier level VoIP system is based on standards and supports various international specifications including itu-tt.120 and H.323. In addition, the system adopts G.711, G.723.1 and G.729A speech coding technology to ensure compatibility with major telephone systems. The following describes the software module of the system (as shown in Figure 1). The software system of VoIP module is divided into two modules, Ca (callagent) module and Mg (media gateway) module. The CA part runs on the CA board and mainly completes call control, protocol processing and resource management. The Mg part runs on the Mg board and mainly completes voice compression and decompression, RTP / RTCP (Real-time Transport Protocol / Real-time Transport Control Protocol) processing, etc. The software structure of Ca part is divided into three modules: CC (call control), RM (Resource Management) and H.323. H. 323 is responsible for the signaling processing at the IP network side, equipment management is responsible for the management of the media gateway board, and call control is responsible for the control of IP call service logic. These services include the IP telephone service of the card number user, the IP telephone service of the calling user, the IP fax service of the card number user, the IP fax service of the calling user, and other VoIP based value-added services, such as Internet call waiting Unified messaging, voice mail and other services. H. 323 protocol stack and call control module are the core of the system. The Mg board software system is divided into three parts: microprocessor software part, dual audio detection / sound generation part and voice / fax DSP part. The microprocessor software part completes the management of the whole mg board and communication with Ca; DTMF / tone completes the reception of DTMF (dual tone multi frequency) tone and the generation of prompt voice; Voice / fax DSP part completes the compression and decompression of voice / fax PCM code stream, as well as mute compression, echo suppression, jitter elimination and other functions. 3 Introduction to H.248 protocol and SIP protocol the main function of H.248 protocol is to establish a good service bearer connection model, separate the call and bearer connection, and realize the service interworking between packet network and PSTN network through the management of various media gateways. H. 248 provides a standard interface for user plane and control plane, making it possible to separate network and service. H. The basic idea of 248 is to disassemble the gateway equipment in the IP phone and divide it into two parts. One is called MGC (media gateway controller), which manages the high-level (above three levels) resources, such as signaling and channel resources allocated to the signaling exchange of the high-level management system; The other part is called Mg (media gateway), which manages lower layer resources, such as voice stream (including encoder), collecting dial-up numbers, providing various dial-up tones, etc. The gateway is decomposed into MGC and Mg, and its schematic diagram is shown in Figure 2. At present, China has determined to adopt ITU-T H.323 protocol system in IP phone, but because of its complex composition and difficult implementation, IETF working group proposed session initialization protocol sip. SIP is a client / server protocol. Protocol messages are divided into two categories. The request message is sent from the client to the server, and the response message is sent from the server to the client. Generally, SIP is implemented through the "invite" method. So "invitation" is the core mechanism of SIP protocol. Compared with H.323, SIP supports basically the same call control functions and services. At the same time, because SIP protocol is simple and adopts text mode, it has better function expansibility and network expansibility, and is easy to implement. H. 323 has relatively complete call and resource management functions, strong media negotiation function and strict backward compatibility. Internationally, IP telephony system based on SIP is developing, and some manufacturers have provided commercial SIP system. Therefore, there is an urgent need for multi protocol stack supporting H.323 and SIP in the future. 4 solutions for H.323, SIP and H.248 multi protocol stack support. The following describes the overall implementation scheme of multi protocol stack, and focuses on the implementation of H.323 and sip support on CA with examples. 4.1 the overall scheme of multi protocol stack, due to the flexibility of SIP message, can realize the SIP signaling call of IP telephone / fax without changing the original ASP signaling between CC and switch. Since H.248 protocol specifies the message flow of a call, and specifically involves the details closely related to the switch, such as ringing, returning ring tone, etc., it can not be realized without changing the ASP signaling, so it is more appropriate to put it on the switch side. However, from the perspective of the overall sp30itg gateway system, it reflects the support of the three protocols (as shown in Figure 3). 4.2 support for H.323 and sip on ca. since the case of initiating a call at the H.323 gateway is consistent with the call establishment process between the usual H.323 gateways, the specific implementation of the multi protocol stack is described below by taking the SIP domain initiating a call and the called party is the multi protocol stack gateway (sp30itg) as an example (the flow chart is shown in Figure 4). The call establishment process corresponding to Fig. 4 is roughly as follows: (1) when the multi protocol stack receives the invite message from the SIP domain, it converts the message format（ 2) The legitimacy of the calling user is demonstrated by sending ARQ to the H.323 gatekeeper（ 3) If the calling user is a legal user, the gatekeeper will send an ACF (authentication confirmation) message to the multi protocol stack（ 4) When the multi protocol stack receives the ACF message, it will send the setup message to the CC module（ 5) The CC module sends a call processing message to the multi protocol stack, which contains the local channel message（ 6) After receiving the callproceeding message sent by CC, the multi protocol stack converts it into a sip 100 trying message response and sends it back to the SIP caller（ 7) The CC module sends an alerting message to the multi protocol stack（ 8) After receiving the alerting message sent by CC, the multi protocol stack converts it into a sip 180 ringing message response and sends it back to the SIP caller（ 9) When H.323 is called off hook, CC module will send connect message to multi protocol stack（ 10) After receiving the connect message sent by CC, the multi protocol stack converts it into a 200 OK message response of SIP and sends it back to the SIP caller. 5 two key problems and solutions in call establishment 5.1 message conversion because the message format of H.323 system follows ASN. 1 standard, while SIP adopts a message format similar to HTTP protocol. In order to realize the call establishment between the two, message conversion must be carried out. The basic coding rule of ASN. 1 is a nested structure. Its basic structure consists of three parts: identifier 8bit group, length 8bit group and content 8bit group. In some cases, content end 8bit group is also required. The message format of SIP is related to the type of message. The request message represents the message from the client to the server, and the response message is the message from the server to the client. Taking the invite message of SIP as an example, the method of message conversion between the two is described below. In different call modes, SIP can be mapped in different ways. For example, in the case of a fast call, the invite message of the IP just maps to an H.323 setup message. In slow calls, the invite message can be converted into setup and H.245 messages. Therefore, different message mapping tables must be selected according to different call modes in the multi protocol stack. In the message transformation, the content transformation of the message should also be included. H. Part of the message content conversion between 323 and SIP is shown in Table 1. 5.2 the conversion of media stream format is described by SDP (Session Description Protocol) protocol in SIP, while H.245 controls the media in H.323. Therefore, the media negotiation process is the interaction process between SDP protocol and H.245 protocol. 6 conclusion based on the developed H.323 protocol stack, this paper puts forward a specific solution supported by multi protocol stack, which is completely feasible on the existing gateway system sp30itg.
reference
1 ITU － t rec.h.323 packet － based multimedia common － communications systems.19972 m.handy.sip: session initiation protocol － rfc254319993 ITU － t rec.h225.0 media stream synchronization for visualtelephone systems on non － guaranteed quality of service lans.1996, technical zone
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