VOIP دىكى SIP كېلىشىمى بىلەن H.323 كېلىشىمىنىڭ پەرقى

In IP telephone, H.323 and SIP are both used as application layer control protocol for multimedia communication.

SIP：Is a text-based application layer control protocol, session for creating, modifying and releasing one or more participants, SIP is an Internet-originated voice over IP session control protocol, these sessions can be internet multimedia conferences, ip telephony or multimedia distribution. Provides complete session creation and session modification services for a variety of instant messaging services.

1. H-323：Communication field, support multipoint communication

         SIP：Internet, only supports single point communication

2. H-323：The transmission mode changed from circuit switching to packet switching

         SIP：Taking ip phone as an application on the Internet increases signaling and QoS requirements compared with other applications such as (FTP, E-mail), both of which use RTP (real-time protocol) as the protocol for media transmission.

3. H-323：Binary based on ASN.1 and compression encoding

         SIP：Based on text protocol, similar to http

4. H-323：Since the multipoint control unit (mcu) centrally performs conference control functions and does not support the signaling multicast function, which limits scalability and reduces reliability, the multipoint control unit is likely to become a bottleneck

         SIP：Distributed call model, with distributed multicast function, not only facilitates conference control, but also simplifies user positioning, group invitation, etc., saving bandwidth

5. H-323：(1) RSA signaling channel (2) Call signaling channel (3) H.245 control channel Reliable transmission protocol required

         SIP：The session request process and the media negotiation process are carried out together, independent of the lower-level protocol, generally using unconnectable protocols such as udp, and using the reliability mechanism of the tissue credit layer to ensure the reliable transmission of messages

6. H-323：Based on IP network environment, the traditional realization of telephone signaling mode, easy to connect with the traditional telephone network, in line with the traditional design ideas in the field of communication

         SIP：Based on the internet, draw on the standard protocols of other internets, follow the principles of simplicity, openness, compatibility and extensibility that the internet has always adhered to, and support third-party call control functions

7. Make full use of the header field (contact header field) defined by SIP and simply extend SIP, for example, call transfer, as long as you add the contact header field in the BYE request message and add the third-party address you want to transfer to, this service can be implemented. SIP is a communication standard with internet application as the background, which popularizes video communication and introduces it into thousands of households. The organic combination of 323 system and SIP system ensures that users can construct a relatively cheap and flexible SIP video system based on the realization of multi-party conferences and other diverse It can realize the interworking between H.323 and SIP.

8. Architecture Comparison：

         H.323：The terminal mainly provides data for media communication, with simple functions, while the realization of call control and media transmission control is mainly completed by the gatekeeper. The H.323 system embodies a centralized and hierarchical control mode, which is easy to manage.

         SIP：Using the cs structure, the control of the call is to encapsulate the control information into the header field of the message and transmit it through the message. The SIP system embodies a distributed control mode, and the distributed management is more complicated.

DTMF

H.323 is a very complex protocol system, which is reflected in the comprehensive application of protocol components, these components include H.225, H.245, H.450, H.26X, G.7XX, RTP, etc., SIP protocol It is simple and easy to implement. SIP can contain all the necessary information in just one request. SIP messages are text based, using the ISO 10646 character set in utf-8 encoding. It has a handshake mechanism and supports the UDP transmission protocol.

In terms of conference calls, the H.323 standard stipulates that a multipoint control unit (MCU) centrally performs conference control functions, and all endpoints participating in a conference send control messages to the MCU. The MCU could be the bottleneck. Moreover, H.323 does not support multicast of signaling, and its unicast function limits scalability. SIP is a standard designed for the distributed call model. It has a distributed multicast function, which not only facilitates conference control, but also simplifies user positioning and saves bandwidth.

The third-party control function of SIP can help realize the management of the teleconferencing system, and can provide system administrators with rich management and control functions.