1. SUMMARY SIP
SIP (Session Initiation Protocol, Session Initiation Protocol) is proposed by the IETF IP telephony signaling protocol. For initiating SIP session establishment and termination of controlling a plurality of participants participate in a multimedia session, and the session can dynamically adjust and modify attributes, such as the required session bandwidth, transmission media type (voice, video, and data, etc.), medium codecs, support for multicast and unicast.
2. SIP features
(1) Protocol format: SIP reference Hypertext Transfer Protocol (HTTP) design ideas and architecture, used many HTTP message type and the header field, Entity (content type description) stream identification information, and considering authentication, authorization, authentication is used, the authorization methods analogous to those used in Web authentication, authorization method. Which indicates the use of H.323 messages and based on ASN.1 encoding rules binary compression method, therefore, easier to read and debug the SIP.
Extensibility (2) protocols: SIP is designed in full consideration of the adaptability of extended to other protocols, and supports a number of addressable address described, including the user name @ host address, called number @ the PSTN gateway address and Tel: 020-62581234 describes other conventional telephone numbers. Thus, according to the SIP callee address calling, the called party can recognize whether the PSTN, and a gateway connected to the PSTN to the called party and to initiate the establishment of the call.
(3) user location: The most power of SIP is to implement user location function via a Uniform Resource Locator (URL). The SIP URL can even be embedded in Web pages or other hypertext links, users simply use the mouse to point to place a call, so that makes applications and other multimedia applications for audio and video integrated easy. And the SIP registration itself contains function registration server, the location server may also be utilized, such as other DNS, LDAP server provides the positioning and the like to enhance their positioning.
(4) call setup: SIP session request and processes the media negotiation process is carried out together, so call setup time is short, and the call set-up channel in H.323 negotiation process and the parameters for media so that the control process is carried out separately. Such fast call setup mechanism can be drawn by comparing the number of messages: H.323 prior to establishing a media channel, the need for initialization message H.225 and H.245 control channels and the associated acknowledgment message, the control protocol is very cumbersome. Using SIP to establish a media channel very efficient - called to the calling media channel in a round trip can be established, and called media channels may be established within a half round-trip time.
(5) complementary expansion function: H.323 defines a special protocol for effecting supplementary services, such as H.450.1, H.450.2 H.450.3 and the like, as long as the full use of SIP has been defined header field, if necessary, a simple extension of the head domain will be able to easily support the supplementary service or intelligent service.
(6) Multicast: H.323 does not support multi-point transmission (Multicast) protocol, only multi-point control unit (MCU) multipoint configuration, and thus can only support a limited multi while point users. SIP itself is developed by the IETF MMUSIC (Multiparty Multimedia Session Control, Multiparty Multimedia Session Control) working group, so that it supports multi-protocol understandable advantage. Session participants and the type of media you can join an existing conference at any time.
3. SIP describes syntax
SDP (Session Description Protocol) Session Description Protocol (SIP) is a standard used by the IETF. The basic object of the SDP message is transmitted to the syntax defined criteria, such as UDP destination port, audio or video coding standard used, event schedules, name of the session / brief description. It is similar to the protocol information for transmitting protocol H.245 capability exchange mechanism. For example, in call, SDP may be used to identify a codec transmission during the information exchange. SDP is also used for real-time information message protocol (RTSP) is.
SDP protocol is a kind of readable text, = several rows, each row ends with CRLF. How different it and H.323 ASN.1 binary encoding, it is to sacrifice bandwidth to facilitate programming and debugging. However, in actual applications, such specialized protocol is not strong participatory convenience of description more easily accepted by the user side.
4. SIP entity
SIP design ideas based on the principle of reciprocity between hosts by layer (Peer-to-Peer) session, SIP defines optional SIP server, to replace the complex H. the addressing process 323 to reduce call setup time. SIP uses model is very suitable for use client / server (Client / Serve) Web type operational environment. Refers to the client application should establish a connection with the server sends a request to the server. User Agent (User Agent) and proxy (Proxy) contains client. Server for the client sent a request to provide services and loopback application response. There are four kinds of basic server.
(1) user agent server: to contact the user when a SIP request, and returns a response on behalf of the user.
(2) Proxy server: a server side can accept the request, while the transmission request as a client as. Proxy requests can be forwarded without any change to the final destination, but also can filter the content of the original request message when the request passes.
(3) weight to the server: receive a SIP request, the request original addresses are mapped to zero or more new addresses, returned to the client.
(4) the registration server: receiving a client registration request, the user to complete the registration address. It works the essence complete user SIP addresses to IP address mapping.
procedures often require the user terminal includes a user agent client and user agent server. Proxy server, redirect server and registration server can be seen as the public nature of the web server. In SIP also frequently mentioned the concept of "location server", but the location server does not belong to the SIP server. SIP server requests location-based services in a manner not within the scope of the SIP.
5. SUMMARY SIP call
(1) call establishment: SIP endpoints by using UDP or TCP signaling invitation message (INVITE) calling another SIP endpoint. Invite message typically contains sufficient information that the called terminal to establish a media connection between the calling endpoint is requested immediately. This information includes the calling endpoint can support media performance, the called terminal and the calling endpoint wishes to transmit media data transfer address.
(2) codec negotiation: when the called terminal does not accept the calling end codec, negotiation failure message will be sent with the cause and the proposed use of the; calling terminal through the network proxy server initiates a re-invite message, after the proxy server converts the called side satisfies the requirements of the codec.
(3) call: transmitting both the media information on a known port address. It is noted that, SIP does not like the concept of logical channels defined by H.323. When a client receives the proposal of several types of media on several UDP or TCP port, it must immediately be ready to receive media on which any port.
(4) the call is terminated: When any party wishes to terminate the call, it should send an end request (BYE) to the other side, the whole call is confirmed after fully completed.
SIP is still in development and testing phase, unlike H.323, as has been widely accepted, but there are many manufacturers (such as 3Com, Cisco, Nortel, etc.) in active development, future development the trend is SIP or H.323 and MGCP used in conjunction.