|
|
|
| SIP Technology Guide |
|
| |
|
|
Overview The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard protocol for establishing, manipulating and tearing down an interactive user session that involves multimedia elements such as audio, video, instant messaging, or other real-time data communications. Even if H.323 was the first protocol that introduces VoIP, annalists estimate that SIP will play a major role in the next years and will replace H.323 in VoIP applications. SIP is a request-response protocol that works in the Application layer of the Open Systems Interconnection (OSI) communications model, and provides capabilities to: Determine the location of the target end point-SIP supports address resolution, name mapping, and call redirection. Determine the media capabilities of the target end point-Via Session Description Protocol (SDP); SIP determines the lowest level of common services between the end points. Conferences are established using only the media capabilities that can be supported by all end points. Determine the availability of the target end point-If a call cannot be completed because the target end point is unavailable, SIP determines whether the called party is already on the phone or did not answer in the allotted number of rings. It then returns a message indicating why the target end point was unavailable. Establish a session between the originating and target end point-If the call can be completed, SIP establishes a session between the end points. SIP also supports mid-call changes, such as the addition of another end point to the conference or the changing of a media characteristic or codec. Handle the transfer and termination of calls-SIP supports the transfer of calls from one end point to another. During a call transfer, SIP simply establishes a session between the transferee and a new end point (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions between all parties. SIP normally runs over UDP or TCP, but it can run over other protocols such as IP, ATM, or X.25. It requires only a datagram service and is independent of the packet layer. It can provide out-of-band call setup services in which the SIP exchanges take place over UDP or TCP, but actual data transmission takes place over the public telephone network. SIP Architecture The SIP architecture identifies two basic components: The SIP User Agent (UA) - the endpoint component, which can be represented by a hardware or software device implementing SIP (e.g., an IP phone), and consists of two main components: User Agent Client (UAC) - that initiates the calls User Agent Server (UAS) - that answers the calls The SIP Network Server - handles signaling associated with multiple calls providing name resolution and user location, and it consists of three main groups: SIP Register Server - receives registration messages from endpoints regarding current user location and maps the SIP addresses with the physical location(s) in the domain where the endpoint is located. These mapping data are stored in a database, which can reside on the same machine or on a remote server SIP Proxy Server - forwards the SIP messages to multiple proxy servers, creating a search tree, in order for the SIP messages to reach their destination. There are two different operating modes for these servers: stateless (the server forgets all the information once the request is sent) and stateful (the server save previous routing information and is able to use it for improving the message transfer) SIP Redirect Server - helps endpoints to find the desired address by redirecting them to try another server 本新闻共4页,当前在第1页 1 2 3 4 Call Setup The diagram below illustrates a basic SIP call setup scenario, using the following messages: INVITE - initiates session. The session description is included in the message body. Re-INVITE is used to change session state ACK - confirms session establishment and can be used only with INVITE BYE - terminates session CANCEL - cancels a pending INVITE OPTIONS - capability inquiry REGISTER - binds a permanent address to current location and it may convey user data If a call is to be routed through a number of different Proxy servers Redirect server is used. When a caller UA sends an INVITE request to the redirect server, the redirect server contacts the location server to determine the path to the called party, and then the redirect server sends that information back to the caller. The caller then acknowledges receipt of the information. The caller then sends a request to the device indicated in the redirection information (which could be the callee or another server that will forward the request). Once the request reaches the callee, it sends back a response and the caller acknowledges the response. RTP is used for the communication between the caller and the callee. Applications SIP has been described as a simple protocol with profound implications. It addresses many of the major issues of the development of Internet telephony - a technology that is predicted to change the way businesses and people talk to each other. The main applications implemented with SIP are: Unified Communications - A SIP session can contain any combination of media (voice, data, video, etc.). These sessions can be modified at any time by adding new parties or by changing the nature of the session. SIP allows browsers to become augmented with multimedia capability. Using SIP, simple, but very powerful, services like click-to-dial become possible. User profiles can be managed through a web interface and voice plug-ins are incorporated into browser technology SIP uses MIME, the de facto standard for describing content on the Internet, to convey information about the protocol used to describe the session and has an URL-style addressing system. It uses the Domain Name System (DNS) to deliver requests to the server that can appropriately handle them Unified Messaging - e-mail, voice-mail, faxes, and phone messages are accessible from the same box. Alternatively, people use many different devices to communicate. Unified messaging helps people that use different communication devices, media, and technologies to communicate at any time and under their own control. Directory Services - Directory services are to a network what white pages are to the telephone system. They store information about things in the real world, such as people, computers, printers, and so on, as objects with descriptive attributes. People can use the service to look up objects by name; or, like the yellow pages, they can be used to look up services. Network managers use directories to manage user accounts and network resources. From a managers viewpoint, a directory service is like an inventory of all the devices on the network. Any device can be located by using a graphic interface or by searching for its name or some properties (e.g., color printer). Once located, a manager can control the device (e.g., disable it or block certain users from accessing it). The directory is a central database where all objects and users are managed IP-PBX functionality - Software based IP_PBX that is compliant with the SIP standard can be utilized in a single office setting or multiple office locations, offering flexibility and options for future expansions Voice-enhanced e-commerce - a website contains click-to dial links that establish a session between the end-user and the website organization. This kind of service could be a part of a value-added web-hosting service offered by a service provider or it could be developed by an enterprises IT department Web Call Centers - a web page may be popped when a particular number is called (with SIP, it is just as easy to direct an user to a web page as it is to a telephone). SIP supports IVR (Interactive Voice Response) features, navigating users through options and providing auto-responses to common requests. In addition, SIPs forking facility is perfect for fulfilling the ACD (Automated Call Distribution) function Instant Messaging (IM) and Presence - because a SIP session can consist of any form of communication, it is possible to promote an IM session to a telephone call or even a whiteboard or video session at the click of a button. It is also easy to invite other people to join your session, creating spontaneous conference calls. Using third party call control, a conference service could even check the presence status of people due to join a conference and when all the parties are available it could establish the session by connecting them all to a conference bridge. Presence goes hand-in-hand with the evolution of voice services. A network that has dynamically updated information about an users preferences and availability can perform more intelligent call routing than todays PSTN or existing find-me/follow-me services. Mobile phones and PDAs - Because SIP client software is lightweight, it can be embedded in mobile phones and PDAs so that these services can cross all platforms. Using SIP as the signaling protocol means that sessions can be established between different devices that then negotiate the appropriate media capability. These devices becomes means of accessing those services associated with a user instead of being closed, proprietary systems Wireless LAN VoIP Telephone Handsets - dedicated portable telephone handsets, supporting Voice over IP on an 802.11 wireless LAN connection. They may use SIP and other proprietary protocols (i.e. SKINNY) and may also support wireless telephony protocols (i.e. GSM) Desktop Call Management - SIP enables a convergence at the desktop. Voice services can be assimilated into other applications to change the way we use our computers. The information management capabilities of the Internet can be used to transform communication systems and improve productivity [1] [2] 下一页
 |
频道声明：本频道的文章除部分特别声明禁止转载的专稿外，可以自由转载.但请务必注明出出处和原始作者 文章版权归本频道与文章作者所有.对于被频道转载文章的个人和网站,我们表示深深的谢意。
| 原始作者:佚名 |
录入时间:2006-12-26 2:27:30 |
| 信息来源:不详 |
投稿信箱:itqoo@126.com |
|
|
 |
|
|
|
| 教程录入：itqoo 责任编辑：itqoo |
|
上一个教程： H.323 Technology Guide
下一个教程： 核心 MPLS IP VPN 体系结构 |
| 【字体：小 大】【发表评论】【加入收藏】【告诉好友】【打印此文】【关闭窗口】 |
|
用户登录 
新用户注册 
网友评论：（只显示最新10条。评论内容只代表网友观点，与本站立场无关！） 