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Before we dive into OpenSIPS, it is very important to understand some important concepts related to Session Initiation Protocol (SIP). In this chapter, we will cover a brief tutorial regarding the concepts used later in this book. By the end of this chapter, we will have covered the following topics:
在我们深入了解OpenSIPS前，我们需要先一步了解Session Initiation Protocol(SIP)。在这一节里，我们将简要介绍本书中所要使用的概念；至本书结尾，我们将介绍以下主题 ：
了解SIP的基本概念和它的用法 描述了SIP结构 阐述各组件的含义 理解并比较各主要SIP消息 解释INVITE和REGISTER消息的各头字段 学习如何处理SIP认证和安全 简单地解释SDP和RTP(Session Description Protocol and Real-Time Protocol)
• Understanding the basics of SIP and its usage • Describing the SIP architecture • Explaining the meaning of its components • Understanding and comparing main SIP messages • Interpreting the header fields' processing for the INVITE and REGISTER messages • Learning how SIP handles identity and privacy • Covering the Session Description Protocol and Real-Time Protocol briefly
SIP是由IETF制定的标准且由RFC在几个文档中进行描述.RFC3261描述SIP 2.x.SIP是一个应用层协议，它用于建议、修改、和终止多媒体呼叫的会议。这些会话可以是音视频对话，网上教育，聊天，或屏幕共享等。SIP非常类似于HTTP协议的设计，去发起，保持和关闭用户间的相互通信会话。如今，SIP是用于网络电话服务提供商、IP PBX、语音应用中的最流行的一个协议。
SIP was standardized by Internet Engineering Task Force (IETF) and is described in several documents known as Request for Comments (RFC). The RFC 3261
describes SIP version 2. SIP is an application layer protocol used to establish, modify, and terminate sessions or multimedia calls. These sessions can be audio and video sessions, e-learning, chatting, or screen sharing sessions. It is similar to Hypertext Transfer Protocol (HTTP) and designed to start, keep, and close interactive communication sessions between users. Nowadays, SIP is the most popular
protocol used in Internet Telephony Service Providers (ITSPs), IP PBXs, and voice
applications.
SIP协议支持创建和关闭多媒体会话的五个特性： 用户定位：从通信中得到接入点的地址 参数协商：确定要使用的媒体和其它参数 用户有效：确认用户是有效的还是不能建立会话的 呼叫建立：建立呼叫中的主叫，被叫等参数，以及整个呼叫过程中的双方的各种通知（如回铃，忙，未找到） 呼叫管理：便于会话转移或关闭
The SIP protocol supports five features to establish and close multimedia sessions: • User location: Determines the endpoint address used for communication • User parameters negotiation: Determines the media and parameters to be used • User availability: Determines if the user is available or not to establish a session • Call establishment: Establishes parameters for caller and callee and informs about the call progress (such as ringing, busy, or not found) to both the parties • Call management: Facilitates session transfer and closing
SIP被设计的作为一个多媒体架构协议族中的一部分，它还包含了其它协议，如： 资源预留协议(RSVP),实时传输协议(RTP),实时会话协议(RTSP),会话描述协议(SDP)，会话 通知协议(SAP).当然，它不依赖其它协议而工作。 The SIP protocol was designed as a part of a multimedia architecture containing other protocols such as Resource Reservation Protocol (RSVP), Real-Time Protocol (RTP), Real-Time Session Protocol (RTSP), Session Description Protocol (SDP), and Session Announcement Protocol (SAP). However, it does not depend on them to work.
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