📄 001__45.htm

📁 通讯类的标准。对要开发SS7的朋友有很大帮助的。（通讯协议）
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<DIV class=im3><IMG height=2 src="001__45.files/001_45_45_4.jpg" width=561 
border=0></DIV><!-- text starts here --><SPAN class=ps0><NOBR>能同时存在<SPAN 
class=em0>H.323</SPAN>和基于文本的协议。更可能的情况是每个载波选择一种协议，优胜者得到载</NOBR></SPAN> <SPAN 
class=ps1><NOBR>波提供的所有新服务。</NOBR></SPAN> <SPAN class=ps2><NOBR><SPAN 
class=ft1>SIP<SPAN 
class=em1>服务器能够运行代理和重定向服务。代理服务器代表它们服务的各方来执行呼叫信令。</SPAN></SPAN></NOBR></SPAN> 
<SPAN class=ps3><NOBR>重定向服务器确定被叫方的当前位置，并指示呼叫方直接与被叫方初始化信令。这两个呼叫</NOBR></SPAN> 
<SPAN class=ps4><NOBR>信令概念分别和<SPAN class=em0>H.323</SPAN>中的<SPAN 
class=em0>GK</SPAN>路由和直接路由模型类似。在两种情况下，服务器查找用户的</NOBR></SPAN> <SPAN 
class=ps5><NOBR>当前位置，在重定向呼叫模型的情况下，它们通知呼叫方被叫方的当前位置。</NOBR></SPAN> <SPAN 
class=ps6><NOBR>下面总结的是呼叫建立和维持使用的<SPAN class=em0>SIP</SPAN>的<SPAN 
class=em0>5</SPAN>个主要方面：</NOBR></SPAN> <SPAN class=ps7><NOBR><SPAN class=ft1>1) 
<SPAN class=em1>呼叫初始化：<SPAN 
class=em0>SIP</SPAN>是独立的，可以建立点对点、多点会议和简单呼叫。</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps8><NOBR><SPAN class=ft1>2) <SPAN 
class=em1>用户位置服务：用户能够移动到其他地方，从远程获得电话特性。这种服务和</SPAN>H.323<SPAN 
class=em1>中</SPAN></SPAN></NOBR></SPAN> <SPAN class=ps9><NOBR><SPAN 
class=ft1>RAS<SPAN class=em1>所提供的服务是等同的。</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps10><NOBR><SPAN class=ft1>3) <SPAN 
class=em1>用户容量：确定将要使用的介质和介质参数。</SPAN>SIP<SPAN class=em1>使用</SPAN>SDP<SPAN 
class=em1>协议格式来协商介质参数，</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps11><NOBR>这一点很像<SPAN class=em0>S/MGCP</SPAN>。<SPAN 
class=em0>H.323</SPAN>使用<SPAN class=em0>H.245</SPAN>信令机制。</NOBR></SPAN> <SPAN 
class=ps12><NOBR><SPAN class=ft1>4) <SPAN 
class=em1>用户可用性：确定被叫方是否愿意参与通信。</SPAN>SIP<SPAN 
class=em1>定义了非常明确的应答编码来提供关</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps13><NOBR>于用户当前可用性的信息。</NOBR></SPAN> <SPAN class=ps14><NOBR><SPAN 
class=ft1>5) <SPAN 
class=em1>呼叫管理：包括已经建立的呼叫、电话特性和呼叫简单终止的转换。这对于公用网中</SPAN></SPAN></NOBR></SPAN> 
<SPAN class=ps15><NOBR>的电话服务十分重要。</NOBR></SPAN> <SPAN class=ps16><NOBR><SPAN 
class=ft1>SIP<SPAN class=em1>使用单一资源定位器（<SPAN class=em0>Uniform Resource 
Locators</SPAN>，<SPAN 
class=em0>URL</SPAN>）来标识信源、当前目标、</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps17><NOBR>最终目标和指定重定向（发出）地址。<SPAN 
class=em0>bill.douskalis@dflx.com</SPAN>格式的<SPAN 
class=em0>Email</SPAN>形式的命名方法可</NOBR></SPAN> <SPAN 
class=ps18><NOBR>以是和主机相关的或不相关的，<SPAN class=em0>UDP</SPAN>端口号可以包含在<SPAN 
class=em0>URL</SPAN>中。<SPAN class=em0>SIP</SPAN>公认的<SPAN 
class=em0>UDP</SPAN>端口是<SPAN class=em0>5060</SPAN>。</NOBR></SPAN> <SPAN 
class=ps19><NOBR><SPAN class=ft1>SIP<SPAN 
class=em1>信令功能和协议的正常运转和其他协议无关，包括</SPAN>RSVP<SPAN class=em1>、</SPAN>RT<SPAN 
class=em1>（</SPAN>C<SPAN class=em1>）</SPAN>P<SPAN 
class=em1>、或者用于建立</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps20><NOBR>多媒体会议、服务质量控制和信令的实时流协议（&nbsp;<SPAN class=em0>Realtime Streaming 
Protocol</SPAN>，<SPAN class=em0>RTSP</SPAN>）。但</NOBR></SPAN> <SPAN 
class=ps21><NOBR>是，介质传输层或硬件的故障会影响系统的全部操作。使用<SPAN 
class=em0>SIP</SPAN>检测和报告异常在很大程度上</NOBR></SPAN> <SPAN 
class=ps22><NOBR>和实现无关。在有些情况下还实现了诊断报告机制，如<SPAN class=em0>SNMP<SPAN 
class=em1>，</SPAN></SPAN>它们和<SPAN class=em0>SIP</SPAN>并行工作。</NOBR></SPAN> <SPAN 
class=ps23><NOBR>像它的两个协议竞争者一样，<SPAN class=em0>SIP</SPAN>使用<SPAN 
class=em0>RTP</SPAN>来进行介质传输，使用<SPAN class=em0>RTCP</SPAN>来监视呼叫质量，</NOBR></SPAN> 
<SPAN class=ps24><NOBR>收集重要的统计数据。通过信令平台的设计可以使服务器获得<SPAN 
class=em0>RTCP</SPAN>产生的数据。</NOBR></SPAN> <SPAN class=ps25><NOBR><SPAN 
class=ft1>2. SIP<SPAN class=em4>信令方法</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps26><NOBR>如表<SPAN class=em0>1-13</SPAN>所示，<SPAN 
class=em0>SIP</SPAN>使用<SPAN class=em0>6</SPAN>种信令方法，它们是<SPAN 
class=em0>INVITE</SPAN>、<SPAN class=em0>ACK</SPAN>、<SPAN 
class=em0>OPTIONS</SPAN>、<SPAN class=em0>BYE</SPAN>、</NOBR></SPAN> <SPAN 
class=ps27><NOBR><SPAN class=ft1>CANCEL<SPAN class=em1>和</SPAN>REGISTER<SPAN 
class=em1>。根据从实际实现中获得的知识，&nbsp;<SPAN 
class=em0>IETF</SPAN>工作组已经提出了其他方法，</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps28><NOBR>但使现在还没有被采纳为<SPAN class=em0>RFC</SPAN>地位。</NOBR></SPAN> <SPAN 
class=ps29><NOBR><SPAN class=ft2>表<SPAN class=em5>1-13 
&nbsp;&nbsp;SIP</SPAN>方法</SPAN></NOBR></SPAN> <SPAN class=ps30><NOBR><SPAN 
class=ft3>SIP &nbsp;&nbsp;<SPAN class=em3>方&nbsp;法</SPAN></SPAN></NOBR></SPAN> 
<SPAN class=ps31><NOBR><SPAN class=ft3>INVITE</SPAN></NOBR></SPAN> <SPAN 
class=ps32><NOBR><SPAN class=ft4>这是呼叫处理周期中呼叫方发送的第一条信息。它在<SPAN 
class=em2>SIP</SPAN>报头中包含的信息，标识了呼叫方、</SPAN></NOBR></SPAN> <SPAN 
class=ps33><NOBR><SPAN class=ft4>呼叫<SPAN 
class=em2>ID</SPAN>、被叫方、呼叫顺序编号和其他一些内容（参见<SPAN 
class=em2>SIP</SPAN>报头定义）。基本上，它说明了一个</SPAN></NOBR></SPAN> <SPAN 
class=ps34><NOBR><SPAN 
class=ft4>呼叫正在被发起。可以在呼叫过程中发送它来修改呼叫的操作状态（例如，让一方等待）。</SPAN></NOBR></SPAN> <SPAN 
class=ps35><NOBR><SPAN class=ft3>INVITE<SPAN 
class=em3>消息通常包含呼叫参数的</SPAN>SDP<SPAN class=em3>说明，如介质类型和传输地址。当提供了</SPAN>SDP<SPAN 
class=em3>参数的多个</SPAN></SPAN></NOBR></SPAN> <SPAN class=ps36><NOBR><SPAN 
class=ft4>选择时，选定的那个在应答消息中和成功编码（<SPAN 
class=em2>200</SPAN>）一起返回</SPAN></NOBR></SPAN> <SPAN class=ps37><NOBR><SPAN 
class=ft3>ACK</SPAN></NOBR></SPAN> <SPAN class=ps38><NOBR><SPAN 
class=ft4>呼叫代理只对已被成功接受（编码<SPAN class=em2>200<SPAN 
class=em3>）</SPAN></SPAN>的<SPAN class=em2>INVITE</SPAN>请求用<SPAN 
class=em2>ACK</SPAN>应答。<SPAN class=em2>ACK</SPAN>表示呼叫方已经收</SPAN></NOBR></SPAN> 
<SPAN class=ps39><NOBR><SPAN class=ft4>到了对<SPAN 
class=em2>INVITE</SPAN>请求的确认。<SPAN class=em2>ACK</SPAN>消息体可能包含被叫方的介质类型容量的<SPAN 
class=em2>SDP</SPAN>描述。如果成</SPAN></NOBR></SPAN> <SPAN class=ps40><NOBR><SPAN 
class=ft4>功应答中没有包含<SPAN class=em2>SDP</SPAN>说明，最初的<SPAN 
class=em2>INVITE</SPAN>消息中的会话描述参数将被用于介质连接</SPAN></NOBR></SPAN> <SPAN 
class=ps41><NOBR><SPAN class=ft3>OPTIONS</SPAN></NOBR></SPAN> <SPAN 
class=ps42><NOBR><SPAN 
class=ft4>这条消息用于查询呼叫代理的容量。要在呼叫之前确定远程用户支持的介质类型，这是一个</SPAN></NOBR></SPAN> <SPAN 
class=ps43><NOBR><SPAN class=ft4>很好的工具</SPAN></NOBR></SPAN> <SPAN 
class=ps44><NOBR><SPAN class=ft5>第<SPAN class=em6>1</SPAN>章<SPAN 
class=em6>IP</SPAN>语音和交互应用的协议</SPAN></NOBR></SPAN> <SPAN class=ps45><NOBR><SPAN 
class=ft6>45</SPAN></NOBR></SPAN> <SPAN class=ps46><NOBR><SPAN 
class=ft7>文档</SPAN></NOBR></SPAN> </BODY></HTML>
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