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📁 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=im0><IMG height=64 src="002_43.files/right.jpg" width=563 
border=0></DIV><!-- text starts here --><SPAN class=ps0><NOBR><SPAN 
class=ft0>AAL2<SPAN class=em1>中在信元丢失情况下的错误恢复十分简单，可以通过</SPAN>CPS<SPAN 
class=em1>子层端点的重新同步完成。</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps1><NOBR>如果中间信元丢失，<SPAN 
class=em0>SN</SPAN>不会双态转换，我们很容易可以看到丢失了一个奇编号的包。部分接</NOBR></SPAN> <SPAN 
class=ps2><NOBR>收到的包被丢弃，在下一个可靠建立的包的有效载荷边界出，该过程重新开始。这种传输机</NOBR></SPAN> <SPAN 
class=ps3><NOBR>制对语音和音频带应用产生了极好的延迟特性。</NOBR></SPAN> <SPAN class=ps4><NOBR>用<SPAN 
class=em0>AAL2</SPAN>处理传真和音频带传输有什么困难呢？如果一个用户的配置说明了使用压缩，当</NOBR></SPAN> <SPAN 
class=ps5><NOBR>他拨叫远程终端时，起始的连接会通过配置中规定的编码解码方案，如<SPAN 
class=em0>G.726</SPAN>来完成。连接会</NOBR></SPAN> <SPAN 
class=ps6><NOBR>保留在这种模式中，直到拨叫远程的传真机号码，并且远程端点将</NOBR></SPAN> <SPAN 
class=ps7><NOBR><SPAN class=ft0>CED<SPAN 
class=em1>音调放到线上为止。</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps8><NOBR>这时，接入点和网络之间的协作功能需要选择一定的比特率，以可靠传输端到端的音频带信</NOBR></SPAN> <SPAN 
class=ps9><NOBR>号。这意味着，当检测到<SPAN class=em0>2100Hz</SPAN>音调时，携带传真呼叫数据的<SPAN 
class=em0>VC</SPAN>中的信道会扩展到<SPAN class=em0>64Kbps</SPAN>。</NOBR></SPAN> <SPAN 
class=ps10><NOBR>扩展会一直有效，直到传真被传送，或者调制解调器信号调制已被关闭。扩展必须快速可靠，</NOBR></SPAN> <SPAN 
class=ps11><NOBR>以便不违反信号握手的定时考虑。</NOBR></SPAN> <SPAN class=ps12><NOBR><SPAN 
class=ft0>AAL2<SPAN class=em1>定义了三种类型的包。</SPAN>1<SPAN 
class=em1>类包，也叫做不受保护的包，作为缺省使用，携带传真传</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps13><NOBR>输的压缩图像。<SPAN 
class=em0>2</SPAN>类包，也叫做部分保护的包，在有效载荷开始处包含内部报头和<SPAN 
class=em0>CRC</SPAN>保护域。</NOBR></SPAN> <SPAN class=ps14><NOBR>报头域是<SPAN 
class=em0>19</SPAN>比特，<SPAN class=em0>CRC</SPAN>占据了<SPAN 
class=em0>3</SPAN>字节报头的剩余<SPAN 
class=em0>5</SPAN>比特。有效载荷的剩余部分是不受保护的。</NOBR></SPAN> <SPAN class=ps15><NOBR><SPAN 
class=ft0>3<SPAN class=em1>类包，也叫做完全受保护的包，用来传输拨叫的数字（<SPAN 
class=em0>DTMF</SPAN>信号）</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps16><NOBR>、<SPAN class=em0>CAS</SPAN>位、传真解调</NOBR></SPAN> <SPAN 
class=ps17><NOBR>控制数据、以及警铃。<SPAN class=em0>3</SPAN>类包可以用来扩充检测远程传真终端的<SPAN 
class=em0>CED</SPAN>信道比特率。</NOBR></SPAN> <SPAN 
class=ps18><NOBR>在网络接入部分，次要的、但仍是强制的需要是，在<SPAN 
class=em0>MG</SPAN>和介质网关控制器间实现了中</NOBR></SPAN> <SPAN 
class=ps19><NOBR>继时所使用的呼叫模型。对于<SPAN 
class=em0>SOHO</SPAN>和语音商业应用，接入中继在大多数情况下是必须的。</NOBR></SPAN> <SPAN 
class=ps20><NOBR>动态或静态的<SPAN 
class=em0>CID</SPAN>分配等因素会影响拨号后延迟以及呼叫用户感受到的总体质量。在接入处</NOBR></SPAN> <SPAN 
class=ps21><NOBR>静态分配<SPAN 
class=em0>CID</SPAN>的优点是连接期间开销少，因为对一个特定端点，虚拟干线的虚拟时隙不需要</NOBR></SPAN> <SPAN 
class=ps22><NOBR>协商或者为每个呼叫分配。在每个呼叫基础上的动态分配提供了灵活性，减少了延迟，适合</NOBR></SPAN> <SPAN 
class=ps23><NOBR>于某些应用。</NOBR></SPAN> <SPAN class=ps24><NOBR><SPAN 
class=ft2>在主干网中合成介质流</SPAN></NOBR></SPAN> <SPAN 
class=ps25><NOBR>当所有接入流合成到云团之内是，需要再次提出古老的添加和引下多路复用（&nbsp;<SPAN class=em0>Add 
and</SPAN></NOBR></SPAN> <SPAN class=ps26><NOBR><SPAN class=ft0>Drop 
Mutiplexing<SPAN class=em1>，</SPAN>ADM<SPAN class=em1>）</SPAN><SPAN 
class=em1>问题。如果主干网是基于</SPAN>ATM<SPAN class=em1>，</SPAN>AAL2<SPAN 
class=em1>的使用可以在某种程度上简</SPAN></SPAN></NOBR></SPAN> <SPAN 
class=ps27><NOBR>化了这个问题。惟一的复杂性在于压缩连接上音频带流的折叠效应。其他主干网技术也使用</NOBR></SPAN> <SPAN 
class=ps28><NOBR><SPAN class=ft0>ATM<SPAN 
class=em1>方案，它们有不同程度的效率、灵活性和延迟。这些技术和网络间介质流路由的信令问</SPAN></SPAN></NOBR></SPAN> 
<SPAN class=ps29><NOBR>题已经超出了本书的范围。</NOBR></SPAN> <SPAN class=ps30><NOBR><SPAN 
class=ft3>第<SPAN class=em2>2</SPAN>章&nbsp;包网络的介质传输</SPAN></NOBR></SPAN> <SPAN 
class=ps31><NOBR><SPAN class=ft4>99</SPAN></NOBR></SPAN> <SPAN 
class=ps32><NOBR><SPAN class=ft5>文档</SPAN></NOBR></SPAN> </BODY></HTML>

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