Encapsulating Voice Packets for Transport参考.ppt

Describe Cisco VoIP Implementations Encapsulating Voice Packets for Transport Voice Transport in Circuit-Based Networks Analog phones connect to CO switches.模拟电话连接到CO交换机. CO switches convert between analog and digital.CO交换机负责模拟与数字的转换. After call is set up, PSTN provides:当呼叫被建立时,PSTN提供如下服务: End-to-end dedicated circuit for this call (DS-0)为端到端的通讯提供专用的电路. Synchronous transmission with fixed bandwidth and very low, constant delay使用较低固定的带宽与持续不变的延迟进行同步传输 Voice Transport in IP Networks Analog phones connect to voice gateways. Voice gateways convert between analog and digital.语音网关负责模拟与数字信号的转换. After call is set up, IP network provides:当呼叫被建立时,IP网络提供如下服务: Packet-by-packet delivery through the network包交换的数据转发 Shared bandwidth, higher and variable delays共享带宽,较高与变化的延迟. Which Protocols to Use for VoIP? Voice Encapsulation Examples Digitized voice is encapsulated into RTP, UDP, and IP.数字化语音被封装到RTP,UDP和IP数据包中. By default, 20 ms of voice is packetized into a single IP packet.默认情况下,20 ms时长的语音被打包成一个IP数据包. Voice Encapsulation Overhead Voice is sent in small packets at high packet rates.使用较小的数据包可以得到较高的包转发率. IP, UDP, and RTP header overheads are enormous:IP,UDP和RTP的包头负载是比较大的: For G.729, the headers are twice the size of the payload.用G.729的头是其载荷的两倍. For G.711, the headers are one-fourth the size of the payload.G.711的报头是其载荷的四分之一. Bandwidth is 24 kbps for G.729 and 80 kbps for G.711, ignoring Layer 2 overhead.G.729需要24kbps的带宽传输20ms的语音数据,而G.711需要80kbps. RTP Header Compression Compresses the IP, UDP, and RTP headersIP,UDP和RTP的报头压缩 Is configured on a link-by-link basis基于链路的配置 Reduces the size of the headers substantially (from 40 bytes to 2 or 4 bytes):充分的减少报头的大小(40字节的报头被压缩成2或4个字节) 4 bytes if the UDP checksum is preserved4个字节时UDP的校验值被保留 2 bytes if the UDP checksum is not sent2个字节时UDP的校验值不会被发送 Saves a considerable amount of bandwidth 节省了一定的网络带宽 When to Use RTP Header Compression Use cRTP: Only on slow links (less than 2 Mbps)低速链路,一般小于2 Mbps If bandwidth needs to be cons