264 3Com® VCX V7111 VoIP Gateway User GuideIn ThroughPacket™ mode, the gateway uses a single UDP port for all incoming multiplexedpackets and a different port for outgoing packets. These ports are configured using theparameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort.When ThroughPacket™ is used, Call statistics are not available (since there is no RTCPflow).Dynamic Jitter Buffer OperationVoice frames are transmitted at a fixed rate. If the frames arrive at the other end at the samerate, voice quality is perceived as good. In many cases, however, some frames can arriveslightly faster or slower than the other frames. This is called jitter (delay variation), anddegrades the perceived voice quality. To minimize this problem, the gateway uses a jitterbuffer. The jitter buffer collects voice packets, stores them and sends them to the voiceprocessor in evenly spaced intervals.The V7111 gateway uses a dynamic jitter buffer that can be configured using twoparameters: Minimum delay, DJBufMinDelay (0 msec to 150 msec):Defines the starting jitter capacity of the buffer. For example, at 0 msec, there is nobuffering at the start. At the default level of 10 msec, the gateway always buffersincoming packets by at least 10 msec worth of voice frames. Optimization Factor, DJBufOptFactor (0 to 12, 13):Defines how the jitter buffer tracks to changing network conditions. When set at itsmaximum value of 12, the dynamic buffer aggressively tracks changes in delay (basedon packet loss statistics) to increase the size of the buffer and does not decays backdown. This results in the best packet error performance, but at the cost of extra delay. Atthe minimum value of 0, the buffer tracks delays only to compensate for clock drift andquickly decays back to the minimum level. This optimizes the delay performance but atthe expense of a higher error rate.The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide agood compromise between delay and error rate. The jitter buffer holds incoming packets for10 msec before making them available for decoding into voice. The coder polls frames fromthe buffer at regular intervals in order to produce continuous speech. As long as delays in thenetwork do not change (jitter) by more than 10 msec from one packet to the next, there isalways a sample in the buffer for the coder to use. If there is more than 10 msec of delay atany time during the call, the packet arrives too late. The coder tries to access a frame and isnot able to find one. The coder must produce a voice sample even if a frame is not available.It therefore compensates for the missing packet by adding a Bad-Frame-Interpolation (BFI)packet. This loss is then flagged as the buffer being too small. The dynamic algorithm thencauses the size of the buffer to increase for the next voice session. The size of the buffermay decrease again if the gateway notices that the buffer is not filling up as much asexpected. At no time does the buffer decrease to less than the minimum size configured bythe Minimum delay parameter.Special Optimization Factor Value: 13
