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VCX V7111 VoIP Gateway User Guide
In ThroughPacket™ mode, the gateway uses a single UDP port for all incoming multiplexed
packets and a different port for outgoing packets. These ports are configured using the
parameters L1L1ComplexTxUDPPort and L1L1ComplexRxUDPPort.
When ThroughPacket™ is used, Call statistics are not available (since there is no RTCP
flow).
Dynamic Jitter Buffer Operation
Voice frames are transmitted at a fixed rate. If the frames arrive at the other end at the same
rate, voice quality is perceived as good. In many cases, however, some frames can arrive
slightly faster or slower than the other frames. This is called jitter (delay variation), and
degrades the perceived voice quality. To minimize this problem, the gateway uses a jitter
buffer. The jitter buffer collects voice packets, stores them and sends them to the voice
processor in evenly spaced intervals.
The V7111 gateway uses a dynamic jitter buffer that can be configured using two
parameters:
Minimum delay, DJBufMinDelay (0 msec to 150 msec):
Defines the starting jitter capacity of the buffer. For example, at 0 msec, there is no
buffering at the start. At the default level of 10 msec, the gateway always buffers
incoming 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 its
maximum value of 12, the dynamic buffer aggressively tracks changes in delay (based
on packet loss statistics) to increase the size of the buffer and does not decays back
down. This results in the best packet error performance, but at the cost of extra delay. At
the minimum value of 0, the buffer tracks delays only to compensate for clock drift and
quickly decays back to the minimum level. This optimizes the delay performance but at
the expense of a higher error rate.
The default settings of 10 msec Minimum delay and 10 Optimization Factor should provide a
good compromise between delay and error rate. The jitter buffer holds incoming packets for
10 msec before making them available for decoding into voice. The coder polls frames from
the buffer at regular intervals in order to produce continuous speech. As long as delays in the
network do not change (jitter) by more than 10 msec from one packet to the next, there is
always a sample in the buffer for the coder to use. If there is more than 10 msec of delay at
any time during the call, the packet arrives too late. The coder tries to access a frame and is
not 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 then
causes the size of the buffer to increase for the next voice session. The size of the buffer
may decrease again if the gateway notices that the buffer is not filling up as much as
expected. At no time does the buffer decrease to less than the minimum size configured by
the Minimum delay parameter.
Special Optimization Factor Value: 13
Summary of Contents for VCX V7111
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