Dynamic Jitter Buffer Operation - 3Com VCX V7122 User Manual

ini File Field Name
*
Web Parameter Name
DTMFTransportType
[DTMF Transport Type]
RFC2833PayloadType
[RFC 2833 Payload Type]
MGCPDTMFDetectionPoint
DTMFInterDigitInterval
DTMFDigitLength

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 VCX V7122 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 70 msec, the gateway always buffers incoming packets by at least 70
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 doesn't 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 70 msec Minimum delay and 7 Optimization Factor should provide a
good compromise between delay and error rate. The jitter buffer "holds" incoming packets for
70 msec before making them available for decoding into voice. The coder polls frames from
the buffer at regular intervals to produce continuous speech. As long as delays in the
network do not change (jitter) by more than 70 msec from one packet to the next, there is
136
Valid Range and Description
0 = Erase digits from voice stream, do not relay to remote.
2 = Digits remain in voice stream.
3 = Erase digits from voice stream, relay to remote according to RFC 2833.
Note: This parameter is automatically updated if one of the following parameters is
configured: IsDTMFUsed, TxDTMFOption or RxDTMFOption.
The RFC 2833 DTMF relay dynamic payload type.
Range: 96 to 99, 106 to 127; Default = 96.
The 100, 102 to 105 range is allocated for proprietary usage.
Cisco is using payload type 101 for RFC 2833.
Note: When RFC 2833 payload type (PT) negotiation is used (TxDTMFOption=4),
this payload type is used for the received DTMF packets. If negotiation isn't used,
this payload type is used for receive and for transmit.
0 = DTMF event is reported on the start of a detected DTMF digit.
1 = DTMF event is reported on the end of a detected DTMF digit (default).
Note: The parameter is used for out-of-band dialing.
Time in msec between generated DTMFs to PSTN side.
Default = 100 (msec).
Time in msec for generating of DTMF tone to PSTN side.
Default = 100 (msec).
3Com VCX V7122 SIP VoIP Gateway User Manual