Dynamic Jitter Buffer Operation - AudioCodes MediaPack MP-102 User Manual

ini File Parameter
Name
IsCiscoSCEMode
5.9.2.1

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 MP-1xx 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 in order 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 always a sample in the
buffer for the coder to use. If there is more than 70 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
One of the purposes of the Jitter Buffer mechanism is to compensate for clock drift. If the two
sides of the VoIP call are not synchronized to the same clock source, one RTP source generates
packets at a lower rate, causing under-runs at the remote Jitter Buffer. In normal operation
(optimization factor 0 to 12), the Jitter Buffer mechanism detects and compensates for the clock
drift by occasionally dropping a voice packet or by adding a BFI packet.
Fax and modem devices are sensitive to small packet losses or to added BFI packets. Therefore
to achieve better performance during modem and fax calls, the Optimization Factor should be set
to 13. In this special mode the clock drift correction is performed less frequently - only when the
Jitter Buffer is completely empty or completely full. When such condition occurs, the correction is
performed by dropping several voice packets simultaneously or by adding several BFI packets
simultaneously, so that the Jitter Buffer returns to its normal condition.
MP-1xx SIP User's Manual
Table 5-35: Channel Settings, ini File Parameters
0 = There isn't a Cisco gateway at the remote side (default).
1 = There is a Cisco gateway at the remote side.
When there is a Cisco gateway at the remote side, the local gateway must set the value of
the "annexb" parameter of the fmtp attribute in the SDP to "no". This logic should be used if
'EnableSilenceCompression = 2' (enable without adaptation). In this case, Silence
Suppression should be used on the channel but not declared in the SDP.
Valid Range and Description
120
MP-1xx SIP
Document #: LTRT-65404