Configuring Rtp/Rtcp Settings; Configuring The Dynamic Jitter Buffer - AudioCodes Mediant 4000 SBC User Manual

and text telephony signals over a voice channel of a packet network with a codec
appropriate for such signals.
For V.152 capability, the device supports T.38 as well as VBD codecs (i.e., G.711 A-law
and G.711 μ-law). The selection of capabilities is performed using the Coder Groups table
(see ''Configuring Coder Groups'' on page 375).
When in VBD mode for V.152 implementation, support is negotiated between the device
and the remote endpoint at the establishment of the call. During this time, initial exchange
of call capabilities is exchanged in the outgoing SDP. These capabilities include whether
VBD is supported and associated RTP payload types ('gpmd' SDP attribute), supported
codecs, and packetization periods for all codec payload types ('ptime' SDP attribute). After
this initial negotiation, no Re-INVITE messages are necessary as both endpoints are
synchronized in terms of the other side's capabilities. If negotiation fails (i.e., no match was
achieved for any of the transport capabilities), fallback to existing logic occurs (according to
the parameter IsFaxUsed).
Below is an example of media descriptions of an SDP indicating support for V.152. In the
example, V.152 implementation is supported (using the dynamic payload type 96 and
G.711 u-law as the VBD codec) as well as the voice codecs G.711 μ-law and G.729.
v=0
o=- 0 0 IN IPV4 <IPAdressA>
s=-
t=0 0
p=+1
c=IN IP4
m=audio <udpPort A> RTP/AVP 18 0
a=ptime:10
a=rtpmap:96 PCMU/8000
a=gpmd: 96 vbd=yes
Instead of using VBD transport mode, the V.152 implementation can use alternative relay
fax transport methods (e.g., fax relay over IP using T.38). The preferred V.152 transport
method is indicated by the SDP 'pmft' attribute. Omission of this attribute in the SDP
content means that VBD mode is the preferred transport mechanism for voice-band data.
14.3

Configuring RTP/RTCP Settings

This section describes configuration relating to Real-Time Transport Protocol (RTP) and
RTP Control Protocol (RTCP).

14.3.1 Configuring the Dynamic Jitter Buffer

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. However, some frames may 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 device uses a jitter buffer. The
jitter buffer collects voice packets, stores them and sends them to the voice processor in
evenly spaced intervals.
The device uses a dynamic jitter buffer that can be configured with the following:

Minimum delay: 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 device
always buffers incoming packets by at least 10 msec worth of voice frames.

Optimization Factor: 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 decay 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
User's Manual
<IPAddressA
184
Mediant 4000 SBC
Document #: LTRT-41729