AudioCodes Mediant 1000B User Manual page 417

User's Manual
Parameter
sbc-rtcp-mode
[IPProfile_SBCRTCPMode]
Jitter Compensation
sbc-jitter-compensation
[IpProfile_SBCJitterCompensatio
n]
SDP Handle RTCP
sbc-sdp-handle-rtcp
[IpProfile_SBCSDPHandleRTCP
Attribute]
Version 7.0
useful for interworking RTCP between SIP entities. For example,
this may be necessary when incoming RTCP is not compatible
with the destination SIP entity's (this IP Profile) RTCP support. In
such a scenario, the device can generate the RTCP and send it
to the SIP entity.

[0] Transparent = (Default) RTCP is forwarded as is (unless
transcoding is done, in which case, the device generates
RTCP on both legs).

[1] Generate Always = Generates RTCP packets during
active and inactive (e.g., during call hold) RTP periods (i.e.,
media is 'a=recvonly' or 'a=inactive' in the INVITE SDP).

[2] Generate only if RTP Active = Generates RTCP packets
only during active RTP periods. In other words, the device
does not generate RTCP when there is no RTP traffic (such
as when a call is on hold).
Note: The corresponding global parameter is SBCRTCPMode.
Enables the on-demand jitter buffer for SBC calls. The jitter
buffer can be used when other functionality such as voice
transcoding are not done on the call. The jitter buffer is useful
when incoming packets are received at inconsistent intervals
(i.e., packet delay variation). The jitter buffer stores the packets
and sends them out at a constant rate (according to the coder's
settings).

[0] Disable (default)

[1] Enable
Notes:

The jitter buffer parameters, 'Dynamic Jitter Buffer Minimum
Delay' (DJBufMinDelay) and 'Dynamic Jitter Buffer
Optimization Factor' (DJBufOptFactor) can be used to
configure minimum packet delay only when transcoding is
employed.

This functionality may require DSP resources. For more
information, contact your AudioCodes sales representative.
Enables the interworking of the RTCP attribute, 'a=rtcp' (RTCP)
in the SDP, for the SIP entity associated with the IP Profile. The
RTCP attribute is used to indicate the RTCP port for media when
that port is not the next higher port number following the RTP
port specified in the media line ('m=').
The parameter is useful for SIP entities that either require the
attribute or do not support the attribute. For example, Google
Chrome and Web RTC do not accept calls without the RTCP
attribute in the SDP. In Web RTC, Chrome (SDES) generates
the SDP with 'a=rtcp', for example:
m=audio 49170 RTP/AVP 0
a=rtcp:53020 IN IP6
2001:2345:6789:ABCD:EF01:2345:6789:ABCD

[0] Don't Care = (Default) The device forwards the SDP as is
without interfering in the RTCP attribute (regardless if present
or not).

[1] Add = The device adds the 'a=rtcp' attribute to the
outgoing SDP offer sent to the SIP entity if the attribute was
417
19. Coders and Profiles
Description
Mediant 1000B Gateway and E- SBC