AudioCodes Mediant 800B User Manual page 449

User's Manual
Parameter
RTCP Mode
sbc-rtcp-mode
[IPProfile_SBCRTCPMode]
Jitter Compensation
sbc-jitter-compensation
[IpProfile_SBCJitterCompen
sation]
ICE Mode
ice-mode
[IPProfile_SBCIceMode]
Version 7.2
in the outgoing SDP offer sent to the SIP entity. Whether RTP
redundancy is implemented depends on the subsequent incoming
SDP answer from the SIP entity. The device does not modify the
incoming SDP offer received from the SIP entity, but if RTP
redundancy is required, it will be supported. Select the option if the
SIP entity requires RTP redundancy.

[2] Disable = The device removes the RTP redundancy payload (if
present) from the SDP offer/answer for calls received from or sent
to the SIP entity. Select the option if the SIP entity does not support
RTP redundancy.
Note:

To enable the device to generate RFC 2198 redundant packets,
use the IPProfile_RTPRedundancyDepth parameter.

To configure the payload type in the SDP offer for RTP
redundancy, use the RFC2198PayloadType.
Defines how the device handles RTCP packets during call sessions
for the SIP entity associated with the IP Profile. This is 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
Note:

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 Interactive Connectivity Establishment (ICE) Lite for the SIP
entity associated with the IP Profile. ICE is a methodology for NAT
traversal, employing the Session Traversal Utilities for NAT (STUN)
and Traversal Using Relays around NAT (TURN) protocols to provide
449
20. Coders and Profiles
Description
Mediant 800B Gateway & E-SBC