AudioCodes Mediant 1000B User Manual page 388

Media gateway & enterprise session border controller (e-sbc)
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Parameter
Echo Canceler
echo-canceller
[IpProfile_EnableEchoCanceller]
Broken Connection Mode
disconnect-on-broken-
connection
[IpProfile_DisconnectOnBrokenC
onnection]
Input Gain
input-gain
[IpProfile_InputGain]
Voice Volume
voice-volume
[IpProfile_VoiceVolume]
Media IP Version Preference
media-ip-version-preference
[IpProfile_MediaIPVersionPrefere
User's Manual
RTPRedundancyDepth.
Enables the device's Echo Cancellation feature (i.e., echo from
voice calls is removed).
[0] Disable
[1] Line (default)
For a detailed description of the Echo Cancellation feature, see
Configuring Echo Cancellation on page 176.
Note: The corresponding global parameter is
EnableEchoCanceller.
Defines the device's handling of calls when RTP packets (media)
are not received within a user-defined timeout (configured by the
BrokenConnectionEventTimeout parameter).
[0] Ignore = The call is maintained despite no media and is
released when signaling ends the call (i.e., SIP BYE).
[1] Disconnect = (Default) The device ends the call.
[2] Reroute = (SBC application only) The device ends the call
and searches the IP-to-IP Routing table for a matching rule
and if found, generates a new INVITE to the corresponding
destination (i.e., alternative routing). You can configure a
routing rule whose matching characteristics is explicitly for
calls with broken RTP connections. This is done using the
Call Trigger parameter, as described in Configuring SBC IP-
to-IP Routing Rules.
Note:
The device can only detect a broken RTP connection if
silence compression is disabled for the RTP session.
If during a call the source IP address (from where the RTP
packets are received by the device) is changed without
notifying the device, the device rejects these RTP packets. To
overcome this, configure the DisconnectOnBrokenConnection
parameter to 0. By this configuration, the device doesn't
detect RTP packets arriving from the original source IP
address and switches (after 300 msec) to the RTP packets
arriving from the new source IP address.
The corresponding global parameter is
DisconnectOnBrokenConnection.
Defines the pulse-code modulation (PCM) input gain control (in
decibels). For the Gateway application: Defines the level of the
received signal for Tel-to-IP calls.
The valid range is -32 to 31 dB. The default is 0 dB.
Note: The corresponding global parameter is InputGain.
Defines the voice gain control (in decibels). For the Gateway
application: Defines the level of the transmitted signal for IP-to-
Tel calls.
The valid range is -32 to 31 dB. The default is 0 dB.
Note: The corresponding global parameter is VoiceVolume.
Defines the preferred RTP media IP addressing version for
outgoing SIP calls (according to RFC 4091 and RFC 4092). The
RFCs concern Alternative Network Address Types (ANAT)
semantics in the SDP to offer groups of network addresses (IPv4
388
Mediant 1000B Gateway and E- SBC
Description
Document #: LTRT-27044

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