Version 7.2 449 Mediant 1000B Gateway & E-SBC
User's Manual 20. Coders and Profiles
Parameter Description
support fax detection (receive or send) and negotiation: one SIP
entity must be assigned an IP Profile where the parameter is set
to [1] or [2], while the peer SIP entity must be assigned an IP
Profile where the parameter is set to [2].
This feature is supported only if at least one of the SIP entities
use the G.711 coder.
This feature utilizes DSP resources. If there are insufficient
resources, the fax transaction fails.
Fax Rerouting Mode
sbc-fax-rerouting-mode
[IpProfile_SBCFaxReroutingM
ode]
Enables the rerouting of incoming SBC calls that are identified as fax
calls to a new IP destination.
[0] Disable (default)
[1] Rerouting without delay
For more information, see Configuring Rerouting of Calls to Fax
Destinations on page 708.
Note: Configure the parameter for the IP leg that is interfacing with
the fax termination.
Media
Broken Connection Mode
disconnect-on-broken-
connection
[IpProfile_DisconnectOnBroke
nConnection]
Defines the device's handling of calls when RTP packets (media) are
not received within a user-defined timeout interval (configured by the
BrokenConnectionEventTimeout parameter). The interval can be
during call setup (configured by the NoRTPDetectionTimeout
parameter) or mid-call when RTP flow suddenly stops (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 on page
695.
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.
Media IP Version Preference
media-ip-version-preference
[IpProfile_MediaIPVersionPref
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