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AudioCodes MEDIAPACK VERSION 6.2 User Manual

Page 40

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SIP Release Notes

40

Document #: LTRT-26901

CPE SIP Products

25.

B-Channel Off-Hook Cut-Through:

Product

MP-11x

MP-124

Mediant 600

Mediant

1000

Mediant 800 MSBG

Mediant

1000

MSBG

Mediant 2000

Mediant 3000/TP-6310

Mediant

3000

HA/TP-6310

Mediant 3000/TP-8410

Mediant

3000

HA/TP-8410

Management Protocol

Web

INI

SNMP

EMS

CLI

This feature enables PSTN CAS channels/endpoints to receive incoming IP calls even
if the B-channel is in off-hook state. This feature is enabled using the new parameter,
DigitalCutThrough.
The following scenario describes the mechanism of this feature:

a.

A Tel-to-IP call is established (connected) by the device for a B-channel.

b.

The device receives a SIP BYE (i.e., IP side ends the call) and plays a reorder
tone to the PSTN side for the duration set by the new parameter,
CutThroughTimeForReOrderTone. The device releases the call towards the IP
side (sends SIP 200 OK).

c.

The PSTN side, for whatever reason, doesn't go on-hook. However, the device
regards the state as on-hook.

d.

If a new IP call is received for this B-channel after the reorder tone has ended, the
device “cuts through” the channel and connects the call immediately (even though
the B-channel is in physical off-hook state) without playing a ring tone. If an IP call
is received while the reorder tone is played, then the device rejects the call.

This new parameter has also been added to the Tel Profile table (TelProfile
parameter), allowing the Digital Cut-Through feature to be applied to specific B-
channels using specific CAS tables (by assigning this Tel Profile to the required B-
channels only).

26.

Sending and Receiving RTP Packets without SIP Messages:

Product

MP-11x

MP-124

Mediant 600

Mediant

1000

Mediant 800 MSBG

Mediant

1000

MSBG

Mediant 2000

Mediant 3000/TP-6310

Mediant

3000

HA/TP-6310

Mediant 3000/TP-8410

Mediant

3000

HA/TP-8410

Management Protocol

Web

INI

SNMP

EMS

CLI

This feature enables the device to start sending and/or receiving RTP (media) packets
to and from remote endpoints without the need to establish a SIP session. The remote
IP address is determined according to the Outbound IP Routing table (Prefix
parameter). The port used is the same port as the local RTP port (configured by the
existing BaseUDPPort parameter and the channel on which the call is received).
This feature can be configured per trunk using the new RTPOnlyModeForTrunk_ID
parameter or for all trunks using the new RTPOnlyMode parameter.