AudioControl VERSION 6.2 manual B-Channel Off-Hook Cut-Through

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.

SIP Release Notes 40 Document #: LTRT-26901