IMG 1010 - IP Bearer Profile

 

 

 

IP Bearer Profile Pane

Description:

Use this pane to define a set of VOIP parameter values such as Silence Suppression, Echo Cancellation and RTP Redundancy.  You assign the profile to a SIP or H.323 Channel Group to define inbound or outbound behavior. To commit the profile, go back to the IP Bearer Traffic Profiles pane and click the Save Profiles button, or simply select another object in the Configuration Tree. After you create the profile, use the Supported Vocoders pane to assign a list of supported codecs to the profile.

The IMG has an adaptive Jitter Buffer which adjusts its size dynamically. The minimum is 75ms and the maximum is 150ms. The Jitter Buffer is not configurable.

Accessing this Pane

Dialogic IMG EMS -> Profiles -> IP Bearer Profiles -> IP Bearer Profile

Maximum Objects:

16 per EMS

Related Topics:

 Configuring IP Bearer Profiles

Media Inactivity Detection Support

Next Configuration:

IMG 1010 - Vocoder Entry

Channel Group

When configuring an IP Bearer Profile, the following happens:

A record is added for each profile that is created.  When the desired number profiles has been added, the user should then press the “Save Profiles” button in the IP Bearer Profiles object.  This causes DataManager to build and send a resource table with all the records to each of the Logical IMGs.

IP Bearer Clientview pane - 10.5.3 SP19

Field Descriptions:

IP Bearer Profile Id:

This field is populated with the next available ID. The value can be changed by selecting a different ID from the drop down menu. This value will subsequently be used in the Channel Group object to specify the IP Bearer Profile instance.

IP Bearer Profile Name:

This value provides a name to the profile being defined. This value is subsequently used in the Channel Group object to specify the IP Bearer Profile instance.

Silence Suppression:

This field does not apply for the AMR or EVRC codecs as they use their own internal Silence Suppression scheme.

Enable

Disable

See Silence Suppression for more information. 

Echo Cancellation:

In compliance with ITU G.168-2000, this feature eliminates echo introduced by impedance mismatched hybrids on TDM channels. The maximum tail length is 128 milliseconds for VoIP Module SLM-IPM-0001 and 64 milliseconds for the VoIP Module SLM-IPM-0001. Echo Cancellation may be implemented for tandem calls on trunks with echo or to clean an incoming signal before connecting to a media resource, such as a Voice Response Unit or Answering Machine Detection.

In software version 10.5.3 SP3 a Non-Linear Processor control support feature was added which allows the enabling and disabling of the Non-Linear sub-component when Echo cancellation is enabled. See selections below.

Enabled (NLP Enabled) (Default) - Echo Cancellation is enabled along with the Non-Linear Processor sub-component. The Nonlinear Processor (NLP) is used to remove the residual echo signal, that is, the components that could not be removed by the linear filter alone. The NLP is not activated during periods of double talk.

Enabled (NLP Disabled) - Echo Cancellation is enabled but the Non Linear Processor sub-component is Disabled. Disabling the NLP while echo cancellation is enabled, is in cases where heavy double-talk scenarios are expected. Double-talk results in clipping (removal) too much voice from the voice path, causing the voice recognition software to fail to detect voice patterns. Disabling the NLP solves this problem.

Disable - Echo Cancellation is disabled.

RTP Redundancy:

This feature provides RTP packet redundancy to guard against network packet loss (RFC 2198) for RTP traffic in voice or fax/modem bypass calls. This is not supported for iLBC, AMR, and EVRC. EVRC and AMR have internal schemes that are not configurable. Note that Open Phone does not support RTP Redundancy. Selections shown below.

No Redundancy

Redundancy Level 1

RTP Payload Type for Redundancy:

This numeric value (96-101, 104, 106-127) defines the packet type used for RTP Redundancy.

For an incoming call, if the sending gateway is using a codec that is configured on the IMG but with a different dynamic payload type, the IMG will accommodate that by sending the dynamic payload type sent by the sending gateway in the Invite message.  For example, if the IMG has iLBC profile configured with a value of 100, an incoming call to the IMG has iLBC set to 98, the IMG will send the 98 back in the 200 OK to accept iLBC. 

Single Number Fax (SIP Only):

The Single Number Fax field along with the Single Number Fax Server field is used when configuring the Single Number for both Voice and Fax Routing feature.

Disabled (Default) - Feature is disabled

Enabled - Feature is enabled. When enabled, the Single Number Fax Server field below appears.

Single Number Fax Server:

The Single Number Fax Server field is only visible when the Single Number Fax field above is set to Enabled. When enabled, the Single Number Fax Server field will have a drop down menu displaying all the Gateways configured. This field is utilized when configuring the Single Number for both Voice and Fax Routing feature. See link above.

Fax Mode:

This field defines the mode of operation for Fax Calls.

Enable Relay (T.38) - The fax is delivered using T.38 packets.

Enable Bypass - The codec configured in the Fax Bypass Codec field below is used to send the fax. This functionality is not supported for the AMR and EVRC codecs.

Relay Fallback to Bypass -  The Fax Fallback feature is a backup mechanism to transmit a fax using Fax Bypass mode when T.38 cannot be negotiated successfully. This feature allows you to configure T.38 Fax Relay as the preferred type, and also allow Bypass Fax when T.38 is not supported by the remote end. The added negotiation will therefore reduce the call setup failure rate by increasing the content of the media offer. In the event neither a T.38 fax nor a Bypass fax can be established in a fax fallback scenario, the IMG allows the voice call to proceed as if no negotiation had happened.

Fax Bypass Codec:

The codec to use when the Fax Mode is set to Enable Bypass. This field does not apply for the AMR or EVRC codecs.

G711 ulaw: 64 kbps North American

G711 alaw: 64 kbps ITU

Fax Packet Redundancy:

This feature provides Fax packet redundancy to guard against network packet loss.  This is only applicable to Relay Fax Mode. This field does not apply for the AMR or EVRC codecs.

No Redundancy - Original Payload is NOT duplicated.

Redundancy Level 1 - Original payload is duplicated one time.

Redundancy Level 2 - Original payload is duplicated two times.

Redundancy Level 3 - Original payload is duplicated three times

Digit Relay:

This setting specifies the method to use to propagate DTMF digits. This field does not apply for the AMR or EVRC codecs.

DTMF In-band - Digits are sent in the same packets as voice.

DTMF Packetized - Digits are sent in separate packet type (RTP Events as defined by RFC 2833) using the payload type specified by the Digit Relay Packet Type. If this field is set to DTMF packetized then the telephone event message will be sent in the CDR otherwise the telephone event message will not be sent. See Below.

DTMF Packetized - Digits are sent in separately.
 

Example of "telephone event" line in SDP

v=0

o=- 1643042763 1643042846 IN IP4 10.129.39.111

s=eyeBeam

c=IN IP4 10.129.39.111

t=0 0

m=audio 7118 RTP/AVP 0 3 8 18 5 101

a=alt:1 1 : A95B1E03 000000F5 10.129.39.111 7118

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv

 

DTMF via H.245 UII (out-of-band, IP suppressed) - Digits are propagated using H.245 signaling. This option is not applicable for SIP Signaling.

If the far end does not support what is configured for Digit Relay the IMG will send DTMF In-band. If this occurs, the call trace will indicate the following:

20:41:07.240 CALL(225) (00:0001:01) Falling Back to In-Band Digit Tx

 

DTMF Digit Mapping - The following is the mapping of received digits.

Event

Decimal

0-9

0-9

*

10

#

11

A-D

12-15

Flash

16

Digit Relay Packet Type:

This numeric value (96-101, 104, 106-127) defines the packet type used for Digit Relay. DTMF digit relay packet type is configurable to any value type in the range (96 - 127). Most endpoints default to 101. However, since this setting is not negotiated during call setup, it is important that the IMG is configured to match the remote endpoint setting.

Modem Behavior:

Bypass - Switches to another codec when you are in a modem call. The codec that you switch to is specified in the Fax Bypass Codec field. For example, if you are using a low bit rate codec, such as G.729, a modem or fax call will probably not be successful. So in this case the IMG changes the codec to what is configured in the Fax Bypass Codec field. This functionality is not supported for the AMR and EVRC codecs.

Disabled:

  • If Modem Behavior is set to Disabled and Fax Mode is set to 'Enable Relay (T.38)', any detected data is assumed to be T.38 fax related. RTP will be shut down and then T.38 protocol will be started.

  • If Modem Behavior is set to Disabled and Fax Mode is set to 'Enable Bypass', any detected data is assumed to be fax related. RTP will switch to G711 ulaw/alaw and fax bypass will be started.

  • If Modem Behavior is set to Disabled and Fax Mode is set to 'Relay Fallback to Bypass', any detected data will be fax related. RTP will be shutdown and signaling will be decided through SIP signaling. (Whether to go to g.711 or T.38)

The Disable Modem feature was added in software 10.5.1 ER2 and is backwards compatible so that it is supported on both Mindspeed and Broadcom VoIP modules.

Baudot Mode:

Enables the detection of baudot tones. Upon such detection, The IMG 1010 uses a custom R-URI parameter, "tty-ind", within a SIP Re-INVITE  to indicate the remote SIP UA that the peer call leg uses baudot tone. 

  • Not supported - No Re-INVITE will be invoked 

  • Detect - a Re-INVITE call flow will be used to notify the remote SIP UA

H.245 Outbound Tunneling:

This field enables/disables the H.245 tunneling feature for outbound calls to be used in specific channel groups. To enable outbound tunneling, you must also enable tunneling globally from the H.323 Signaling object. By enabling tunneling globally you enable tunneling for inbound calls. This field is ignored when SIP signaling is used.

Initial Media Inactivity Detection:

The Initial Media Inactivity Timer was added in software 10.5.0 er1. This timer is started when the channel is setup/outseized. If no RTP packets are received for the configured amount of time, an event is generated to the signaling layers. The Default value for the Initial Media Inactivity Timer field is Disabled. To Enable the Initial Media Inactivity Timer, click in the Initial Media Inactivity Timer field and select Enable. When you Enable the Initial Media Inactivity Timer field, another field, Initial Media Inactivity Detection Value will appear just below it. This field will allow configuring a value in seconds to this timer.

Media Inactivity Timer:

The Media Inactivity Timer was added in software 10.5.0 ER1. This timer is used to indicate that RTP packets have stopped flowing for the configured amount of time. When RTP packets stop flowing for the configured amount of time, an event is generated to the signaling layers and the signaling releases the channel. The Default value for the Media Inactivity Timer field is Disabled. To enable the Media Inactivity Timer, click in the Media Inactivity Timer field and select Enable. When you Enable the  Media Inactivity Timer field, another field , Media Inactivity Detection Value, will appear just below it. This field will allow configuring a value in seconds to this timer.

Comedia Mode:

See Symmetric NAT Traversal for more information.

Disable - Feature disabled (Default)

Active - This applies to SIP only. It used when the IMG is behind a NAT and you want the IMG to send or accept SIP INVITEs. The IMG will  convey that it is behind a NAT to the distant endpoint in the SDP. The IMG advertises to other gateways to change their RTP port and IP address to map with the NAT by sending the direction attribute a=direction<active> in the SIP SDP.

Passive - Used when the IMG is on a public network and you want the IMG to allow calls to a distant endpoint that is behind a NAT. The IMG uses the Source IP and Ports of the incoming RTP, RTCP, and T38 packets as the Destination IP and Port of the outgoing RTP, RTCP, and T38 packets.

Source Port Validate: (Software 10.5.1):

Select from drop down menu whether to enable or disable Source Port Validate Feature.

Enable (Default) - RTP Source Port Validation is enabled and the source IP address and UDP port of the incoming RTP packet is examined to ensure that the packet came from the IP address and UDP port number that the IMG is transmitting RTP data to.

Disable - The feature is disabled

 

See RTP Source Port Validate for more information.

Object Table (Software 10.5.0+):

This table lists the Supported Vocoders to be used by SIP or H.323 for codec negotiation during call. The entries are listed in descending priority.

Pane Appearance in Old Software Versions

Before 10.5.1:

10.5.1:

10.5.3 SP5:

 

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