Use this pane to define a set of VoIP parameter values such as Silence Suppression, Echo Cancellation and RTP Redundancy. After creating and configuring the IP Profile object, it can then be assigned to a SIP channel group to define inbound or outbound behavior. To create the IP Profile object, right click on the Profiles object and select New IP Profile. After creating the IP Profile object, commit the profile by going back to the Profiles object pane and click the Download Tables button. After creating the profile, use the Supported Vocoders pane to assign a list of supported codecs to the profile.

The IMG 2020 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.

For codecs G729, and G723, when High Jitter is selected in webUI, values for minimum, nominal and maximum delays are changed to hard coded values:

G729

G723

Web GUI Page

Dialogic > Profiles > IP Profiles > New IP Profile

Maximum Objects

Up to 16 IP Profiles can be created per Profiles object.

Related Topics and Dependencies

After creating and configuring the IP Profiles object, assign it to a SIP channel group to define inbound or outbound behavior.

IP Profiles

Vocoder Profile

Type of Service (TOS) Settings

Video Profile

Field Descriptions

Name

Click in the Name field and enter a name that identifies the IP Profile being created. A maximum of 19 characters is allowed. Click on the Valid Characters link. The link displays the valid characters that can be used in the naming process. The Name value provides a name to the profile being defined. The Name is subsequently used in the Channel Group object to specify the IP Bearer Profile instance.

Silence Suppression

During a normal voice conversation, much of the time is wasted on silence from one or both ends. Ethernet bandwidth can be conserved if during these periods of silence, RTP packets are sent with silence-encoded compressed payloads. The IMG 2020 Silence Suppression feature supports the concepts of Voice Activity Detection (VAD) and Comfort Noise Generation (CNG). When enabled, Silence suppression will not send RTP traffic during periods of silence, saving bandwidth usage. At the beginning of a silence period, a single packet will be sent to the distant end to inform it that a period of silence is being entered, and that the distant end should begin to regenerate comfort noise to its TDM stream. Silence Suppression can be enabled or disabled on an established connection. Silence Suppression does not apply to the AMR codec as it has its own internal Silence Suppression scheme.

Enable - Silence Suppression algorithm is enabled

Disable - Silence Suppression algorithm is disabled.

For SIP calls silence suppression is enabled by default. In the default configuration the attribute is not sent unless it is specifically set to off. In the case of Fax re-invites if Silence Suppression is set to off the line a=silenceSupp:off - - - - is included in the SIP SDP message.

Echo Cancellation

In compliance with ITU G.168-2000, this feature eliminates echo in the case of a PSTN to VoIP call that is introduced by impedance mismatched hybrids on TDM channels. The maximum tail length is 128 milliseconds for the DSP Modules installed in the IMG 2020. 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. Echo cancellation is not supported in the AMR codec.

A Non-Linear Processor control support feature is included in the selections below. This subcomponent of Echo Cancellation can be enabled and disabled when Echo cancellation is set to enabled. See 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.

Echo Cancellation provided by the IMG 2020 is not applicable in case of TDM<>TDM calls. In this case the user may want to implement Echo Cancellation externally 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. 

RTP Redundancy

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

No Redundancy - RTP packet redundancy is disabled.

Redundancy Level 1 - RTP packet redundancy is enabled. When redundancy level 1 is selected, the RTP Payload Type Redundancy field below can now be modified and the packet payload type can be modified. 

RTP Payload Type for Redundancy

This field can be modified when RTP Redundancy field above is set to Redundancy Level 1.

The numeric value displayed in drop down menu (96-127) defines the packet type being sent for RTP Redundancy. For an incoming call, if the sending gateway is using a codec that is configured on the IMG 2020 but is sent with a different dynamic payload type, the IMG 2020 will accommodate that by responding with the dynamic payload type sent by the sending gateway in the INVITE message.  For example, if the IMG 2020 has an AMR profile configured with a value of 100, an incoming call to the IMG 2020 has AMR set to 98, the IMG 2020 will send the 98 back in the 200 OK to accept the AMR Codec.

Digit Relay

This setting specifies the method used to propagate DTMF digits. The Digit Relay field does not apply to the AMR codec.

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

DTMF Packetized- Digits are sent in a 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 example Below.

Example of "telephone event" line in SDP:

DTMF via Signaling - Digits are sent separately.

If the far end does not support what is configured for Digit Relay the IMG 2020 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.

Fax Mode

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 codec.

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 2020 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. Below are the selections in drop down menu This field does not apply to the AMR codec.

Fax Packet Redundancy

Provides Fax packet redundancy to guard against network packet loss. This is only applicable to Relay Fax Mode. This field does not apply to the AMR codec.

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. 

Initial Media Inactivity Timer

This timer is started when the DS-0 is setup/outseized. If no RTP packets are received for the configured amount of time configured in the Initial Media Inactivity Timer Value below, an event is generated to the signaling layers. The Default value for the Initial Media Inactivity Timer field is Disable. To Enable the Initial Media Inactivity Timer, click in the Initial Media Inactivity Timer field and select Enable from drop down menu. When the Initial Media Inactivity Timer field is enabled, the Initial Media Inactivity Timer Value field below it can now be modified. See below.

Initial Media Inactivity Timer Value (Default = 181 seconds)

Set the Initial Media Inactivity Timer Value by either adjusting the box on the sliding scale or click in the Seconds box and enter the desired value.

The Initial Media Inactivity Timer Value is shaded green and cannot be modified. Once the Initial Media Inactivity Timer field above is set to enabled, the Initial Media Inactivity Timer Value field will change to white and can now be modified.

Media Inactivity Timer

The Media Inactivity Timer is used to indicate that RTP packets have stopped flowing for the configured amount of time configured in the Media Inactivity Timer Value below. 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 Disable. To enable the Media Inactivity Timer, click in the Media Inactivity Timer field and select Enable from drop down menu. When the Media Inactivity Timer field is set to enable, the Media Inactivity Timer Value field can now be modified. Refer to Media Inactivity Timers (RTP) for more information.

Disable - Media Inactivity Timer is disabled.

Enable - Media Inactivity Timer is enabled. The Initial Media Inactivity Timer Value below will need to be set. 

Media Inactivity Timer Value (Default = 30 seconds)

Set the Media Inactivity Timer Value by either adjusting the box on the sliding scale or click in the Seconds box and enter the desired value. For reference, view the screen capture in the Initial Media Inactivity Timer Value field above. Below is a table displaying the default value as well as the range of values that can be configured.

The Media Inactivity Timer Value is shaded green and cannot be modified. Once the Media Inactivity Timer field above is set to enabled, the Media Inactivity Timer Value field will change to white and can now be modified.

CAUTION: Depending on the Network conditions, setting the Initial Media Inactivity Timer or Media Inactivity Timer to a low value may cause calls to release before they have completely established. 

Digit Relay Packet Type

This numeric value (96 - 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). The Default is 101 which is what most endpoints default to. However, if it is not negotiated during call setup, the IMG 2020 will accept the remote endpoint value if it does not match the value configured.

Digit Relay Packet Type (16000)

This numeric value (96 - 127) defines the packet type used for Digit Relay for AMR-WB codec. DTMF digit relay packet type is configurable to any value type in the range (96 - 127). The Default is 102 which is what most endpoints default to. However, if it is not negotiated during call setup, the IMG 2020 will accept the remote endpoint value if it does not match the value configured.

Modem Behavior

Bypass (Default) - Switches to another codec when in a modem call. The codec that it is switched to is specified in the Fax Bypass Codec field. For example, if 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 2020 changes the codec to what is configured in the Fax Bypass Codec field. This functionality is not supported for the AMR codec.

Disabled:

• If Modem Behavior is set to Disable, 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 Disable, 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 Disable, 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) 

Baudot Mode

Not supported (Default) -  baudot tone detection is not supported.

Detect: When baudot tone is detected, the SIP layer will trigger a SIP Re-INVITE

Source Port Validate

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 2020 is transmitting RTP data to.

Disable - The feature is disabled.

For more information, refer to RTP Source Port Validate topic.

High Jitter

Select from drop down menu whether to enable or disable High Jitter.

Enable  - High Jitter is enabled.

Disable - The feature is disabled.

SIP Maximum Call Duration

Select from drop down menu whether to enable or disable SIP Maximum Call Duration.

Enable  - SIP Maximum Call Duration is enabled. When the IMG 2020 receives an incoming call for a channel group that has the incoming IP Profile set with the SIP Maximum Call Duration field enabled, the call will be ended after the specified number of minutes specified in the SIP Maximum Call Duration Value field.

Disable - The feature is disabled.

SIP Maximum Call Duration Value (Default = 60 minutes)

Set the SIP Maximum Call Duration Value by either adjusting the box on the sliding scale or click in the Minutes box and enter the desired value. The configurable range is from 1 to 5000 minutes.

The SIP Maximum Call Duration Value is shaded green and cannot be modified. Once the SIP Maximum Call Duration Valuer field above is set to enabled, the SIP Maximum Call Duration Value field will change to white and can now be modified.

Rx and Tx Digital Gain (Default to 0DB)

Each SIP signaling type channel group has an incoming and an outgoing IP profile. The Rx digital gain property in the IP profile, will apply the gain configuration to the incoming RTP steam while the Tx digital gain property will be applied to the outgoing RTP packets. 

Possible range of values go from -20db to 20 db.

Note that in a SIP to SIP call scenario, the 2 gains will be added.