VoIP Settings page field descriptions for MPP

Last Updated : Jun 05, 2026 |

Use this page to configure Voice over IP (VoIP) on your Experience Portal system.

This page contains the:

Port Ranges group

Field

Description

UDP

The range of port numbers used by User Datagram Protocol (UDP) transactions.

Enter the lower value of the range of port numbers in the Low field, and the higher value in the High field. The range must be within 1024 to 65535.

The default range is 11000 to 30999.

Note:

Each call that uses ASR and TTS resources requires a total of six UDP ports.

TCP

The TCP range which you specify in this field must be large enough for the network connections, which vary based on the configuration as well as the load.

Enter the lower value of the range of port numbers in the Low field, and the higher value in the High field. The range must be within 1024 to 65535.

The default range is 31000 to 33499.

Important:

Do not limit the TCP range to the absolute minimum as it will possibly impact functionality such as failover.

Do not overlap ranges for the TCP and MRCP protocols.

MRCP

The port numbers used by Transmission Control Protocol (TCP) transactions.

Enter the lower value of the range of port numbers in the Low field, and the higher value in the High field. The range must be within 1024 to 65535.

The default range is 34000 to 36499.

The number of MRCP ports you need depends on the setting of the New Connection per Session option for the Experience Portal speech servers.

If this option is enabled, you need one MRCP port for each speech server license. If this option is not enabled, you need one MRCP port per speech server.

For example, if you have one ASR server with five ASR licenses and one TTS server with two TTS licenses and:

  • The New Connection per Session option is enabled for both the ASR and TTS server, you would need seven MRCP ports because you have seven total licenses.

  • The New Connection per Session option is enabled for the ASR server but not the TTS server, you would need six MRCP ports because you have five ASR licenses plus one TTS server.

  • The New Connection per Session option is not enabled for either the ASR or TTS server, you would need two MRCP ports because you have two speech servers.

H.323 Station

The H.323 Station port range configures a range of UDP ports that are used exclusively for gatekeeper discovery and registration. However, the bulk of H.323 communication occurs over a TCP socket that is allocated from the TCP range. For each H.323 station, you need to configure one UDP and one TCP port. If either port fails to be allocated, the H.323 station will be marked out of service.

Enter the lower value of the range of port number in the Low field, and the last number of the range in the High field. The range must be within 1024 to 65535.

The default range is 37000 to 39499.

Important:

The H.323 Station range must not overlap with UDP range.

RTCP Monitor Settings group

Field

Description

Host Address

The network address of the RTCP monitor, which collects status data about RTP sessions from the MPP and other components in the system.

This must be a valid network address in the form of a fully qualified hostname or an IP address.

Port

The number of the port on the RTCP monitor that the EPM uses to communicate with the RTCP monitor.

VoIP Audio Formats group

Field

Description

MPP Native Format

The audio encoding codec the MPP uses as the default for audio recording within the Avaya Voice Browser (AVB) when the speech application does not specify the format for recording caller inputs.

The options are:

  • audio/basic: The AVB uses the mu-Law encoding format, which is used mostly in the United States and Japan.

    If you select this option then the codec set on the switch must include G711MU.

  • audio/x-alaw-basic: The AVB uses the A-Law encoding format, which is used in most countries other than the United States and Japan.

    If you select this option then the codec set on the switch must include G711A.

With either option, the AVB records input using a G.711-compliant format that is a raw (headerless) 8kHz 8-bit mono [PCM] single channel format.

Note:

The AVB ignores this setting if a recording format is specified in a given speech application.

If you make any change to the setting of this field, you must restart the MPP for the changes to take effect.

Audio Codecs group

Field

Description

Offer

When sending a SIP INVITE, Experience Portal offers the supported codecs in a priority order that the administrator can configure. The default order is:

  • G729

  • G711uLaw

  • G711aLaw

Answer

When receiving a SIP INVITE, Experience Portal accepts the supported codecs based on a priority order that the administrator configures.

By default, Experience Portal accepts the first codec offered by the other side that Experience Portal supports.

Packet Time

The interval in milliseconds, for transmitting each audio packet.

The time intervals you can select are: 10, 20, 30, 40, 50, 60, 70, and 80.

The default is 20.

G729

G.729 codec is used for audio data compression for both H.323 and SIP connections. It supports G.729 Annexes A and B.

The options are:

  • Yes: Select Yes to enable this option.

  • No: Select No to disable this option.

The default is Yes.

Reduced Complexity Encoder

The G.729A reduced complexity encoding algorithm lowers the performance cost of G.729 transcoding. This setting affects only the encoding of G.729 audio sent by Experience Portal. The audio quality is reduced slightly when you enable this option.

Experience Portal continues to receive and decode G.729 and G.729A audio data, regardless of the option selected in this field.

Note:

This field is enabled only if you have selected Yes in the G729 field.

The options are:

  • Yes: Select Yes to enable this option.

  • No: Select No to disable this option.

The default is Yes.

Discontinuous Transmission

The G.729B discontinuous transmission algorithm allows Experience Portal to access and process a far end media offer with G.729B. The Annexe B specification further reduces network bandwidth as it sends only the audio packets that contain speech data (packets that contain silence are not transmitted).

Note:

This field is enabled only if you have selected Yes in the G729 field.

The options are:

  • Yes: Select Yes to enable this option.

  • No: Select No to disable this option. With this option Annex B is not used.

    Note:

    The G.729 offers may still be accepted.

QoS Parameters group

Quality of Service (QoS) is used in network routing to improve performance for certain data streams. For example, RTP negotiated by various signaling protocols, but not the signaling itself. This is especially valuable for VoIP traffic because VoIP is susceptible to jitter caused by network delays. The QoS settings in this group are defined as per the signaling protocols parameters, but apply to the RTP streams that are the result of these signaling connections. This allows the various categories of RTP data to be prioritized independently. The QoS settings, however, do not apply to the signaling connections which are much less sensitive to latency and bandwidth limitations.

Note:

The QoS settings are not in a continuous range. Increasing or decreasing the values will disable QoS. The numbers must exactly match the configuration on the network routers for the settings to have any effect. Therefore, if you are using QoS and the defaults do not seem to be working, contact your network administrator for suggested values.

Field

Description

H.323

The H.323 QoS parameters are:

  • VLAN. The QoS settings for H.323 connections running over a virtual LAN.

  • Diffserv. The QoS settings for H.323 connections running over a network using the Differentiated Services architecture.

The default for VLAN is 6 and the default for Diffserv is 46.

SIP

The Session Initiation Protocol (SIP) QoS parameters are:

  • VLAN. The QoS settings for SIP connections running over a virtual LAN.

  • Diffserv. The QoS settings for SIP connections running over a network using the Differentiated Services architecture.

The default for VLAN is 6 and the default for Diffserv is 46.

RTSP

The Real-Time Streaming Protocol (RTSP) QoS parameters are:

  • VLAN. The QoS settings for Real RTSP running over a virtual LAN.

  • Diffserv. The QoS settings for RTSP running over a network using the Differentiated Services architecture.

The default for VLAN is 6 and the default for Diffserv is 46.

Out of Service Threshold group

The Trigger settings in this group determine when an MPP server issues an event or alarm message based on the percentage of ports that have gone out of service. In all cases, once the MPP server has issued an event or alarm message, it will not issue another message until the percentage of out of service ports changes to the value set in the associated Reset field or below.

For example, if the Warn Trigger value is 10 and the Reset value is 0, then the MPP will respond in the following manner as the percentage of out of service ports changes:

Percentage of out of service ports

MPP server response

10%

A warning event is generated and the MPP enters the Degraded state.

8%

No event is generated.

12%

No event is generated because it has not yet fallen below the Reset value.

0%

No event is generated but the MPP returns to the Running state.

6%

No event is generated.

11%

An event is generated and the MPP returns to the Degraded state.

Note:

When the warning event is generated after the percentage of out of service ports reaches the trigger value, no more warning is generated until you reach reset value again.

Field

Description

Warn

The Trigger field determines the percentage of ports that must go out of service on an MPP before the MPP sends a warning-level event to Experience Portal and enters the Degraded state.

Once the Reset value is reached, the MPP returns to the Running state.

The Trigger default is 10 and the Reset default is 0.

Error

The Trigger field determines the percentage of ports that must go out of service on an MPP before the MPP sends an error-level event to Experience Portal and enters the Degraded state.

Once the reset value is reached, the MPP will send another error event when appropriate, but it does not return to the Running state until the Reset value associated with the Warn field has been reached.

The Trigger default is 20 and the Reset default is 10.

Fatal

The Trigger field determines the percentage of ports that must go out of service on an MPP before the MPP issues a fatal-level alarm and enters the Degraded state.

Once the reset value is reached, the MPP will send another fatal event when appropriate, but it does not return to the Running state until the Trigger value associated with the Warn field has been reached.

The Trigger default is 100 and the Reset default is 50.

Call Progress group

Field

Description

Threshold

The options are:

  • Voice — The Voice Threshold parameter is used by the call classification engine in determining whether a given frame of audio data should be interpreted as voice energy. Lower values are more inclusive and tends toward marking even white background noise as voice. Higher values will reject more noise, picking out only audio frames that are very clearly voice sounds. The range is 0.0-1.0 and the default value is 0.50

  • Tone — The Tone Threshold parameter is used conjunctively along with Periodicity Threshold parameter by the call classification engine to determine if a given frame of audio data contains a pure telephony tone. Lower values for either or both will allow the call classification engine to accept more distorted signals as valid telephony tones, but makes certain voice sounds more likely to be detected as pure tones, for example, Talk Off. The range is 0.0-1.0 and the default value is 0.95.

  • Periodicity — The Periodicity Threshold parameter is used conjunctively with Tone Threshold by the call classification engine to determine if a given frame of audio data contains a pure telephony tone. Lower values for either or both will allow the call classification engine to accept more distorted signals as valid telephony tones, but makes certain voice sounds more likely to be detected as pure tones, for example, Talk Off. The range is 0.0-1.0 and the default is 0.97.

  • Ring Count — Ring Count Threshold detects the number of ring back cycles . The purpose is to let the call progress engine to use different timers when a call is answered “quickly”, as determined by the ring count, tailoring the parameters appropriately under the assumption that calls answered after a couple of rings are more likely to be a live person. The cycles range from 0-8 and the default value is 4.

Cut Through

The Cut Through time is the number of milliseconds of consecutive silence after some voice energy that must be heard to determine that a live speaker is done talking.

Note:

Setting this value lower, increases the responsiveness of the system. The system finishes Live voice detection to finish sooner and the VoiceXML dialog starts playing sooner. However, the system risks false detections for recordings where there are long gaps between speech energy. For example, a voice mail system might play a short recorded greeting, followed by a TTS name, followed by some more recorded prompts. If the silence between the recorded greeting and the TTS name is longer than the cut through time, the voice mail machine is mistakenly identified as a live person. Setting the value higher increases the likelihood that a call is properly classified.

The following are the values of Cut Through time:

  • Initial — The Initial values are loaded first when the call classification engine is initialized for any given call. The value ranges from 200-2000 milliseconds and the default is 1100 milliseconds.

  • Short — The values for the Short timers are loaded when the call classification engine detects at least one ring back cycle . The value ranges from 200-2000 milliseonds and the default is 700 milliseconds.

  • Long — The values for the Long timers are loaded if the call classification engine detects successive ring back cycles and the count equals or exceeds the Ring Count Threshold. The value ranges from 200–2000 milliseconds and the default is 1100 milliseconds.

Which value is used is determined by the number of ring back cycles heard by the call classification engine and the Ring Count Threshold. The purpose is to let the call progress engine to use different timers when a call is answered “quickly”, as determined by the ring count, tailoring the parameters appropriately under the assumption that calls answered after a couple of rings are more likely to be a live person.

Max Voice

The Max Voice time is the number of milliseconds of continuous voice energy except gaps of silence shorter than the Cut Through time that must be heard to determine that a greeting is a recorded message.

Note:

Setting this value lower biases the call classification engine toward detecting answering machines. All but very short greetings are assumed to be answering machines. Conversely, setting the value higher biases the call classification engine toward detecting live voice. For instance, a greeting must be very long to be considered an answering machine.

The following are the values of Max Voice time:

  • Initial — The Initial value is loaded first when the call classification engine is initialized for any given call. The value ranges from 1100-4000 milliseconds and the default is 2500 milliseconds.

  • Short — The values for the Short timers are loaded when the call classification engine detects at least one ring back cycle. The value ranges from 1100–4000 milliseconds and the default is 2500 milliseconds.

  • Long — The values for the Long timers are loaded if the call classification engine detects successive ring back cycles and the count equals or exceeds the Ring Count Threshold. The value ranges from 1100–4000 milliseconds and the default is 2500 milliseconds.

Which value is used is determined by the number of ring back cycles heard by the call classification engine and the Ring Count Threshold. The purpose is to let the call progress engine to use different timers when a call is answered “quickly”, as determined by the ring count, tailoring the parameters appropriately under the assumption that calls answered after a couple of rings are more likely to be a live person.

Miscellaneous group

Field

Description

Inband DTMF Detection Enabled

This option allows Experience Portal to interoperate with media gateways and SIP endpoints

  • Yes: Select Yes to enable this option.

  • No: Select No to disable this option.

The default is No.

Note:

If you make any change to the Inband DTMF Detection Enabled field, you must restart the MPP for the changes to take effect.

Pre-Energy Record Time

The maximum number of milliseconds of audio data that are inserted in the recordings before the system detects the energy.

  • Range: 0 to 30000.

  • Default: 0

Note:

If you make any change to the Pre-Energy Record Time field, you must restart the MPP for the changes to take effect.

H323 Force Registration

Controls the behavior of the H.323 station registration process. The settings are:

  • Always: The setting Always causes the MPP to forcibly register an extension, immediately bringing a port into service by, potentially, taking control from any other endpoint that may currently have the extension registered. Any calls in progress on the other endpoint are dropped when control of the station is removed.

  • Never: The default setting of Never causes station registration to fail if a registering extension is already in use and registered by another endpoint. The port does not immediately come into service, but the station remains in use by the other endpoint until the station is released or unregistered.

The default is Never.