VoIP Settings page field descriptions for MPP

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.

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.

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.

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.

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

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.

• G729

• G711uLaw

• G711aLaw

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

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

The default is 20.

The options are:

• Yes: Select Yes to enable this option.

• No: Select No to disable this option.

The default is Yes.

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.

	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.

• 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.

• 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.

• 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.

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

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

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.

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.

• 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.

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.

	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.

• 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.

• 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.

• 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.