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Use this page to view the Voice over IP (VoIP) settings for Experience Portal.
This page contains the:
Field |
Description |
|---|---|
UDP |
The range of port numbers used by User Datagram Protocol (UDP) transactions.
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. |
MRCP |
The port numbers used by Transmission Control Protocol (TCP) transactions. |
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. |
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. |
Port |
The number of the port on the RTCP monitor that the EPM uses to communicate with the RTCP monitor. |
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:
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. |
Field |
Description |
|---|---|
Packet Time |
The interval in milliseconds, for transmitting each audio packet. |
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. |
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. |
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). |
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.
Field |
Description |
|---|---|
H.323 |
The H.323 QoS parameters are:
|
SIP |
The Session Initiation Protocol (SIP) QoS parameters are:
|
RTSP |
The Real-Time Streaming Protocol (RTSP) QoS parameters are:
|
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.
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. |
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. |
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. |
Field |
Description |
|---|---|
Threshold |
The options are:
|
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:
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:
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. |
Field |
Description |
|---|---|
Inband DTMF Detection Enabled |
This option allows Avaya Experience Portal to interoperate with media gateways and SIP endpoints.
The default is No. |
Pre-Energy Record Time |
The maximum number of milliseconds of audio data that are inserted in the recordings before the system detects the energy.
|
H323 Force Registration |
The options are:
The default is Never. |