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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.
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| 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.
|
| 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:
|
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| 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.
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. |
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| 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.
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. |