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| Field | Description | ||||
|---|---|---|---|---|---|
| Call Leg | The number of the call leg. | ||||
| Contact ID | The unique identifier for the call assigned
by the Media Server. If the call uses an H.323 connection, this field is in the format MPPName-Port-SessionID, where:
If the call uses a SIP connection, this field is in the format MPPName-SIPPortGroup-SIPChannel-SessionID, where:
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| Session ID | The unique identifier for the session assigned by the Media Server. | ||||
| Port | The number of the port used for the call. | ||||
| Start Date/Time | The day and time that the call leg started. | ||||
| Contact Type | The options are:
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| Duration | The length of the call in seconds. | ||||
| Application Name | The name of the application that handled the call. | ||||
| Originating Number | The calling party number or Automatic Number Identification (ANI). | ||||
| Destination Numbers | The destination number or numbers to which this call was routed. | ||||
| Reason Code | The options are:
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| End Type | The options are:
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| End Details | Additional information about why the call leg ended, including the event that triggered the end of the call leg. | ||||
| Media Server | The name of the Media server that handled the call. | ||||
| Switch Protocol | The options are:
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| Time Until Call Connected | The number of milliseconds until the call was connected. | ||||
| First Prompt Latency | The number of milliseconds after Avaya Experience Portal received the call before the initial prompt began playing.
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| Round Trip Between Server and Far End | The delay, expressed in milliseconds,
between receiving the last SR packet from source SSRC_n and sending a reception report block.
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| Far End Reported Jitter | The Jitter reported from the far end of the RTP stream. | ||||
| Number of Packets Media Server Sent | The total number of RTP data packets transmitted by the sender. | ||||
| Number of Packets Far End Lost | The total RTP data packets lost as reported by the far end of the RTP stream. | ||||
| Media Server | An estimate of the statistical variance of the RTP data packet inter-arrival time, measured by comparing timestamp units and expressed as an unsigned integer. Jitter provides a measure of transient network congestion, where lower numbers represent better audio quality. | ||||
| Number of Packets Server Received | The total number of RTP data packets received from the source. | ||||
| Number of Packets Server Lost | The total number of RTP data
packets from source that have been lost since the beginning of reception. This is the number of packets expected minus the number of packets actually received, where the number of packets received includes any which are late, or duplicates. |