Call Detail Report (CDR)

When enabled, the Call Detail Report (CDR) feature keeps a record of every incoming, outgoing, and missed call that occurs on the system. If a call does not connect, the report shows the reason. In multipoint calls, each far site is shown as a separate call, but all have the same conference number.

The CDR database is limited to the 150 most recent entries. If you are concerned about tracking all CDR records, ensure that you download the records at regular intervals so that the limit of 150 entries is not exceeded and records are not lost.

The size of a CDR can become unmanageable if you don't download the record periodically. A full report with 150 entries results in a CDR of approximately 50 KB. Your connection speed can also affect how fast the CDR downloads. You can set up a schedule to download and save the CDR after every 120 calls to keep track of all call entries and make the file easy to download and view.

Note: The RealPresence Resource Manager system captures CDR information for the EagleEye Producer and the EagleEye Director II cameras and generates it to the RealPresence Resource Manager system CDR. The call details include People Minutes and People Count (Call Begin) at the beginning of a call and People Count (Peak Value) at the end of a call.

Data

Description

Row ID

Each call is logged on the first available row. A call is a connection to a single site, so there might be more than one call in a conference.

Start Date

The call start date, in the format dd-mm-yyyy.

Start Time

The call start time, in 24-hour format hh:mm:ss.

End Date

The call end date.

End Time

The call end time.

Call Duration

The length of the call.

Account Number

If Require Account Number to Dial is enabled on the system, the value entered by the user is displayed in this field.

Remote System Name

The far site’s system name.

Call Number 1

The number dialed from the first call field, not necessarily the transport address.

For incoming calls — The caller ID information from the first number received from a far site.

Call Number 2

(If applicable for call)

For outgoing calls — The number dialed from the second call field, not necessarily the transport address.

For incoming calls — The caller ID information from the second number received from a far site.

Transport Type

The type of call — Either H.323 (IP) or SIP.

Call Rate

The bandwidth negotiated with the far site.

System Manufacturer

The name of the system manufacturer, model, and software version, if they can be determined.

Call Direction

In—For calls received.

Out—For calls placed from the system.

Conference ID

A number given to each conference. A conference can include more than one far site, so there might be more than one row with the same conference ID.

Call ID

Identifies individual calls within the same conference.

Total H.320 Channels Used

Number of narrow-band channels used in the call.

Endpoint Alias

The alias of the far site.

Reserved

Polycom use only.

View Name

Names the web or local interface used in the call.

User ID

Lists the ID of the user who made the call.

Endpoint Transport Address

The actual address of the far site (not necessarily the address dialed).

Audio Protocol (Tx)

The audio protocol transmitted to the far site, such as G.728 or G.722.1.

Audio Protocol (Rx)

The audio protocol received from the far site, such as G.728 or G.722.

Video Protocol (Tx)

The video protocol transmitted to the far site, such as H.263 or H.264.

Video Protocol (Rx)

The video protocol received from the far site, such as H.261 or H.263.

Video Format (Tx)

The video format transmitted to the far site, such as CIF or SIF.

Video Format (Rx)

The video format received from the far site, such as CIF or SIF.

Disconnect Local ID and Disconnect Reason

The identity of the user who initiated the call and the reason the call was disconnected.

Q.850 Cause Code

The Q.850 cause code showing how the call ended.

Total H.320 Errors

The number of H.320 errors experienced during the call.

Average Percent of Packet Loss (Tx)

The combined average of the percentage of both audio and video packets transmitted that were lost during the five seconds preceding the moment at which a sample was taken. This value does not report a cumulative average for the entire call. However, it does report an average of the sampled values.

Average Percent of Packet Loss (Rx)

The combined average of the percentage of both audio and video packets received that were lost during the five seconds preceding the moment at which a sample was taken. This value does not report a cumulative average for the entire call. However, it does report an average of the sampled values.

Average Packets Lost (Tx)

The number of packets transmitted that were lost during a call.

Average Packets Lost (Rx)

The number of packets from the far site that were lost during a call.

Average Latency (Tx)

The average latency of packets transmitted during a call based on round-trip delay, calculated from sample tests done once per minute.

Average Latency (Rx)

The average latency of packets received during a call based on round-trip delay, calculated from sample tests done once per minute.

Maximum Latency (Tx)

The maximum latency for packets transmitted during a call based on round-trip delay, calculated from sample tests done once per minute.

Maximum Latency (Rx)

The maximum latency for packets received during a call based on round-trip delay, calculated from sample tests done once per minute.

Average Jitter (Tx)

The average jitter of packets transmitted during a call, calculated from sample tests done once per minute.

Average Jitter (Rx)

The average jitter of packets received during a call, calculated from sample tests done once per minute.

Maximum Jitter (Tx)

The maximum jitter of packets transmitted during a call, calculated from sample tests done once per minute.

Maximum Jitter (Rx)

The maximum jitter of packets received during a call, calculated from sample tests done once per minute.

Call Priority

The AS-SIP call precedence level assigned to the call (populated only when AS-SIP is enabled on the system).