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Using Unified Messaging Tools

Exchange 2010
 

Applies to: Exchange Server 2010 SP3, Exchange Server 2010 SP2

Topic Last Modified: 2010-05-24

The new Unified Messaging (UM) reporting tools added in Microsoft Exchange Server 2010 Service Pack 1 (SP1) can be used to gather usage statistics for UM servers and call statistics for UM-enabled users in your organization. You can access Unified Messaging server statistic reports by using the Call Statistics tool and access call logs for UM-enabled users by using the User Call Logs tool.

Both tools are found in the Toolbox node of the Exchange Management Console. These reports are displayed in the Exchange Control Panel.

The reports generated by the new tools in Exchange 2010 SP1 provide aggregated statistical information about calls for UM servers and calls for UM-enabled users in your organization. With the new tools, reports:

  • Are more scalable than the reports available in Exchange Server 2007.

  • Give tenant administrators, cross-premises administrators, and on-premises administrators the ability to gather statistics about the UM servers and UM-enabled users in their organizations.

  • Provide summaries from the data that's gathered. This data can be stored for 90 days and to be archived for up to 2 years to meet retention requirements.

Administrators can use the new reports to:

  • Verify how UM servers deployed in the organization are used over a given period of time.

  • Plan for UM server capacity for their on-premises organization.

  • Easily verify the availability of the voice mail system and UM servers in the organization for a given period of time.

  • Verify the overall audio quality for incoming calls to UM servers that are deployed.

To support the UM reporting tools in the EMC, the following cmdlets have been added for SP1:

  • Get-UMCallSummaryReport

  • Get-UMCallDataRecord

In Exchange 2007, 3 reports related to the Unified Messaging server role were available to administrators who used System Center Operations Manager. These Unified Messaging reports were based on the values of performance counters on each UM server that was deployed in an organization. However, generating reports using performance counters on each UM server had limitations. When System Center and UM performance counters were used to create reports based on aggregated data from all the UM servers in an organization and a user's call data, the results weren't scalable and couldn't be used in cross-premises organizations.

Without the ability to generate scalable reports or reports that could be used cross-premises, the administrator of an organization wouldn't be able to:

  • Verify how UM servers deployed in the organization were used over a given period of time. This was a critical issue for a tenant administrator or an administrator in a cross-premises deployment, because no information, including performance counters from the UM server, would be available to them.

  • Plan for UM server capacity for their on-premises organization.

  • Easily verify the availability of the voice mail system and UM servers in the organization for a given period of time.

  • Verify the overall audio quality of the UM servers that were deployed.

The Call Statistics tool provides aggregated statistical information about calls forwarded to or placed by UM servers and can be used by administrators who are interested in overall statistics for the Exchange 2010 Unified Messaging servers in their organization. Call statistics reports that you initiate in the EMC are displayed in the Exchange Control Panel user interface.

Call Statistics report displayed in the Exchange Control Panel

Call statistics report displayed in the ECP

Reports can be filtered to show call statistics by month or by day for the past 90 days or since UM was deployed in your organization. You can then filter these results by UM dial plan and UM IP gateway within your organization.

Call statistics reports display:

  • The total number of calls organized by type of call (for example, missed calls, Outlook Voice Access calls, or fax calls).

  • Whether the call was accepted or rejected.

  • The average audio quality.

  • The day or the month covered in the report, or all calls.

You can export the call logs to a Microsoft Office Excel template, or copy the call statistics information to the Clipboard so that it can be pasted into another application. You can use the Audio Quality Details button to display more specific information about the call, including the information in the following table.

 

Data Description

Date

The date and time of the incoming or outgoing call.

UM dial plan

The name of the UM dial plan that's associated with the UM IP gateway used for handling the incoming or outgoing call.

UM IP gateway

The name of the UM IP gateway used for the incoming or outgoing call.

Type of audio codec

The audio codec that's used when sending RTP or SRTP data across a network. The audio codec can be RTAudio (Wide band), RTAudio (Narrow band), G.711, or G.723.1, depending on the audio codec that's configured on the IP gateway or IP PBX or whether Microsoft Office Communications Server 2007 R2 or a later version is used.

NMOS

The mean opinion score for the audio across the network. The UM audio quality indicator will be calculated based on the Network MOS (NMOS) that's gathered from the RTP (and SRTP).

“Mean opinion score” (MOS) is a number on a scale from 1 to 5 (5 being excellent) that indicates the audio quality of the call. MOS metrics are directly linked to the audio codec that's used. This means that users will get a different audio quality if they use different audio codecs.

The following is the NMOS maximum for the audio codecs that are supported:

  • RTAudio (Wide band): 4.10

  • RTAudio (Narrow band): 2.95

  • G.711 a/u: 3.61

  • G.723.1: 2.63

NMOS degradation

Total NMOS degradation is how far the reported NMOS value is from the top value for the audio codec that was used for the call.

Jitter

The average Jitter for the incoming or outgoing call. In data networks, the term jitter is used as a measure of the packet latency across a network. A network with constant latency has no variation (or jitter). Jitter is sometimes expressed as the average deviation from the network mean latency.

Pack loss

The average percentage of network packet loss during the call.

Round trip time

For the selected call, this is the time, in milliseconds, for a round trip (between the UM IP gateway and the UM server) of the audio packets that carried the voice data over the network.

Burst Loss Duration

The average duration of packet loss during burst losses for the incoming call.

You can use the User Call Logs tool to display the call statistics for a selected UM-enabled user. The report is displayed in the Exchange Control Panel and is useful in helpdesk-type situations where you have to gather information about specific calls for a UM-enabled user to assist them in diagnosing and fixing issues. After you click the Select a user button and specify the user, the following information will be displayed for calls of the user you selected:

  • Date and time

  • Duration of the call

  • Type of call

  • The calling number

  • The called number

  • The UM IP gateway

  • Audio quality

User Call Logs report displayed in the Exchange Control Panel

User call logs report displayed in the ECP

You can copy the user's call statistics to the Clipboard and then paste them into another application. You can use the Audio Quality Details button to display more specific information about the call, including the information in the following table.

 

Data Description

Date

The date and time of the incoming or outgoing call.

UM dial plan

The name of the UM dial plan that's associated with the UM IP gateway used for handling the incoming or outgoing call.

UM IP gateway

The name of the UM IP gateway that was used for the incoming or outgoing call.

Type of audio codec

The audio codec that's used when sending RTP or SRTP data across a network. The audio codec can be RTAudio (Wide band), RTAudio (Narrow band), G.711, or G.723.1, depending on the audio codec that's configured on the IP gateway or IP PBX or whether Office Communications Server R2 or a later version is used.

NMOS

The mean opinion score for the audio across the network. The UM audio quality indicator will be calculated based on the Network MOS (NMOS) that's gathered from the RTP (and SRTP).

“Mean opinion score” (MOS) is a number on a scale from 1 to 5 (5 being excellent) that indicates the audio quality of the call. MOS metrics are directly linked to the audio codec that's used. This means that users will get a different audio quality if they use different audio codecs.

The following is the NMOS maximum for the audio codecs that are supported:

  • RTAudio (Wide band): 4.10

  • RTAudio (Narrow band): 2.95

  • G.711 a/u: 3.61

  • G.723.1: 2.63

NMOS degradation

Total NMOS degradation is how far the reported NMOS value is from the top value for the audio codec that was used for the call.

Jitter

The average Jitter for the incoming or outgoing call. In data networks, the term jitter is used as a measure of the packet latency across a network. A network with constant latency has no variation (or jitter). Jitter is sometimes expressed as the average deviation from the network mean latency.

Pack loss

The average percentage of network packet loss during the call.

Round trip time

For the calls within the selected time range, this is the average time, in milliseconds, for a round trip (between the UM IP gateway and the UM server) of the audio packets that carried the voice data over the network.

Burst Loss Duration

The average duration of packet loss during burst losses for the incoming call.

Number of calls sampled

This is the total number of incoming calls that were sampled to determine the audio quality for the call.

For detailed information about other tools in the EMC Toolbox, see Managing Tools in the Toolbox.

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