Details

No

When you click this item, the report shows additional information on the call.

Caller

Yes

SIP address of the person who initiated the call.

Callee

Yes

SIP address of the person who was called.

Start time

Yes

Date and time that the call started.

End time

Yes

Date and time that the call ended.

Caller user agent

Yes

Software used by the endpoint of the person who initiated the call.

Callee user agent

Yes

Software used by the endpoint of the person who was called.

Round trip (ms)

Yes

Average amount of (in milliseconds) required for a real-time transport protocol (RTP) packet to travel to another endpoint and then back. Round-trip times of 100 milliseconds or less are considered of acceptable quality.

High round-trip values can be caused by international call routing, a routing misconfiguration, or an overloaded media server. High round-trip times result in difficulties with two-way, real-time audio conversations.

Degradation (MOS)

Yes

Average amount of mean opinion score (MOS) degradation experienced during a call. Degradation values can range from a low of 0.0 to a high of 5.0. A value of 0.5 or less represents acceptable degradation. Historically, mean options scores were calculated by having users rate the quality of a call on a scale of 1-to-5. In Microsoft Lync Server 2010, the Monitoring Server uses a set of algorithms to predict how users would have rated a call.

High degradation values can be caused by congestion, lack of bandwidth, wireless congestion or interference, or an overloaded media server or endpoint. High degradation results in distorted or lost audio.

Packet loss

Yes

Average rate of RTP packet loss. (Packet loss occurs when RTP packets, a protocol used for transmitting audio and video across the Internet, failed to reach their destination.) High loss rates are generally caused by congestion, lack of bandwidth, wireless congestion or interference, or an overloaded media server. Packet loss typically results in distorted or lost audio.

Jitter

Yes

Average jitter detected between RTP packet arrivals. (Jitter is a measure of the "shakiness" of a call.) High jitter values are typically caused by congestion or an overloaded media server, and result in distorted or lost audio.

Healer concealed ratio

Yes

Average ratio of concealed audio samples to the total to the total number of samples. (A concealed audio sample is a technique used to smooth out the abrupt transition that would usually be caused by dropped network packets.) High values indicate significant levels of loss concealment applied caused by packet loss or jitter, and results in distorted or lost audio.

Healer stretched ratio

Yes

Average ratio of stretched audio samples to the total to the total number of samples. (Stretched audio is audio that has been expanded to help maintain call quality when a dropped network packet has been detected.) High values indicate significant levels of sample stretching caused by jitter, and result in audio sounding robotic or distorted.

Healer compressed ratio

Yes

Average ratio of compressed audio samples to the total number of samples. (Compressed audio is audio that has been compressed to help maintain call quality when a dropped network packet has been detected.) High values indicate significant levels of sample compression caused by jitter, and result in audio sounding accelerated or distorted.

Connectivity

Yes

Type of wireless communication link. Typically, this is one of the following:

• Relay
  
  
• Direct