SessionTime

datetime

Referenced from the MediaLine Table.

SessionSeq

int

Referenced from the MediaLine Table.

StreamId

int

Unique ID within a media line.

StartTime

datetime

Start time of the session.

EndTime

datetime

End time of the session.

DialogCategory

bit

Dialog category: 0 is the Lync Server to Mediation Server leg; 1 is the Mediation Server to PSTN gateway leg.

MediationServerBypassFlag

bit

Flag indicating if the call was bypassed or not.

MediaBypassWarningFlag

int

If present, indicates why a call was not bypassed even if the bypass IDs matched. Only one value is defined:

0x0001 – Unknown bypass ID for Default network adapter.

CallPriority

int

Priority of the call.

CallerPool

nvarchar(256)

Caller pool FQDN.

CalleePool

nvarchar(256)

Callee pool FQDN.

Caller

nvarchar(450)

Caller’s URI.

Callee

nvarchar(450)

Callee’s URI.

CallerUserAgent

nvarchar(256)

Caller’s user agent string.

CallerUserAgentType

smallint

Type of the caller’s user agent. See the UserAgent Table for details.

CallerUserAgentCategory

nvarchar(64)

Category of the caller’s user agent. See the UserAgentDef Table (QoE) for details.

CalleeUserAgent

nvarchar(256)

Callee’s user agent string.

CalleeUserAgentType

smallint

Type of callee’s user agent. See the UserAgent Table for information.

CalleeUserAgentCategory

nvarchar(64)

Category of callee’s user agent. See the UserAgentDef Table (QoE) for information.

CallerEndpoint

nvarchar(256)

Caller’s endpoint name.

CalleeEndpoint

nvarchar(256)

Callee’s endpoint name.

CallerOS

nvarchar(128)

Operating system (OS) of the caller’s endpoint.

CalleeOS

nvarchar(128)

Operating system (OS) of the callee’s endpoint.

CallerCPUName

nvarchar(128)

CPU name of the caller’s endpoint.

CalleeCPUName

nvarchar(128)

CPU name of the callee’s endpoint.

CallerCPUNumberOfCores

smallint

Number of CPU cores in the caller’s endpoint.

CalleeCPUNumberOfCores

smallint

Number of CPU cores in the callee’s endpoint.

CallerCPUProcessorSpeed

int

CPU processor speed of the caller’s endpoint.

CalleeCPUProcessorSpeed

int

CPU processor speed of the callee’s endpoint.

CallerVirtualizationFlag

tinyint

Indicates whether the caller’s system is running in a virtualized environment. See the Endpoint Table for more information.

CalleeVirtualizationFlag

tinyint

Indicates whether the callee’s system is running in a virtualized environment. See the Endpoint Table for more information.

CorrelationKey

Correlation key. Referenced from the SessionCorrelation Table.

ConnectivityIce

tinyint

Information about the media path, such as direct or relayed. See the MediaLine Table for more information.

CallerIceWarningFlags

int

Information about Interactive Connectivity Establishment (ICE) process described in bits flags for the caller. For details, refer to the Quality of Experience Monitoring Server Protocol Specification.

CalleeIceWarningFlags

int

Information about Interactive Connectivity Establishment (ICE) process described in bits flags for the callee. For details, refer to the Quality of Experience Monitoring Server Protocol Specification.

Transport

tinyint

Transport type: 0 is UDP, 1 is TCP.

CallerIPAddr

var(50)

IP address of the caller. This may be either an IPv4 or an IPv6 address.

CallerPort

int

Port used by the caller.

CallerInside

bit

Indicates whether the caller is inside the interval network: 1 means caller is inside the enterprise network, 0 means the caller is outside the network.

CalleeIPAddr

var(50)

IP address of the callee. This may be either an IPv4 or an IPv6 address.

CalleePort

int

Port used by the callee.

CalleeInside

bit

Indicates whether the callee is inside the interval network: 1 means callee is inside the enterprise network, 0 means the callee is outside the network.

CallerUserSite

nvarchar(128)

Name of the caller’s site.

CallerRegion

nvarchar(128)

Name of the country/region of the caller’s site.

CalleeUserSite

nvarchar(128)

Name of the callee’s site.

CalleeRegion

nvarchar(128)

Name of the country/region of the callee’s site.

CallerRelayIPAddr

var(50)

IP Address of the A/V Edge service used by the caller. See the IPAddress Table for more information.

CallerRelayPort

int

Port used on the A/V Edge service used by the caller.

CalleeRelayIPAddr

var(50)

IP Address key of the A/V Edge service used by the callee. See the IPAddress Table for more information.

CalleeRelayPort

int

Port used on the A/V Edge service used by the callee.

CallerCaptureDev

varchar(256)

Caller’s capture device name.

CallerRenderDev

varchar(256)

Caller’s render device name.

CallerCaptureDevDriver

varchar(256)

Caller’s capture device driver name.

CallerRenderDriver

varchar(256)

Caller’s render device driver name.

CalleeCaptureDev

varchar(256)

Callee’s capture device name.

CalleeRenderDev

varchar(256)

Callee’s render device name.

CalleeCaptureDevDriver

varchar(256)

Callee’s capture device driver name.

CalleeRenderDevDriver

varchar(256)

Callee’s render device driver name.

CallerNetworkConnectionType

tinyint

Caller’s network connection type: 0 is wired, 1 is wireless.

CallerVPN

bit

Indicates whether the caller connected over a virtual private network: 1 is virtual private network (VPN), 0 is non-VPN.

CallerLinkSpeed

decimal(18,0)

Network link speed for the caller's endpoint in bps.

CalleeNetworkConnectionType

tinyint

Callee’s network connection type: 0 is wired, 1 is wireless.

CalleeVPN

bit

Indicates whether the caller connected over a virtual private network: 1 is virtual private network (VPN), 0 is non-VPN.

CalleeLinkSpeed

decimal(18,0)

Network link speed for the callee's endpoint in bps.

ConversationalMOS

decimal(3,2)

Narrowband Conversational MOS of the audio sessions (based on both audio streams).

AppliedBandwidthLimit

int

Actual bandwidth applied to the given send side stream given various policy settings (TURN, API, SDP, Policy Server, and so on). This is not to be confused with the effective bandwidth because there can be a lower effective bandwidth based on the bandwidth estimate. This is basically the maximum bandwidth the send stream can take barring limits imposed by the bandwidth estimate

JitterInterArrival

int

Average network jitter from Real Time Control Protocol (RTCP) statistics.

JitterInterArrivalMax

int

Maximum network jitter during the call.

PacketLossRate

decimal(5,4)

Average packet loss rate during the call.

PacketLossRateMax

decimal(5,4)

Maximum packet loss observed during the call.

BurstDensity

decimal(9,4)

Average density of packet loss during bursts of losses during the call.

BurstDuration

int

Average duration of packet loss during bursts of losses during the call.

BurstGapDensity

decimal(9,4)

Average density of packet loss during gaps between bursts of packet loss.

BurstGapDuration

int

Average duration of gaps between bursts of packet loss.

PacketUtilization

int

Packet count for the audio stream.

BandwidthEst

int

Bandwidth estimates for the audio stream.

DegradationAvg

decimal(3,2)

Network MOS Degradation for the whole call. Range is 0.0 to 5.0. This metric shows the amount the Network MOS was reduced because of jitter and packet loss. For acceptable quality it should less than 0.5.

DegradationMax

decimal(3,2)

Maximum Network MOS degradation during the call.

DegradationJitterAvg

decimal(3,2)

Network MOS degradation caused by jitter.

DegradationPacketLossAvg

decimal(3,2)

Network MOS degradation caused by packet loss.

PayloadDescription

int

The audio codec used for the call, referenced from the PayloadDescription Table.

AudioSampleRate

int

Sampling rate for the audio stream.

CallerSendSignalLevel

int

Post-Analog Gain Control audio signal level for the audio the caller sent. The unit for this metric is dBmo. For acceptable quality, it should be at least 30 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CallerRecvSignalLevel

int

Audio signal level for the audio the caller received. The unit for this metric is dBmo. For acceptable quality, it should be at least 30 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CallerSendNoiseLevel

int

Post-Analog Gain Control audio noise level for the audio the caller sent. The unit for this metric is dBmo. For acceptable quality, it should be less than 35 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CallerRecvNoiseLevel

int

Post-Analog Gain Control audio noise level for the audio the caller received. The unit for this metric is dBmo. For acceptable quality, it should be less than 35 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CallerEchoReturn

int

Echo Return Loss Enhancement for the caller. The unit for this metric is dB. Lower values represent less echo. This metric is not reported by the A/V Conferencing Server or IP phones.

CallerSpeakerGlitchRate

int

Average glitches per five minutes for the caller’s loudspeaker rendering. For good quality, this should be less than one per five minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

CallerMicGlitchRate

int

Average glitches per five minutes for the caller’s microphone capture. For good quality this should be less than one per five minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

CallerTimestampDriftRateMic

decimal(9,2)

Caller’s microphone device clock drift rate, relative to CPU clock.

CallerTimestampDriftRateSpk

decimal(9,2)

Caller’s speaker device clock drift rate, relative to CPU clock.

CallerTimestampErrorMicMs

decimal(9,2)

Average microphone capture stream time stamp error, in milliseconds, in the last 20 seconds of the call.

CallerTimestampErrorSpkMs

decimal(9,2)

Average of the caller’s speaker render stream time stamp error, in milliseconds, in the last 20 seconds of the call.

CallerVsEntryCauses

smallint

Voice switch is a half-duplex mode with reduced interruption ability. See the MediaLine Table for more information.

CallerEchoEventCauses

tinyint

Causes of an echo event for the caller. See the MediaLine Table for more information.

CallerEchoPercentMicIn

decimal(5,2)

Percentage of time when echo is detected in the caller’s microphone capture stream. If headset is used, the value should be low.

CallerEchoPercentSend

decimal(5,2)

Percentage of time when echo is detected in the caller’s sent stream. High echo percentage in send streams an indication of echo leak.

CallerRxAGCSignalLevel

int

Received signal level on the Mediation Server from the Gateway for the caller’s audio; this applies only to the Mediation Server. The unit of this metric is dBoV. For good quality, the acceptable range should be -30 to -18 dBoV.

CallerRxAGCNoiseLevel

int

Received signal level on the Mediation Server from the Gateway for the caller’s audio. This applies only to the Mediation Server. The unit of this metric is dBoV. For good quality, the acceptable range should be less than -50 dBoV.

CallerRxAGCGain

int

Automatic gain control (AGC) on the Mediation Server side applied to the caller’s audio.

CallerInitialSignalLevelRMS

float

Root mean square (RMS) of the incoming signal to the caller for up to the first 30 seconds of the call.

CalleeSendSignalLevel

int

Represents the Post-Analog Gain Control audio signal level for the audio the callee sent. The unit for this metric is dBmo. For acceptable quality, it should be at least 30 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CalleeRecvSignalLevel

int

Audio signal level for the audio the callee received. The unit for this metric is dBmo. For acceptable quality, it should be at least 30 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CalleeSendNoiseLevel

int

Post-Analog Gain Control audio noise level for the audio the callee sent. The unit for this metric is dBmo. For acceptable quality, it should be less than 35 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CalleeRecvNoiseLevel

int

Post-Analog Gain Control audio noise level for the audio the callee received. The unit for this metric is dBmo. For acceptable quality, it should be less than 35 dBmo. This metric is not reported by the A/V Conferencing Server or IP phones.

CalleeEchoReturn

int

Echo Return Loss Enhancement for the callee. The unit for this metric is dB. Lower values represent less echo. This metric is not reported by the A/V Conferencing Server or IP phones.

CalleeSpeakerGlitchRate

int

Average glitches per five minutes for the callee’s loudspeaker rendering. For good quality, this should be less than one per five minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

CalleeMicGlitchRate

int

Average glitches per five minutes for the callee’s microphone capture. For good quality this should be less than one per five minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

CalleeTimestampDriftRateMic

decimal(9,2)

Callee’s microphone device clock drift rate, relative to CPU clock.

CalleeTimestampDriftRateSpk

decimal(9,2)

Callee’s speaker device clock drift rate, relative to CPU clock.

CalleeTimestampErrorMicMs

decimal(9,2)

Average microphone capture stream time stamp error, in milliseconds, in the last 20 seconds of the call.

CalleeTimestampErrorSpkMs

decimal(9,2)

Average of the callee’s speaker render stream time stamp error, in milliseconds, in the last 20 seconds of the call.

CalleeVsEntryCauses

smallint

Voice switch is a half-duplex mode with reduced interruption ability. See the MediaLine Table for more information.

CalleeEchoEventCauses

tinyint

Causes of an echo event for the callee. See the MediaLine Table for more information.

CalleeEchoPercentMicIn

decimal(5,2)

Percentage of time when echo is detected in the callee’s microphone capture stream. If headset is used, the value should be low.

CalleeEchoPercentSend

decimal(5,2)

Percentage of time when echo is detected in the callee’s sent stream. High echo percentage in send streams an indication of echo leak.

CalleeRxAGCSignalLevel

int

Received signal level on the Mediation Server from the Gateway for the callee’s audio; this applies only to the Mediation Server. The unit of this metric is dBoV. For good quality, the acceptable range should be [-30 to -18] dBoV.

CalleeRxAGCNoiseLevel

int

Received signal level on the Mediation Server from the Gateway for the callee’s audio. This applies only to the Mediation Server. The unit of this metric is dBoV. For good quality, the acceptable range should be less than -50 dBoV.

CalleeRxAGCGain

int

Automatic gain control (AGC) on the Mediation Server side applied to the callee’s audio.

CalleeInitialSignalLevelRMS

float

Root mean square (RMS) of the incoming signal to the callee for up to the first 30 seconds of the call.

RatioConcealedSamplesAvg

decimal(5,2)

Average ratio of concealed samples generated by audio healing to typical samples.

RatioStretchedSamplesAvg

decimal(5,2)

Average ratio of stretched samples generated by audio healing to typical samples.

RatioCompressedSamplesAvg

decimal(5,2)

Average ratio of compressed samples generated by audio healing to typical samples.

RoundTrip

int

Round trip time from RTCP statistics.

RoundTripMax

int

Maximum round trip time for the audio stream.

OverallAvgNetworkMOS

decimal(3,2)

Average wideband Network MOS for the call. This metric depends on the packet loss, jitter, and codec used. Range is 1.0 to 5.0.

OverallMinNetworkMOS

decimal(3,2)

Minimum wideband Network MOS for the call.

SendListenMOS

decimal(3,2)

Average predicted wideband Listening MOS score for audio sent, including speech level, noise level and capture device characteristics.

SendListenMOSMin

decimal(3,2)

Minimum SendListenMOS for the call.

RecvListenMOS

decimal(3,2)

Average predicted wideband Listening MOS score for audio received from the network including speech level, noise level, codec, network conditions and capture device characteristics.

RecvListenMOSMin

decimal(3,2)

Minimum RecvListenMOS for the call.

AudioFECUsed

bit

Indicates whether audio FEC was used for the call.

SenderIsCallerPAI

bit

Indicates direction of the p-asserted identify information; 1 means the stream direction is from the caller to the callee; 0 means the stream direction is from the callee to the caller.