AudioStreamDetail view

The AudioStreamDetail View stores information about each audio stream in the database. This view was introduced in Microsoft Lync Server 2013.

Column Data Type Details
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 Skype for Business 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.