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AudioStream Table

Microsoft Office Communications Server 2007 and Microsoft Office Communications Server 2007 R2 will reach end of support on January 9, 2018. To stay supported, you will need to upgrade. For more information, see Resources to help you upgrade your Office 2007 servers and clients.

Each record represents one audio stream. One audio media line usually contains two audio streams.

Column Data Type Key/Index Details

ConferenceDateTime

Datetime

primary

Referenced from the MediaLine Table.

SessionSeq

Int

primary

Referenced from the MediaLine Table.

MediaLineLabel

tinyint

primary

Referenced from the MediaLine Table.

StreamID

int

primary

Unique ID within a media line.

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.

AudioPayloadDescription

varchar(256)

 

The audio codec used for the call.

AudioPayloadType

int

 

Not used.

AudioSampleRate

int

 

Sampling rate for the audio stream.

InboundAudioSignalLevel *

int

 

Represents the Post-Analog Gain Control audio signal level. 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. This is reported by the Mediation Server on approximately 2% of the calls on the Mediation Server to gateway leg.

InboundAudioNoiseLevel *

int

 

Represents the Post-Analog Gain Control audio noise level. 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. This is reported by the Mediation Server on approximately 2% of the calls on the Mediation Server to gateway leg.

InboundAudioSignalEchoReturn*

int

 

Echo Return Loss Enhancement metric. 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. This is reported by the Mediation Server on approximately 2% of the calls on the Mediation Server to gateway leg.

OutboundAudioSignalLevel*

int

 

Represents the Post Analog Gain Control audio Signal level. The unit for this metric is dBmo. For acceptable quality it should be at least dBmo. This metric is not reported by the A/V Conferencing Server or IP phones. This is reported by the Mediation Server on approximately 2% of the calls on the Mediation Server to gateway leg.

OutboundAudioNoiseLevel*

int

 

Represent the Post Analog Gain control audio noise level. 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. This is reported by the Mediation Server on approximately 2% of the calls on the Mediation Server to gateway leg.

OutboundAudioSignalEchoReturn*

int

 

Echo Return Loss Enhancement metric. 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. This is reported by the Mediation Server on approximately 2% of the calls on the Mediation Server to gateway leg.

AudioSpeakerFeedbackMicIn*

int

 

This is the microphone input level from the loudspeaker signal which comes from the far end. The unit is dBoV. For acceptable quality this value should be less than 20 dBoV. If this number is too high, it means that the near end microphone is getting too much feedback from the near end loudspeaker. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioSpeechLevelMicIn*

int

 

This is the speech level into the microphone at the near end. The unit is dBoV. For acceptable quality it should be between -18 dBoV and -35 dBoV, if greater than -18 dBoV, then signal clipping or echo is occurring when both parties talk. If it is less than -35 dBoV, then speech might be distorted. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioSpeechLevelPostProcess*

int

 

Overall average speech level sent to the far end (after signal processing) from the near end. The unit for this metric is dBoV. For acceptable quality it should be within [-30 to -18] dBoV. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioSignalLevelLoudSpeaker*

int

 

Speaker/Headphone input level (at the near end). The unit for this metric is dBoV. For acceptable quality it should range between [-35 to -15] dBoV. If too high there may be clipping. If too low then there might be low volume issues. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioBackGroundNoiseMicIn*

int

 

Background Noise Input to the microphone at the near end. The unit for this metric is dBoV. For acceptable quality the range should be less than -35 dBoV. If noise is too high, this indicates a bad device or bad device setup which is degrading audio quality. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioBackGroundNoiseSent*

int

 

Background noise left over after noise suppression. This is the noise sent to the far end. The unit for this is dBoV. For acceptable call quality this should be less than -45 dBoV. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioLocalSpeechToEcho*

int

 

This the ratio of speech to echo. The unit for this is dB. For acceptable quality it should be greater than 10 dB. If less than 10dB then speech level is too low compared to echo level, and distorted speech might occur. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioSpeakerGlitchRate

int

 

Average glitches per 5 minutes for the loudspeaker rendering. For good quality, this should be less than 1 per 5 minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioMicGlitchRate

int

 

Average glitches per 5 minutes for the microphone capture. For good quality this should be less than 1 per 5 minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioSpeakerClipRate

int

 

Clipping occurrences per 5 minutes for loudspeaker rendering. For good quality, this should be less than 1 per 5 minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioMicClipRate

int

 

Clipping per 5 minutes for microphone capture. For good quality this should be less than 1 per 5 minutes. Not reported by A/V Conferencing Servers, Mediation Servers, or IP phones.

AudioRxAGCSignalLevel*

int

 

Received signal level on the Mediation Server from the gateway; 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.

AudioRxAGCNoiseLevel*

int

 

Received Received signal level on the Mediation Server from the gateway; 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.

RoundTrip

int

 

Round trip time from RTCP statistics. For acceptable quality this should be less than 100ms.

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)

 

The minimum wideband Network MOS for the call.

SendListenMOS

decimal(3,2)

 

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

SendListenMOSMin

decimal(3,2)

 

The minimum SendListenMOS for the call.

RecvListenMOS

decimal(3,2)

 

The 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)

 

The minimum RecvListenMOS for the call.

Inbound

bit

 

Stream data on receiver side is received.

Outbound

bit

 

Stream data on sender side is received.

SenderIsCallerPAI

bit

 

1 means the stream direction is from Caller to Callee.

0 means the stream direction is from Callee to Caller.

Note: * means, the metric is scaled by *-100 when stored in QoE DB. Dividing the number by -100 will give the ranges listed in this table.