ConferenceDateTime

Datetime

primary

Referenced from the MediaLine Table.

SessionSeq

Int

primary

Referenced from the MediaLine Table.

MediaLineLabel

tinyint

primary

Referenced from the MediaLine Table.

FromCaller

bit

primary

0: Callee’s data

1: Caller’s data

SendSignalLevel

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.

RecvSignalLevel

Int

See SendSignalLevel.

SendNoiseLevel

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.

RecvNoiseLevel

Int

See SendNoiseLevel.

EchoReturn

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.

AudioSpeakerGlitchRate

Int

Average glitches per five minutes for the 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.

AudioMicGlitchRate

Int

Average glitches per five minutes for the 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.

AudioTimestampDriftRateMic

Decimal(9,2)

Microphone device clock drift rate, relative to CPU clock.

AudioTimestampDriftRateSpk

Decimal(9,2)

Speaker device clock drift rate, relative to CPU clock.

AudioTimestampErrorMicMs

Decimal(9,2)

Speaker device clock drift rate, relative to CPU clock.

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

AudioTimestampErrorSpkMs

Decimal(9,2)

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

VsEntryCauses

tinyint

Voice switch is a half-duplex mode with reduced interruption ability. Causes of voice switch entry:

ENTER_VS_BADTS 0x01

ENTER_VS_ECHO 0x02

ENTER_VS_FORCEORCONVERGENCE 0x04

ENTER_VS_DNLP 0x08

The cause can be a combination of those individual causes. ENTER_VS_FORCEORCONVERGENCE can only be enabled by regkey for test purpose.

EchoEventCauses

tinyint

Causes of an echo event:

ECHO_EVENT_BAD_TIMESTAMP 0x01

ECHO_EVENT_POSTAEC_ECHO 0x02

ECHO_EVENT_ANLP 0x04

ECHO_EVENT_DNLP 0x08

ECHO_EVENT_MIC_CLIPPING 0x10

ECHO_EVENT_BAD_STATE 0x20

The cause can be a combination of those individual causes.

EchoPercentMicIn

Decimal(5,2)

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

EchoPercentSend

Decimal(5,2)

Foreign

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

RxAGCSignalLevel

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.

RxAGCNoiseLevel

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 less than -50 dBoV.

RxAvgAGCGain

int

Automatic gain control (AGC) on the Mediation Server side.

InitialSignalLevelRMS

float

The RMS of the incoming of up to the first 30 seconds of the call.