AudioSignal Table

 

Topic Last Modified: 2011-02-03

Each record represents audio signal metrics for one endpoint. Usually, each call has two records, one is for caller, and one is for callee.

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.
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.