AudioStream Table
Topic Last Modified: 2010-12-14
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 |
int |
Foreign |
The audio Codec used for the call, referenced from PayloadDescription Table. |
AudioSampleRate |
int |
|
Sampling rate for the audio stream. |
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. |
AudioFECUsed |
bit |
Flag indicating if audio FEC was used for 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. |
|
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 the caller to the callee. 0 means the stream direction is from the callee to the caller. |