AudioDetails Class |
Namespace: Avaya.ClientServices
public class AudioDetails
The AudioDetails type exposes the following members.
Name | Description | |
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AverageLocalJitterMilliseconds |
The average jitter buffer size (in milliseconds) the local end is experiencing on
the received RTP stream.
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AverageRemoteJitterMilliseconds |
The average jitter buffer size (in milliseconds) the remote end is experiencing on
the transmitted RTP stream.
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BytesReceived |
The total number of RTP payload bytes received.
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BytesTransmitted |
The total number of RTP payload bytes transmitted.
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Codec |
The name of the audio codec being used.
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CurrentAccelerateRate |
The fraction of data removed through acceleration.
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CurrentBufferSizeMilliseconds |
The current jitter buffer size in milliseconds.
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CurrentDiscardRate |
The percentage of packets discarded (late).
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CurrentExpandRate |
The fraction of synthesized speech frames inserted through expansion
in total frame count in buffer.
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CurrentPacketLossRate |
The percentage of packets lost (network + late).
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CurrentPreemptiveRate |
The fraction of synthesized speech frames inserted through pre-emptive
expansion in total frame count in buffer.
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DtmfPayloadType |
The dynamic payload type used for telephony events (DTMF tones).
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EncryptionType |
Media encryption type.
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IsMediaProxied |
Status indication for whether tunneled Audio goes through proxy or not.
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IsMediaTunneled |
Status indication for whether Audio tunneled or not.
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LocalFractionLost |
The fractional loss seen locally.
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LocalIPAddress |
The local IP address used for the audio session.
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LocalPort |
The local RTP receive port for the audio session (as per SDP
offer/answer exchange).
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PacketizationMilliseconds |
The packetization interval in milliseconds.
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PacketsReceived |
The total number of RTP packets received.
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PacketsTransmitted |
The total number of RTP packets transmitted.
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PreferredBufferSizeMilliseconds |
The preferred (optimal) jitter buffer size in milliseconds.
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RemoteFractionLost |
The fractional loss seen remotely which far end sends to local end through RTCP.
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RemoteIPAddress |
The remote IP address used for the audio session.
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RemotePort |
The remote RTP receive port for the audio session (as per SDP
offer/answer exchange).
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RoundTripTimeMilliseconds |
The round-trip audio delay (in milliseconds) calculated as per RFC
3550.
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RtcpEncrypted |
Indicates whether RTCP is encrypted or not.
True if feedback RTCP packets are encrypted false otherwise.
If it is true RTCP packets should be decrypted before reading
audio stream details.
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