AvayaWebRTCConnectSDK
AOAudioDetails Class Reference

#import <AOAudioDetails.h>

Inheritance diagram for AOAudioDetails:
<AOCallDetails>

Properties

NSUInteger DTMFPayloadType
 
NSUInteger packetizationIntervalMilliseconds
 
NSUInteger packetsTransmitted
 
NSUInteger packetsReceived
 
NSUInteger bytesTransmitted
 
NSUInteger bytesReceived
 
NSUInteger fractionLostReceived
 
NSUInteger fractionLostTransmitted
 
NSUInteger averageJitterReceivedMilliseconds
 
NSUInteger averageJitterTransmittedMilliseconds
 
NSUInteger currentBufferSizeMilliseconds
 
NSUInteger preferredBufferSizeMilliseconds
 
NSUInteger currentPacketLossRate
 
NSUInteger currentDiscardRate
 
NSUInteger currentExpandRate
 
NSUInteger currentPreemptiveRate
 
NSUInteger currentAccelerationRate
 
- Properties inherited from <AOCallDetails>
NSString * localIPAddress
 
NSUInteger localPort
 
NSString * remoteIPAddress
 
NSUInteger remotePort
 
NSString * codec
 
AOMediaEncryptionType mediaEncryptionType
 
BOOL mediaTunnelled
 
BOOL mediaProxied
 
BOOL mediaEncrypted
 
NSUInteger roundTripTimeMilliseconds
 

Detailed Description

Audio-related details for a session.

Property Documentation

◆ averageJitterReceivedMilliseconds

- (NSUInteger) averageJitterReceivedMilliseconds
readnonatomicassign

The average jitter buffer size in milliseconds the local end is experiencing on the received RTP stream. In VoIP, a jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Variations in packet arrival time, called jitter, can occur because of network congestion, timing drift, or route changes. The jitter buffer, which is located at the receiving end of the voice connection, intentionally delays the arriving packets so that the end user experiences a clear connection with very little sound distortion.

◆ averageJitterTransmittedMilliseconds

- (NSUInteger) averageJitterTransmittedMilliseconds
readnonatomicassign

The average jitter buffer size in milliseconds the remote end is experiencing on the transmitted RTP stream. In VoIP, a jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Variations in packet arrival time, called jitter, can occur because of network congestion, timing drift, or route changes. The jitter buffer, which is located at the receiving end of the voice connection, intentionally delays the arriving packets so that the end user experiences a clear connection with very little sound distortion.

◆ bytesReceived

- (NSUInteger) bytesReceived
readnonatomicassign

The total number of RTP payload bytes received.

◆ bytesTransmitted

- (NSUInteger) bytesTransmitted
readnonatomicassign

The total number of RTP payload bytes transmitted.

◆ currentAccelerationRate

- (NSUInteger) currentAccelerationRate
readnonatomicassign

Fraction of data removed through acceleration. In case of a full jitter buffer speech frames will be deleted. This process is called "acceleration".

◆ currentBufferSizeMilliseconds

- (NSUInteger) currentBufferSizeMilliseconds
readnonatomicassign

The current jitter buffer size in milliseconds. In VoIP, a jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Variations in packet arrival time, called jitter, can occur because of network congestion, timing drift, or route changes. The jitter buffer, which is located at the receiving end of the voice connection, intentionally delays the arriving packets so that the end user experiences a clear connection with very little sound distortion.

◆ currentDiscardRate

- (NSUInteger) currentDiscardRate
readnonatomicassign

The percentage of packets discarded (late).

◆ currentExpandRate

- (NSUInteger) currentExpandRate
readnonatomicassign

Fraction of synthesized speech frames inserted through expansion in total frame count in buffer. In case of a starved jitter buffer synthesized speech frames will be added. This process is called "expand".

◆ currentPacketLossRate

- (NSUInteger) currentPacketLossRate
readnonatomicassign

The percentage of packets lost (network + late).

◆ currentPreemptiveRate

- (NSUInteger) currentPreemptiveRate
readnonatomicassign

Fraction of synthesized speech frames inserted through pre-emptive expansion in total frame count in buffer. In case of a shallow jitter buffer synthesized speech frames will be added. This process is called "pre-emptive expand" and based on previous frames.

◆ DTMFPayloadType

- (NSUInteger) DTMFPayloadType
readnonatomicassign

The dynamic payload type used for telephony events (DTMF tones).

◆ fractionLostReceived

- (NSUInteger) fractionLostReceived
readnonatomicassign

The fractional loss seen locally. This is 8-bits size value. The fraction of RTP data packets from source lost since the previous SR or RR packet was sent, expressed as a fixed point number with the binary point at the left edge of the field. (That is equivalent to taking the integer part after multiplying the loss fraction by 256.) This fraction is defined to be the number of packets lost divided by the number of packets expected. If the loss is negative due to duplicates, the fraction lost is set to zero.

◆ fractionLostTransmitted

- (NSUInteger) fractionLostTransmitted
readnonatomicassign

The fractional loss seen remotely. This is 8-bits size value. The fraction of RTP data packets from source lost since the previous SR or RR packet was sent, expressed as a fixed point number with the binary point at the left edge of the field. (That is equivalent to taking the integer part after multiplying the loss fraction by 256.) This fraction is defined to be the number of packets lost divided by the number of packets expected. If the loss is negative due to duplicates, the fraction lost is set to zero.

◆ packetizationIntervalMilliseconds

- (NSUInteger) packetizationIntervalMilliseconds
readnonatomicassign

The packetization interval in milliseconds. This represents the time duration of the audio data contained in each packet. It is retrieved from the ptime attribute from the SDP audio codec config.

◆ packetsReceived

- (NSUInteger) packetsReceived
readnonatomicassign

The total number of RTP packets received.

◆ packetsTransmitted

- (NSUInteger) packetsTransmitted
readnonatomicassign

The total number of RTP packets transmitted.

◆ preferredBufferSizeMilliseconds

- (NSUInteger) preferredBufferSizeMilliseconds
readnonatomicassign

The preferred (optimal) jitter buffer size in milliseconds. In VoIP, a jitter buffer is a shared data area where voice packets can be collected, stored, and sent to the voice processor in evenly spaced intervals. Variations in packet arrival time, called jitter, can occur because of network congestion, timing drift, or route changes. The jitter buffer, which is located at the receiving end of the voice connection, intentionally delays the arriving packets so that the end user experiences a clear connection with very little sound distortion.


The documentation for this class was generated from the following file: