Jabber Guest iOS SDK
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CJGuestAudioStatistics Class Reference
This class holds parameters specific to the audio stream of the call. More...
#import <JabberGuest/JabberGuest.h>
Inheritance diagram for CJGuestAudioStatistics:
Additional Inherited Members | |
Properties inherited from CJGuestMediaStatistics | |
CJGuestCodecStatistics * | receiveCodec |
Codec properties used to receive the media. | |
NSInteger | receivedPackets |
Packets received over the current media stream. | |
NSInteger | receivePacketsLost |
Packets lost over the current media stream. | |
float | receivePercentLost |
Percent of packets lost during the period of last observation. | |
float | receiveCumPercentLost |
Overall percent of packets lost over the current media stream. | |
NSInteger | receiveBitrate |
Average bitrate for stream received during the period of last observation (bits per second). | |
NSInteger | receiveJitter |
Average jitter (in milliseconds) for stream received during the period of last observation. | |
NSInteger | sentPackets |
Packets sent over the current media stream. | |
CJGuestCodecStatistics * | transmitCodec |
Codec properties used to transmit the media. | |
NSInteger | transmitPacketsReceived |
Packets received by remote endpoint(s) over the current media stream. | |
NSInteger | transmitPacketsLost |
Number of packets expected but not received by remote endpoint(s) over the current media stream. | |
float | transmitPercentLost |
Percent of packets expected but not received by remote endpoint(s) during the period covered by the most recent RTCP reception report received from the remote endpoint. | |
float | transmitCumPercentLost |
Percent of packets expected but not received by remote endpoint(s) over the current media stream. | |
NSInteger | transmitBitrate |
Average bitrate for stream sent during the period of last observation. | |
NSInteger | transmitJitter |
Jitter (in milliseconds) for stream sent during the period covered by the most recent RTCP reception report received from the remote endpoint. | |
NSInteger | transmitRoundTrip |
Round trip calculation based on most recent sent and received RTP and RTCP statistics. | |
Detailed Description
This class holds parameters specific to the audio stream of the call.