RTCOutboundRtpStreamStats

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RTCOutboundRtpStreamStats dictionary is the RTCStats -based object which provides metrics and statistics related to an outbound RTP stream being sent by an RTCRtpSender .

特性

RTCOutboundRtpStreamStats dictionary includes the following properties in addition to those it inherits from RTCSentRtpStreamStats , RTCRtpStreamStats ，和 RTCStats .

averageRtcpInterval: A floating-point value indicating the average RTCP interval between two consecutive compound RTCP packets.
firCount: An integer value which indicates the total number of Full Intra Request (FIR) packets which this RTCRtpSender has sent to the remote RTCRtpReceiver . This is an indicator of how often the stream has lagged, requiring frames to be skipped in order to catch up. Valid only for video streams.
framesEncoded: The number of frames that have been successfully encoded so far for sending on this RTP stream. Only valid for video streams.
lastPacketSentTimestamp: DOMHighResTimeStamp indicating the time at which the last packet was sent for this SSRC. The timestamp property, on the other hand, indicates the time at which the RTCOutboundRtpStreamStats object was generated.
nackCount: An integer value indicating the total number of Negative ACKnolwedgement (NACK) packets this RTCRtpSender has received from the remote RTCRtpReceiver .
perDscpPacketsSent: A record of key-value pairs with strings as the keys mapped to 32-bit integer values, each indicating the total number of packets this RTCRtpSender has transmitted for this source for each Differentiated Services Code Point (DSCP).
pliCount: An integer specifying the number of times the remote receiver has notified this RTCRtpSender that some amount of encoded video data for one or more frames has been lost, using Picture Loss Indication (PLI) packets. Only available for video streams.
qpSum: A 64-bit value containing the sum of the QP values for every frame encoded by this RTCRtpSender . Valid only for video streams.
qualityLimitationDurations: A record mapping each of the quality limitation reasons in the RTCRemoteInboundRtpStreamStats enumeration to a floating-point value indicating the number of seconds the stream has spent with its quality limited for that reason.
qualityLimitationReason: A value from the RTCQualityLimitationReason enumerated type explaining why the resolution and/or frame rate is being limited for this RTP stream. Valid only for video streams .
remoteId: A string which identifies the RTCRemoteInboundRtpStreamStats object that provides statistics for the remote peer for this same SSRC. This ID is stable across multiple calls to getStats() .
retransmittedBytesSent: The total number of bytes that have been retransmitted for this source as of the time the statistics were sampled. These retransmitted bytes comprise the packets included in the value returned by retransmittedPacketsSent .
retransmittedPacketsSent: The total number of packets that have needed to be retransmitted for this source as of the time the statistics were sampled. These retransmitted packets are included in the value returned by packetsSent .
senderId: The {domxref("RTCStats.id", "id")}} of the RTCAudioSenderStats or RTCVideoSenderStats object containing statistics about this stream's RTCRtpSender .
sliCount
targetBitrate: A value indicating the bit rate the RTCRtpSender 's codec is configured to attempt to achieve in its output media.
totalEncodedBytesTarget: A cumulative sum of the target frame sizes (the targeted maximum size of the frame in bytes when the codec is asked to compress it) for all of the frames encoded so far. This will likely differ from the total of the actual frame sizes.
totalEncodeTime: A floating-point value indicating the total number of seconds that have been spent encoding the frames encoded so far by this RTCRtpSender .
trackId: id 的 RTCSenderAudioTrackAttachmentStats or RTCSenderVideoTrackAttachmentStats object containing the current track attachment to the RTCRtpSender responsible for this stream.

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规范	状态	注释
WebRTC 1.0: Real-time Communication Between Browsers The definition of 'RTCOutboundRtpStreamStats' in that specification.	候选推荐	最初的规范。

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The compatibility table on this page is generated from structured data. If you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request.

No compatibility data found. Please contribute data for "api.RTCOutboundRtpStreamStats" (depth: 1) to the MDN 兼容性数据存储库 .

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元数据

最后修改： Apr 11, 2019

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