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• 特性
• 规范
• 浏览器兼容性
• 相关话题

RTCRtpStreamStats dictionary is returned by the RTCPeerConnection.getStats() , RTCRtpSender.getStats() ，和 RTCRtpReceiver.getStats() methods to provide detailed statistics about WebRTC connectivity. While the dictionary has a base set of properties that are present in each of these cases, there are also additional properties added depending on which interface the method is called on.

RTCRtpStreamStats is the base class for all RTP-related statistics reports. It's based on RTCStats and adds the following additional fields.

注意： This interface was called RTCRTPStreamStats until a specification update in the spring of 2017. Check the 浏览器兼容性 table to know if and when the name change was implemented in specific browsers.

特性

RTCRtpStreamStats dictionary is based on RTCStats , and inherits its properties. In addition, some or all of the following properties are available.

Standard fields included for all media types

codecId

DOMString which uniquely identifies the object which was inspected to produce the RTCCodecStats object associated with this RTP 流。

kind

DOMString whose value is "audio" if the associated MediaStreamTrack is audio-only or "video" if the track contains video. This value will match that of the media type indicated by RTCCodecStats.codec , as well as the track's kind property. Previously called mediaType .

ssrc

The 32-bit integer which identifies the source of the RTP packets this RTCRtpStreamStats object covers. This value is generated per the RFC 3550 规范。

trackId

DOMString which uniquely identifies the RTCMediaStreamTrackStats object representing the associated MediaStreamTrack . This is not the same as the value of MediaStreamTrack.id .

transportId

DOMString uniquely identifying the object which was inspected to produce the RTCTransportStats object associated with this RTP stream.

Obsolete fields

mediaType

Renamed to kind in the specification in February 2018. See 浏览器兼容性 in RTCRtpStreamStats.kind to determine when browsers made the transition.

Local-only measurements

These properties are computed locally, and are only available to the device receiving the media stream. Their primary purpose is to examine the error resiliency of the connection, as they provide information about lost packets, lost frames, and how heavily compressed the data is.

firCount

A count of the total number of Full Intra Request (FIR) packets received by the sender. This statistic is only available to the device which is receiving the stream and is only available for video tracks. A FIR packet is sent by the receiving end of the stream when it falls behind or has lost packets and is unable to continue decoding the stream; the sending end of the stream receives the FIR packet and responds by sending a full frame instead of a delta frame, thereby letting the receiver "catch up." The higher this number is, the more often a problem of this nature arose, which can be a sign of network congestion or an overburdened receiving device.

nackCount

The number of times the receiver notified the sender that one or more RTP packets has been lost by sending a Negative ACKnowledgement (NACK, also called "Generic NACK") packet to the sender. This value is only available to the receiver.

pliCount

The number of times the receiving end of the stream sent a Picture Loss Indiciation (PLI) packet to the sender, indicating that it has lost some amount of encoded video data for one or more frames. Only the receiver has this value, and it's only valid for video tracks.

qpSum

The sum of the Quantization Parameter (QP) values associated with every frame received to date on the video track described by this RTCRtpStreamStats object. In general, the higher this number is, the more heavily compressed the video track was. Combined with RTCReceivedRtpStreamStats.framesDecoded or RTCSentRtpStreamStats.framesEncoded , you can approximate the average QP over those frames, keeping in mind that codecs often vary the quantizer values even within frames. Also keep in mind that the values of QP can vary from codec to codec, so this value is only potentially useful when compared against the same codec.

sliCount

The number of times the receiver notified the sender that one or more consecutive (in scan order) encoded video macroblocks have been lost or corrupted; this notification is sent by the receiver to the sender using a Slice Loss Indication (SLI) packet. This is a fairly technical part of how codecs work and while the higher this value is, the more errors occurred in the stream, generally most of the time this value is only interesting to very intensively hardcore media developers.

规范

规范	状态	注释
Identifiers for WebRTC's Statistics API The definition of 'RTCRtpStreamStats' in that specification.	候选推荐	初始定义。

浏览器兼容性

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. 更新 GitHub 上的兼容性数据

	桌面						移动
	Chrome	Edge	Firefox	Internet Explorer	Opera	Safari	Android webview	Chrome for Android	Firefox for Android	Opera for Android	Safari on iOS	Samsung Internet
RTCRtpStreamStats	Chrome No	Edge No	Firefox 63 63 27 Alternate Name Alternate Name Uses the non-standard name: RTCRTPStreamStats	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 63 63 27 Alternate Name Alternate Name Uses the non-standard name: RTCRTPStreamStats	Opera Android No	Safari iOS ?	Samsung Internet Android No
codecId	Chrome No	Edge No	Firefox 27	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 27	Opera Android No	Safari iOS ?	Samsung Internet Android No
firCount	Chrome No	Edge No	Firefox 55	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 55	Opera Android No	Safari iOS ?	Samsung Internet Android No
isRemote 弃用	Chrome No	Edge No	Firefox 27 27 In Firefox 28, a default value of false was added for isRemote , to match a specification update.	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 27 27 In Firefox 28, a default value of false was added for isRemote , to match a specification update.	Opera Android No	Safari iOS ?	Samsung Internet Android No
kind	Chrome No	Edge No	Firefox 63	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 63	Opera Android No	Safari iOS ?	Samsung Internet Android No
mediaTrackId 弃用	Chrome No	Edge No	Firefox 27	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 27	Opera Android No	Safari iOS ?	Samsung Internet Android No
mediaType 弃用	Chrome No	Edge No	Firefox 34	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 34	Opera Android No	Safari iOS ?	Samsung Internet Android No
nackCount	Chrome No	Edge No	Firefox 55	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 55	Opera Android No	Safari iOS ?	Samsung Internet Android No
pliCount	Chrome No	Edge No	Firefox 55	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 55	Opera Android No	Safari iOS ?	Samsung Internet Android No
qpSum	Chrome No	Edge No	Firefox 66	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 66	Opera Android No	Safari iOS ?	Samsung Internet Android No
remoteId 弃用	Chrome No	Edge No	Firefox 27	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 27	Opera Android No	Safari iOS ?	Samsung Internet Android No
sliCount	Chrome No	Edge No	Firefox No	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android No	Opera Android No	Safari iOS ?	Samsung Internet Android No
ssrc	Chrome No	Edge No	Firefox 27	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 27	Opera Android No	Safari iOS ?	Samsung Internet Android No
transportId	Chrome No	Edge No	Firefox 27	IE No	Opera No	Safari ?	WebView Android No	Chrome Android No	Firefox Android 27	Opera Android No	Safari iOS ?	Samsung Internet Android No

图例

完整支持

完整支持

不支持

不支持

兼容性未知 ?

兼容性未知

弃用。不要用于新网站。

弃用。不要用于新网站。

见实现注意事项。

使用非标名称。

元数据

• 最后修改： Mar 18, 2019

相关话题

1. WebRTC API
2. RTCRtpStreamStats
3. 特性

   1. codecId
   2. firCount
   3. kind
   4. nackCount
   5. pliCount
   6. qpSum
   7. sliCount
   8. ssrc
   9. trackId
   10. transportId
4. Related pages for WebRTC

   1. MediaDevices.getUserMedia()
   2. Navigator.mediaDevices
   3. RTCCertificate
   4. RTCDTMFSender
   5. RTCDTMFToneChangeEvent
   6. RTCDataChannel
   7. RTCDataChannelEvent
   8. RTCDtlsTransport
   9. RTCErrorEvent
   10. RTCIceCandidate
   11. RTCIceTransport
   12. RTCPeerConnection
   13. RTCPeerConnectionIceErrorEvent
   14. RTCPeerConnectionIceEvent
   15. RTCRtpReceiver
   16. RTCRtpSender
   17. RTCRtpTransceiver
   18. RTCSctpTransport
   19. RTCSessionDescription
   20. RTCStatsEvent
   21. RTCStatsReport
   22. RTCTrackEvent