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Ff142 RTCInboundRtpStreamStats fixes (#40495)
* RTCInboundRtpStreamStats - update to only supported members (placholders) * RTCInboundRtpStreamStats.keyFramesDecoded - add * RTCInboundRtpStreamStats.headerBytesReceived - add * RTCInboundRtpStreamStats.frameHeight/.frameWidth - add * RTCInboundRtpStreamStats.framesPerSecond - add * RTCInboundRtpStreamStats.framesReceived - add * RTCInboundRtpStreamStats.audioLevel - add * RTCInboundRtpStreamStats.totalAudioEnergy * RTCInboundRtpStreamStats.concealedSamples * RTCInboundRtpStreamStats.concealmentEvents * RTCInboundRtpStreamStats.estimatedPlayoutTimestamp * RTCInboundRtpStreamStats.silentConcealedSamples * RTCInboundRtpStreamStats.framesAssembledFromMultiplePackets add * RTCInboundRtpStreamStats.totalAssemblyTime add * RTCInboundRtpStreamStats.freezeCount add * RTCInboundRtpStreamStats.insertedSamplesForDeceleration add * RTCInboundRtpStreamStats.jitterBufferDelay add * Improve inserted/removedSamples * Jitter related stats * RTCInboundRtpStreamStats.packetsDiscarded add * RTCInboundRtpStreamStats.packetsLost add * RTCInboundRtpStreamStats.packetsReceived add * RTCInboundRtpStreamStats.pauseCount add * RTCInboundRtpStreamStats.totalPausesDuration add * RTCInboundRtpStreamStats.playoutId add * RTCInboundRtpStreamStats.totalDecodeTime add * RTCInboundRtpStreamStats tidy a few * Update files/en-us/web/api/rtcinboundrtpstreamstats/playoutid/index.md Co-authored-by: github-actions[bot] <41898282+github-actions[bot]@users.noreply.github.com> * totalFreezesDuration - add * totalInterFrameDelay - add * totalProcessingDelay - add * totalSamplesDuration - add * totalSamplesReceived - add * totalSquaredInterFrameDelay - add * trackIdentifier - add * accel/decel fixes * RTCInboundRtpStreamStats.averageRtcpInterval - delete/redirect RTCInboundRtpStreamStats/receiverId - delete/redirect RTCInboundRtpStreamStats.packetsFailedDecryption - delete/redirect RTCInboundRtpStreamStats/packetsDuplicated - delete/redirect RTCInboundRtpStreamStats/perDscpPacketsReceived - delete/redirect RTCInboundRtpStreamStats.sliCount - delete/redirect * Try revert screwed up _wikihistory.json * playoutid fix * Apply suggestions from code review Co-authored-by: wbamberg <[email protected]> --------- Co-authored-by: github-actions[bot] <41898282+github-actions[bot]@users.noreply.github.com> Co-authored-by: wbamberg <[email protected]>
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‎files/en-us/_redirects.txt‎

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/en-US/docs/Web/API/RTCIceTransport/onselectedcandidatepairchange /en-US/docs/Web/API/RTCIceTransport/selectedcandidatepairchange_event
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/en-US/docs/Web/API/RTCIceTransport/onstatechange /en-US/docs/Web/API/RTCIceTransport/statechange_event
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/en-US/docs/Web/API/RTCIceTransportState /en-US/docs/Web/API/RTCIceTransport/state
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/averageRtcpInterval /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/firCount /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/lastPacketReceivedTimesta /en-US/docs/Web/API/RTCInboundRtpStreamStats/lastPacketReceivedTimestamp
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/packetsDuplicated /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/packetsFailedDecryption /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/perDscpPacketsReceived /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/pliCount /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/receiverId /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/sliCount /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCInboundRtpStreamStats/trackId /en-US/docs/Web/API/RTCInboundRtpStreamStats
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/en-US/docs/Web/API/RTCOfferAnswerOptions /en-US/docs/Web/API/RTCPeerConnection/createAnswer
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/en-US/docs/Web/API/RTCOfferAnswerOptions/voiceActivityDetection /en-US/docs/Web/API/RTCPeerConnection/createAnswer

‎files/en-us/glossary/jitter/index.md‎

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**Jitter** is the term used in computer networking to refer to [packet delay variation](https://en.wikipedia.org/wiki/Packet_delay_variation) — the variance in the rate at which packets arrive at a destination (the average of the squared difference from the mean packet arrival rate).
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"High jitter" indicates that packets are arriving at significantly varying rates, possibly due to network congestion, packet loss, and routing of packets in a stream through different paths.
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High jitter can significantly degrade the performance of real-time web applications, including voice/video streaming and online gaming.
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## Jitter buffer
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Network protocols and applications, such as WebRTC, use a "jitter buffer" to mitigate the effects of packet delay variation.
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Jitter buffers temporarily store incoming packets and then release them at a steady, synchronized rate.
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They may also use forward error correction to fix corrupted packets, re-request missing packets, [insert audio samples to conceal missing packets](https://en.wikipedia.org/wiki/Packet_loss_concealment), slow down or speed up sample playout, and so on.
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This process smooths out the arrival times of packets and ensures a more consistent playout of real-time audio and video.
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## See also
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- [How WebRTC's NetEQ Jitter Buffer Provides Smooth Audio](https://webrtchacks.com/how-webrtcs-neteq-jitter-buffer-provides-smooth-audio/#post-4560-_mv3ivinthkf5) (webrtchacks.com, June 2025)

‎files/en-us/web/api/rtcinboundrtpstreamstats/audiolevel/index.md‎

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---
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title: "RTCInboundRtpStreamStats: audioLevel property"
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short-title: audioLevel
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slug: Web/API/RTCInboundRtpStreamStats/audioLevel
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page-type: web-api-instance-property
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browser-compat: api.RTCStatsReport.type_inbound-rtp.audioLevel
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---
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{{APIRef("WebRTC")}}
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The **`audioLevel`** property of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates the audio level of the received (remote) track.
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The `audioLevel` is averaged over some small interval, using the algorithm described under {{domxref("RTCInboundRtpStreamStats.totalAudioEnergy","totalAudioEnergy")}}.
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The interval used is implementation dependent.
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> [!NOTE]
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> The value is undefined for video streams.
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## Value
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A real number.
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The value is between 0..1 (linear), where 1.0 represents 0 dBov ([decibels relative to full scale (DBFS)](https://en.wikipedia.org/wiki/DBFS)), 0 represents silence, and 0.5 represents approximately 6 dB SPL change in the [sound pressure level](https://en.wikipedia.org/wiki/Sound_pressure#Sound_pressure_level) from 0 dBov.
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## Specifications
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{{Specifications}}
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## Browser compatibility
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{{Compat}}
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## See also
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- {{domxref("RTCAudioSourceStats.audioLevel")}} for audio levels of local tracks (that are being sent)

‎files/en-us/web/api/rtcinboundrtpstreamstats/averagertcpinterval/index.md‎

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‎files/en-us/web/api/rtcinboundrtpstreamstats/bytesreceived/index.md‎

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{{APIRef("WebRTC")}}
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The {{domxref("RTCInboundRtpStreamStats")}} dictionary's
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**`bytesReceived`** property is an integer value which
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indicates the total number of bytes received so far from this synchronization source
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(SSRC).
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The **`bytesReceived`** property of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates the total number of bytes received so far from this synchronization source (SSRC), not including header and padding bytes.
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## Value
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An unsigned integer value indicating the total number of bytes received so far on this
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RTP stream, not including header and padding bytes. This value can be used to calculate
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an approximation of the average media data rate:
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The value can be used to calculate an approximation of the average media data rate:
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```js
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avgDataRate = rtcInboundRtpStreamStats.bytesReceived / elapsedTime;
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```
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This value gets reset to zero if the sender's SSRC identifier changes for any reason.
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The property value is reset to zero if the sender's SSRC identifier changes for any reason.
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## Value
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An positive integer.
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## Specifications
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‎files/en-us/web/api/rtcinboundrtpstreamstats/concealedsamples/index.md‎

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---
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title: "RTCInboundRtpStreamStats: concealedSamples property"
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short-title: concealedSamples
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slug: Web/API/RTCInboundRtpStreamStats/concealedSamples
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page-type: web-api-instance-property
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browser-compat: api.RTCStatsReport.type_inbound-rtp.concealedSamples
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---
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{{APIRef("WebRTC")}}
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The **`concealedSamples`** property of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates the total number of concealed samples for the received audio track over the lifetime of this stats object.
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A concealed sample is a sample that was lost or arrived too late to be played out, and therefore had to be replaced with a locally generated synthesized sample.
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Note that lost samples are reported in {{domxref("RTCInboundRtpStreamStats.packetsLost", "packetsLost")}}, while late packets are reported in {{domxref("RTCInboundRtpStreamStats.fecPacketsDiscarded", "fecPacketsDiscarded")}}.
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> [!NOTE]
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> The value is undefined for video streams.
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## Value
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A positive integer.
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## Specifications
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{{Specifications}}
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## Browser compatibility
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{{Compat}}
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## See also
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- {{domxref("RTCInboundRtpStreamStats.silentConcealedSamples")}}
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- {{domxref("RTCInboundRtpStreamStats.concealmentEvents")}}
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- [Packet loss concealment](https://en.wikipedia.org/wiki/Packet_loss_concealment) on Wikipedia

‎files/en-us/web/api/rtcinboundrtpstreamstats/concealmentevents/index.md‎

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---
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title: "RTCInboundRtpStreamStats: concealmentEvents property"
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short-title: concealmentEvents
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slug: Web/API/RTCInboundRtpStreamStats/concealmentEvents
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page-type: web-api-instance-property
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browser-compat: api.RTCStatsReport.type_inbound-rtp.concealmentEvents
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---
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{{APIRef("WebRTC")}}
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The **`concealmentEvents`** property of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates the total number of concealment events for the received audio track over the lifetime of this stats object.
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A concealed sample is a sample that was lost or arrived too late to be played out, and therefore had to be replaced with a locally generated synthesized sample.
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Any number of consecutive concealed samples following a non-concealed sample comprise a single concealment event.
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The value in this property will therefore be less than or equal to {{domxref("RTCInboundRtpStreamStats.concealedSamples", "concealedSamples")}}, which counts every sample.
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> [!NOTE]
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> The value is undefined for video streams.
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## Value
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A positive integer.
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## Specifications
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{{Specifications}}
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## Browser compatibility
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{{Compat}}
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## See also
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- {{domxref("RTCInboundRtpStreamStats.concealedSamples")}}
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- {{domxref("RTCInboundRtpStreamStats.silentConcealedSamples")}}
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- [Packet loss concealment](https://en.wikipedia.org/wiki/Packet_loss_concealment) on Wikipedia

‎files/en-us/web/api/rtcinboundrtpstreamstats/estimatedplayouttimestamp/index.md‎

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---
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title: "RTCInboundRtpStreamStats: estimatedPlayoutTimestamp property"
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short-title: estimatedPlayoutTimestamp
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slug: Web/API/RTCInboundRtpStreamStats/estimatedPlayoutTimestamp
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page-type: web-api-instance-property
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browser-compat: api.RTCStatsReport.type_inbound-rtp.estimatedPlayoutTimestamp
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---
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{{APIRef("WebRTC")}}
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The **`estimatedPlayoutTimestamp`** property of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates the estimated playout time of this receiver's track.
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This is the [Network Time Protocol (NTP)](https://en.wikipedia.org/wiki/Network_Time_Protocol) timestamp of the last playable audio sample or video frame that has a known timestamp, extrapolated with the time elapsed since it was ready to be played out.
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In other words, it is the estimated current playout time of the track in the NTP clock time of the sender, and can be present even if there is no audio currently playing.
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This can be used to estimate how much audio and video tracks from the same source are out of sync.
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## Value
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A {{domxref("DOMHighResTimeStamp")}}.
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## Specifications
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{{Specifications}}
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## Browser compatibility
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{{Compat}}

‎files/en-us/web/api/rtcinboundrtpstreamstats/fecpacketsdiscarded/index.md‎

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{{APIRef("WebRTC")}}
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The **`fecPacketsDiscarded`** property
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of the {{domxref("RTCInboundRtpStreamStats")}} dictionary is a numeric value
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indicating the number of {{Glossary("RTP")}} Forward Error Correction (FEC) packets
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that have been discarded.
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The **`fecPacketsDiscarded`** property of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates the number of {{Glossary("RTP")}} Forward Error Correction (FEC) packets that have been discarded.
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## Value
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A FEC packet provides parity information that can be used to attempt to reconstruct RTP data packets which have been corrupted in transit.
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This kind of packet might be discarded if all the packets that it covers have already been received or recovered using another FEC packet, or if the FEC packet arrived outside the recovery window and the lost RTP packets have already been skipped during playback as a result.
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The value of {{domxref("RTCInboundRtpStreamStats/fecPacketsReceived","fecPacketsReceived")}} includes these discarded packets.
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An unsigned integer value indicating how many FEC packets have been received whose
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error correction payload has been discarded.
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## Value
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This can happen if all the packets covered by the FEC packet have already been received
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window and the lost RTP packets have already been skipped during playback as a result.
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The value of `fecPacketsReceived` includes these discarded packets.
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An positive integer value.
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## Specifications
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‎files/en-us/web/api/rtcinboundrtpstreamstats/fecpacketsreceived/index.md‎

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{{APIRef("WebRTC")}}
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The **`fecPacketsReceived`** property
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of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates how many
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Forward Error Correction (FEC) packets have been received by this RTP receiver
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from the remote peer.
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The **`fecPacketsReceived`** property of the {{domxref("RTCInboundRtpStreamStats")}} dictionary indicates how many Forward Error Correction (FEC) packets have been received by this RTP receiver from the remote peer.
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An FEC packet provides parity information which can
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be used to attempt to reconstruct RTP data packets which have been corrupted in
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transit.
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A FEC packet provides parity information that can be used to attempt to reconstruct RTP data packets which have been corrupted in transit.
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## Value
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An unsigned integer value which indicates the total number of FEC packets which have
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been received from the remote peer during this RTP session. Forward Error Correction
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uses an exclusive-or method to perform parity checks on the received data.
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An positive integer value.
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By using the FEC parity information to attempt to reconstruct damaged packets, it is
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possible to avoid the need to retransmit damaged packets, which in turn helps to
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reduce lag, or the need to skip damaged frames entirely.
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## Description
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> [!NOTE]
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> This counter may also be incremented when FEC packets arrive
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> in-band along with media content; this can happen with Opus, for example.
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This property indicates the total number of FEC packets which have been received from the remote peer during this RTP session.
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Forward Error Correction uses an exclusive-or method to perform parity checks on the received data.
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By using the FEC parity information to attempt to reconstruct damaged packets, it is possible to avoid the need to retransmit damaged packets, which in turn helps to reduce lag, or the need to skip damaged frames entirely.
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## Usage notes
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> [!NOTE]
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> This counter may also be incremented when FEC packets arrive in-band along with media content; this can happen with Opus, for example.
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It's possible that a subset of the FEC packets which have been received were discarded
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instead of being used. This can happen if the packets covered by the FEC packets have
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already been received successfully or have already been reconstructed using a
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previously-received FEC packet. This may also happen if the FEC packet arrives outside
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the window of time in which the client will attempt to use it.
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It's possible that a subset of the FEC packets which have been received were discarded instead of being used.
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This can happen if the packets covered by the FEC packets have already been received successfully or have already been reconstructed using a previously-received FEC packet.
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This may also happen if the FEC packet arrives outside the window of time in which the client will attempt to use it.
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If you wish to know how many of the received packets were discarded, you can examine
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the value of {{domxref("RTCInboundRtpStreamStats.fecPacketsDiscarded", "fecPacketsDiscarded")}}.
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If you wish to know how many of the received packets were discarded, you can examine the value of {{domxref("RTCInboundRtpStreamStats.fecPacketsDiscarded", "fecPacketsDiscarded")}}.
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## Specifications
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