Table of Contents

Class RTCOutboundRtpStreamStats

Namespace
CSharpToJavaScript.APIs.JS
Assembly
CSharpToJavaScript.dll

The RTCOutboundRtpStreamStats dictionary of the WebRTC API is used to report metrics and statistics related to an outbound {{Glossary("RTP")}} stream being sent by an RTCRtpSender.

[ToObject]
public class RTCOutboundRtpStreamStats : RTCSentRtpStreamStats
Inheritance
RTCOutboundRtpStreamStats
Inherited Members

Remarks

The statistics can be obtained by iterating the RTCStatsReport returned by GetStats(MediaStreamTrack?) or GetStats() until you find a report with the type of outbound-rtp.

-RTCStatsReport

See also on MDN

Constructors

RTCOutboundRtpStreamStats() 

public RTCOutboundRtpStreamStats()

Fields

Active

NOTE
Experimental
The active property of the RTCOutboundRtpStreamStats dictionary is a boolean that indicates whether this RTP stream is configured to be sent, or is disabled. 
[Value("active")]
public bool Active

Field Value

bool

true if the stream is enabled for sending, false otherwise.

Remarks

For example, this would be false if the track corresponding to the stream had its Enabled property set to false.
Note that an active stream may still not be sending if, for example, the stream is limited by network conditions.

See also on MDN

EncoderImplementation 

[Value("encoderImplementation")]
public string EncoderImplementation

Field Value

string

FirCount 

[Value("firCount")]
public ulong FirCount

Field Value

ulong

FrameHeight

The frameHeight property of the RTCOutboundRtpStreamStats dictionary is a positive integer that indicates the height of the last encoded frame, in pixels.

[Value("frameHeight")]
public ulong FrameHeight

Field Value

ulong

A positive integer.

Remarks

Note that the resolution of the encoded frame may be lower than that of the media source, which is provided in Height.

NOTE

The property is undefined for audio streams, and before the first frame is encoded.

See also on MDN

FrameWidth

The frameWidth property of the RTCOutboundRtpStreamStats dictionary is a positive integer that indicates the width of the last encoded frame, in pixels.

[Value("frameWidth")]
public ulong FrameWidth

Field Value

ulong

A positive integer.

Remarks

Note that the resolution of the encoded frame may be lower than that of the media source, which is provided in Width.

NOTE

The value does not exist for audio, or before the first frame is encoded.

See also on MDN

FramesEncoded

The framesEncoded property of
the RTCOutboundRtpStreamStats dictionary indicates the total number of
frames that have been encoded by this RTCRtpSender for this media
source.

[Value("framesEncoded")]
public ulong FramesEncoded

Field Value

ulong

An integer value indicating the total number of video frames that this sender has
encoded so far for this stream.

NOTE
This property is only valid for video streams.

Remarks

See also on MDN

FramesPerSecond

The framesPerSecond property of the RTCOutboundRtpStreamStats dictionary indicates the number of encoded frames sent in the last second.

[Value("framesPerSecond")]
public Number FramesPerSecond

Field Value

Number

A positive number.

Remarks

Note that this may be lower than the media source frame rate, which is provided in FramesPerSecond.

NOTE

The value does not exist for audio, or before the first second of encoding.

See also on MDN

FramesSent

The framesSent property of the RTCOutboundRtpStreamStats dictionary indicates the total number of encoded frames sent on this RTP stream over its lifetime.

[Value("framesSent")]
public ulong FramesSent

Field Value

ulong

A positive number.

Remarks

Note that this may be lower than the total number of media source frames, which is provided in Frames.

NOTE

The value does not exist for audio.

See also on MDN

HeaderBytesSent

The headerBytesSent property of the RTCOutboundRtpStreamStats dictionary indicates the total number of RTP header and padding bytes sent for this synchronization source (SSRC).

[Value("headerBytesSent")]
public ulong HeaderBytesSent

Field Value

ulong

A positive integer.

Remarks

Note that the total number of bytes sent as payload over the transport is equal to: headerBytesSent + RTCOutboundRtpStreamStats.BytesSent.

See also on MDN

HugeFramesSent 

[Value("hugeFramesSent")]
public ulong HugeFramesSent

Field Value

ulong

KeyFramesEncoded

NOTE
Experimental
The keyFramesEncoded property of the RTCOutboundRtpStreamStats dictionary represents the total number of key frames successfully encoded in this RTP media stream.
This includes, for example, key frames in VP8 ({{rfc("6386")}}) or IDR-frames in H.264 ({{rfc("6184")}}). 
[Value("keyFramesEncoded")]
public ulong KeyFramesEncoded

Field Value

ulong

A positive integer.

Remarks

Note that the number of delta frames is calculated by subtracting this value from the total number of frames (FramesEncoded - keyFramesEncoded).

NOTE

The value does not exist for audio.

See also on MDN

MediaSourceId

The mediaSourceId property of the RTCOutboundRtpStreamStats dictionary represents the ID of the statistics object of the track currently attached to the sender of this stream.

[Value("mediaSourceId")]
public string MediaSourceId

Field Value

string

A string containing the ID of a media source statistics object.

Remarks

This can then be used to obtain the corresponding RTCVideoSourceStats or RTCAudioSourceStats.

See also on MDN

Mid

The mid property of the RTCOutboundRtpStreamStats dictionary is a string that contains the media ID negotiated between the local and remote peers.
This uniquely identifies the pairing of source and destination for the transceiver's stream.

[Value("mid")]
public string Mid

Field Value

string

The value of the corresponding Mid, unless that value is null, in which case this statistic property is not present.

Remarks

See also on MDN

NackCount

The nackCount property of the
RTCOutboundRtpStreamStats dictionary is a numeric value indicating the
number of times the RTCRtpSender described by this object received a
NACK packet from the remote receiver.

[Value("nackCount")]
public ulong NackCount

Field Value

ulong

An integer value indicating how many times the sender received a NACK packet from the
receiver, indicating the loss of one or more packets.

Remarks

A NACK (Negative
ACKnowledgement, also called "Generic NACK") packet is used by the
RTCRtpReceiver to inform the sender that one or more {{Glossary("RTP")}}
packets it sent were lost in transport.

See also on MDN

PliCount 

[Value("pliCount")]
public ulong PliCount

Field Value

ulong

PowerEfficientEncoder 

[Value("powerEfficientEncoder")]
public bool PowerEfficientEncoder

Field Value

bool

QpSum

The qpSum property of the RTCOutboundRtpStreamStats dictionary is a value generated by adding the Quantization Parameter (QP) values for every frame this sender has produced to date on the video track corresponding to this RTCOutboundRtpStreamStats object.

[Value("qpSum")]
public ulong QpSum

Field Value

ulong

An unsigned 64-bit integer value which indicates the sum of the quantization parameter (QP) value for every frame sent so far on the track described by the RTCOutboundRtpStreamStats object.
Since the value of QP is typically larger to indicate higher compression factors, the larger this sum is, the more heavily compressed the stream generally has been.

NOTE
This value is only available for video media.

Remarks

In general, the higher this number is, the more heavily compressed the video data is.

See also on MDN

QualityLimitationDurations

NOTE
Experimental
The qualityLimitationDurations property of the RTCOutboundRtpStreamStats dictionary is a map of the reasons that a media stream's quality has been reduced by a codec during encoding, and the time during which the quality was reduced for each reason. 
[Value("qualityLimitationDurations")]
public Dictionary<string, Number> QualityLimitationDurations

Field Value

Dictionary<string, Number>

A Map of quality limitation reasons to a number which represents the time in seconds that the stream has been quality limited for that reason.The allowed quality limitation reason values are the strings:

Remarks

This quality reduction may include changes such as reduced frame rate or resolution, or an increase in compression factor.
The information can be used to diagnose throughput issues and optimize performance.

NOTE

This property only exists for video media.

See also on MDN

QualityLimitationReason

NOTE
Experimental
The qualityLimitationReason property of the RTCOutboundRtpStreamStats dictionary is a string indicating the reason why the media quality in the stream is currently being reduced by the codec during encoding, or none if no quality reduction is being performed. 
[Value("qualityLimitationReason")]
public RTCQualityLimitationReason QualityLimitationReason

Field Value

RTCQualityLimitationReason

A string with one of the following values:

Remarks

This quality reduction may include changes such as reduced frame rate or resolution, or an increase in compression factor.
Note that the user agent report the most limiting factor.
If the most limiting factor cannot be determined the result will be reported using the priority order: &quot;bandwidth&quot;, &quot;cpu&quot;, &quot;other&quot;.

The amount of time the encoded media has had its quality reduced in each of the potential ways that can be done can be found in QualityLimitationDurations.

NOTE

The property is undefined for audio streams.

See also on MDN

QualityLimitationResolutionChanges 

[Value("qualityLimitationResolutionChanges")]
public ulong QualityLimitationResolutionChanges

Field Value

ulong

RemoteId

The remoteId property of the RTCOutboundRtpStreamStats dictionary specifies the RTCOutboundRtpStreamStats.Id of the RTCRemoteInboundRtpStreamStats object representing the remote peer's RTCRtpReceiver which is sending the media to the local peer for this SSRC.

[Value("remoteId")]
public string RemoteId

Field Value

string

A string containing the ID of the RTCRemoteInboundRtpStreamStats object that represents the remote peer's RTCRtpReceiver for the synchronization source represented by this stats object.

Remarks

See also on MDN

RetransmittedBytesSent

The retransmittedBytesSent property of the RTCOutboundRtpStreamStats dictionary represents the total number of payload bytes retransmitted for the source associated with this stream.

[Value("retransmittedBytesSent")]
public ulong RetransmittedBytesSent

Field Value

ulong

A positive integer indicating the number of retransmitted payload bytes for the source.

Remarks

RTX (Retransmission) is one of several mechanisms that WebRTC uses for retransmitting lost payload bytes, and broadly refers to support for the special packets defined in {{rfc("4588")}}.
RTX has to be negotiated between the sender and receiver during initial handshake.
If RTX has not been negotiated the retransmitted bytes will be sent over the same SSRC; otherwise the bytes are sent on a separate SSRC but are still accounted for in this property.

The retransmitted bytes sent are included in the count reported by RTCOutboundRtpStreamStats.BytesSent.

See also on MDN

RetransmittedPacketsSent

The retransmittedPacketsSent property of the RTCOutboundRtpStreamStats dictionary represents the total number of packets retransmitted for the source associated with this stream.

[Value("retransmittedPacketsSent")]
public ulong RetransmittedPacketsSent

Field Value

ulong

A positive integer indicating the number of retransmitted packets for the source.

Remarks

RTX (Retransmission) is one of several mechanisms that WebRTC uses for retransmitting lost payload bytes, and broadly refers to support for the special packets defined in {{rfc("4588")}}.
RTX has to be negotiated between the sender and receiver during initial handshake.
If RTX has not been negotiated the retransmitted packets will be sent over the same SSRC; otherwise the bytes are sent on a separate SSRC but are still accounted for in this property.

The retransmitted packets are included in the count reported by RTCOutboundRtpStreamStats.PacketsSent.

See also on MDN

Rid

The rid property of the RTCOutboundRtpStreamStats dictionary is a string that specifies the RTP stream ID, if defined.

[Value("rid")]
public string Rid

Field Value

string

The value of the encodings.rid argument passed to SetParameters(RTCRtpSendParameters, RTCSetParameterOptions), if any, when the corresponding transceiver is created.
The property is undefined if the stream ID was not set.

NOTE
The property is undefined for audio streams.

Remarks

The property is only defined where the rid has been set for the corresponding RTCRtpSender.
If set, this value will be present regardless of whether the RID RTP header extension has been negotiated.

See also on MDN

RtxSsrc 

[Value("rtxSsrc")]
public ulong RtxSsrc

Field Value

ulong

ScalabilityMode

NOTE
Experimental
The scalabilityMode property of the RTCOutboundRtpStreamStats dictionary represents the scalability mode for the RTP stream, if one has been configured.
If not configured, the property is undefined. 
[Value("scalabilityMode")]
public string ScalabilityMode

Field Value

string

A string indicating the configured scalability mode, such as &quot;L1T1&quot;.
If no mode has been configured, the property is undefined.The set of allowed modes are defined in Scalable Video Coding (SVC) Extension for WebRTC (w3c.github.io/webrtc-svc).

Remarks

The mode is used in Scalable Video Coding (SVC) to define how define how a video stream may be encoded to allow for adaptation to different network conditions and receiver capabilities.
Different codecs allow for various kinds of scalability, such as providing different resolution, different frame rates, or different quality levels for a particular video.

The mode may be configured, for example, by specifying the scalability mode as a parameter option in Configure(VideoEncoderConfig), SetParameters(RTCRtpSendParameters, RTCSetParameterOptions), or AddTransceiver(Union231, RTCRtpTransceiverInit).

NOTE

The property is undefined for audio streams.

See also on MDN

TargetBitrate

The targetBitrate property of the RTCOutboundRtpStreamStats dictionary represents the current target bitrate of the codec in use by the stream.

[Value("targetBitrate")]
public Number TargetBitrate

Field Value

Number

A number representing the target rate in bits per second.
This is defined in the same way as the {{rfc("3890","Transport Independent Application Specific (TIAS)")}} bitrate.

Remarks

The value is dynamic, reflecting the encoder settings and other factors such as network conditions and available bandwidth.
It should correlate closely to the payload bytes sent per second (excluding retransmissions).

-RTCOutboundRtpStreamStats.BytesSent
-RetransmittedBytesSent

See also on MDN

TotalEncodeTime

The totalEncodeTime property of the RTCOutboundRtpStreamStats dictionary represents the total number of seconds spent encoding the RTCOutboundRtpStreamStatsframesEncoded frames for this stream (RTCRtpSender).

[Value("totalEncodeTime")]
public Number TotalEncodeTime

Field Value

Number

A number representing the total time spent encoding frames, in seconds.

Remarks

The average encode time can be calculated by dividing this value by framesEncoded.
The time it takes to encode one frame is the time between sending the encoder a frame and the encoder returning encoded data for that frame.
It does not include any time spent packetizing the resulting data.

NOTE

The property is undefined for audio streams.

See also on MDN

TotalEncodedBytesTarget

NOTE
Experimental
The totalEncodedBytesTarget property of the RTCOutboundRtpStreamStats dictionary represents the sum of the target frame sizes for all of the frames encoded so far. 
[Value("totalEncodedBytesTarget")]
public ulong TotalEncodedBytesTarget

Field Value

ulong

The sum of the target frame sizes in bytes, represented as a positive integer.

Remarks

The codec has a target maximum size for each frame that it is asked to compress, in bytes.
This property indicates the cumulative total of the target sizes for each frame at the current time.
This will probably differ from the total of the actual frame sizes.
You can compare to RTCOutboundRtpStreamStatsbytesSent in order to estimate how closely the codec is matching its target.

The value increases every time RTCOutboundRtpStreamStatsframesEncoded goes up.

NOTE

The property is undefined for audio streams.

See also on MDN

TotalPacketSendDelay

The totalPacketSendDelay property of the RTCOutboundRtpStreamStats dictionary represents the total time in seconds that packets have spent buffered locally before being transmitted.

[Value("totalPacketSendDelay")]
public Number TotalPacketSendDelay

Field Value

Number

The delay in seconds, represented as a number.

Remarks

The individual packet delay is the time between a packet being emitted from the RTP packetizer and it being handed over to the OS network socket.
The individual delay is added to totalPacketSendDelay when RTCOutboundRtpStreamStatsframesEncoded is incremented.

NOTE

The property is undefined for audio streams.

See also on MDN

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