kranky geek sao paulo 2016 - webrtc statistics and analytics

WebRTC Statistics and Analytics

Varun Singh CEO, @callstatsio

twitter: @vr000m São Paulo, 22-11-2016

Multimedia Systems

• connected to a network

• capture and transmit media

• receive and render media

• video codecs: e.g., H.264, VP8, VP9, …

• audio codecs: e.g., G.711, Opus, …

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Real-time Transport Protocol (RTP)

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Sender Receiver

• Dejittering, sync, playout• Monitoring + reporting

RTP media stream(encoded media, FEC, repair)

RTP Control Protocol (RTCP)

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Sender Receiver

RTCP Sender Reports (SRs)

RTCP Receiver Reports (RRs)



RTCP Sender Report (SR)

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Sender Receiver


RTCP Sender Reports (SRs)• Timing, synchronisation• Sending rate, packet count

RTCP Receiver Reports (RRs)


RTCP Receiver Report

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Sender Receiver



RTCP Receiver Reports (RRs)• Rough statistics• Congestion cues


Adaptation

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Sender Receiver



RTCP Receiver Reports (RRs)• Rough statistics• Congestion cues

• Dejittering, sync, playout• Monitoring + reportingShort-term adaptation

• Error-resilience (NACK, PLI)• Congestion control• Adaptive source coding

http://geek-and-poke.com/

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http://geek-and-poke.com/

It is not all about throughput*

… it is about latency†! *)capacity bandwidth

speed

audio only call

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†)delay, one-way delay

time (sec)

Measuring User Experience

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collect metrics and user actions C

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measure setup time

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Ask the right question!

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represents overall quality

What to Measure?• Network metrics

• bits per second, RTT, jitter, packet losses, …

• Multimedia pipeline metrics• playout delay, frames metrics, …

• Quality Models from metrics

• Annoyances• resolution/frame rate changes, interface changes, …

• failure to setup (NATs, incompatibility)

• dropped calls (insufficient capacity, high latency, high losses)

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getStats() API

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Track

pc.getStats(mst, …)

IceTransportDtlsTransport

PeerConnection

Tracks

Datachannels

RtpReceiver

SctpTransport

caninvoketheAPIasoften(>150ms)

mostmetricsarecumulativemetrics

pc.getStats(null, …)

partial interface RTCPeerConnection { Promise<RTCStatsReport> getStats(optional MediaStreamTrack? selector = null);};

getStats() APIExample: var selector = pc.getRemoteStreams()[0].getAudioTracks()[0];

pc.getStats(selector).then(function (report) {

processStats(report);

})

.catch(function (error) {

log(error.toString()); //if something went wrong

});

function processStats(report) {

for (var i in report) {

var now = report[i];

if (now.type == "outbound-rtp") {

//do something

}

}

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Output:

{

timestamp:1479753828

ssrc:3938144725

packetsSent:15656

bytesSent:1562150

roundTripTime:31

…

}

Example

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Internet

PeerConnection

PeerConnection

Alice

Bob

media direction

Sender-side pipeline

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Internet


PeerConnection

Tracks RtpReceiver

DtlsTransport IceTransport

PeerConnection

Tracks RtpSender

media direction

Receiver-side pipeline

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PeerConnection

Tracks RtpReceiver


PeerConnection

Tracks RtpSender

Internetmedia

direction

Frame height

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PeerConnection

Tracks RtpSender

RTP Stats

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PeerConnection

Tracks RtpSender

ICE Stats

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PeerConnection

Tracks RtpSender

Simplified E-model (audio)

23ITU-TREC-G.114https://en.wikipedia.org/wiki/Mean_opinion_score

https://en.wikipedia.org/wiki/Mean_opinion_score

Example (1/3)

var selector = pc.getRemoteStreams()[0].getAudioTracks()[0];var rttMeasures = [];

setTimeout(function () { pc.getStats(selector, function (report) {

// do something }, logError);}, 1000);

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Example (2/3)var selector = pc.getRemoteStreams()[0].getAudioTracks()[0];var rttMeasures = [];

setTimeout(function () { pc.getStats(selector, function (report) { for (var i in report) { var now = report[i]; if (now.type == "outbound-rtp") { //do something }

} }, logError);}, 1000);

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Example (3/3)var selector = pc.getRemoteStreams()[0].getAudioTracks()[0];var rttMeasures = [];

setTimeout(function () { pc.getStats(selector, function (report) { for (var i in report) { var now = report[i]; if (now.type == "outbound-rtp") { rttMeasures.append(now.roundTripTime); var avgRoundTripTime = average(rttMeasures); var emodel = simplemodel(avgRoundTripTime); console.log ("e-model: "+str(emodel));

//can add jitter, losses, framesDiscarded, etc }

} }, logError);}, 1000);

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What more can you do?

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ICE States

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https://w3c.github.io/webrtc-pc/

https://w3c.github.io/webrtc-pc/

Disruption: loss of connectivity when network interfaces change, low available capacity,or high delay

The light grey vertical lines show disruption, highlighted by the red bounding boxes.29

Network Disruptions

Disruptions and user behaviour

User Behaviour: The user tries to correct for the disruption by turning on and off video

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Lessons learnt

• Deploy TURN servers with TCP support

• Detect crashes, disruptions and re-setup connections

• media pipelines sometimes crash

• screen sharing plugin crash

• network connectivity re-establishment

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Summary

• RTP and RTCP

• getStats() API

• Simplified E-model

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kranky geek sao paulo 2016 - webrtc statistics and analytics

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