An analytics perspective

Varun Singh, CEO

Orlando, 07.03.2016

@callstatsio @vr000m

http://geek-and-poke.com/ 2

callstats.ioFirst cloud-based monitoring service for WebRTC

• 40 paying customers

• +80 developers accounts

Jan Feb Mar Apr May Jun Jul Aug Sep Oct Nov Dec Jan Feb3

Very Simple API

• 3 lines of code —> 5 minutes.

1. include <script>

2. initialize() —> needs registration keys

3. addNewFabric(pc,…) —> call started

• optionally: send user events: audio muted/unmuted, video paused/resumed, call terminated, call held.

• optionally: collect and send user feedback

http://www.callstats.io/api/4

It is not all about throughput*• … it is about latency or delay!

*)capacity bandwidth speed

audio only call

Disruptions

Disruption: loss of connectivity when network in ter faces change, low available capacity,or high delay

The light grey vertical lines show disruption, highlighted by the red bounding boxes.6

Disruptions and user behaviour

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

7

state of webrtc

(as we observe it)

8

multiparty?

0 %

10 %

20 %

30 %

40 %

50 %

60 %

70 %

80 %

90 %

100 %

2 3 4 >5

9

65%

31%

3% 1%

~12% calls fail to set up

10

Failure Reasons

14% 85%<1%

~20% of the setup calls have issues

11

Types of Relays

0 %

10 %

20 %

30 %

40 %

50 %

60 %

70 %

80 %

90 %

100 %

No Relay TURN/UDP TURN/TCP TURN/TLS

78%

13% 7% 2%

12

22% of calls need infrastructure

getUserMedia() errors?

0 %

25 %

50 %

75 %

100 %

Permiss

ion Den

ied

Device

not fo

und

Permiss

ion Dism

issed

cons

traint

not s

atisfie

d

device

not s

upport

ed

1 %3 %5 %

35 %

56 %

13

0 %

20 %

40 %

60 %

80 %

100 %

no rejoins once twice thrice

3 %4 %17 %

76 %

Churn* *participant rejoins the same call repeatedly

Average joins per participant per conference

14

Dropped calls

0 %

20 %

40 %

60 %

80 %

100 %

conn

ectivi

ty los

t

high d

elay

high l

oss

battery

crash

16 %3 %14 %23 %

44 %

15

one-way audio/video

Time to Failure

0 %

20 %

40 %

60 %

80 %

100 %

<10s

10-30

s

30-60

s>60

s

16

People are very patient!

*not mobile

50%

33%

7% 10%

Lessons Learnt• ask for audio and video device permissions

separately

• deploy TURN servers with UDP and TCP support

• detect crashes and re-setup connections

• media pipelines sometimes crash

• screen sharing plugin crashes

Summary (1/2)

• Browsers: Chrome dominates WebRTC

• OSes: Windows and Mac are pretty even

• Participants: ~3 participants in a call on average

• Relays: ~20% sessions need a TURN server

18

Summary (2/2)

• Setup time: 80% sessions setup in <5s

• Call Setup Failures: ~10% of calls fail to setup

• Reason for failure: 85% due to NAT/FW

• Call Drop: 20% calls fail after setup

• Churn: 25% of sessions have a participant rejoining

19

VisionGather

Report

Analyze

FIX

Deploy?

webrtc-internal?

via email?

Has

this

bee

n re

port

ed b

efor

e?

Did it solve the issue?Until next report?

Collect

Diagnose

Fix

Deploy

A world where real-time communication is both frictionless and effortless to set up, operate, and scale.

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Free developer account

https://dashboard.callstats.io/register