Varun Singh, callstats.io CEO, gave a presentation at Enterprise Connect 2016. He talked about how WebRTC is being used and what are the analytics behind it.

1. An analytics
perspective
Varun Singh, CEO
Orlando, 07.03.2016
@callstatsio @vr000m

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

3. callstats.io
First 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

4. 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

5. It is not all about throughput*
• … it is about latency or delay!
*)capacity
bandwidth
speed
audio only call

6. Disruptions
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.
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7. Disruptions and user behaviour
User Behaviour: The user tries
to correct for the disruption by
turning on and off video
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8. state of webrtc
(as we observe it)
8

9. multiparty?
0 %
10 %
20 %
30 %
40 %
50 %
60 %
70 %
80 %
90 %
100 %
2 3 4 >5
9
65%
31%
3% 1%

10. ~12% calls
fail to set up
10

11. Failure Reasons
14% 85%<1%
~20% of the setup
calls have issues
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12. 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

13. getUserMedia() errors?
0 %
25 %
50 %
75 %
100 %
Perm
ission
D
enied
D
evice
notfound
Perm
ission
D
ism
issed
constraintnotsatisfied
device
notsupported
1 %3 %5 %
35 %
56 %
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14. 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
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15. Dropped calls
0 %
20 %
40 %
60 %
80 %
100 %
connectivity
lost
high
delay
high
loss
battery
crash
16 %
3 %14 %
23 %
44 %
15
one-way
audio/video

16. Time to Failure
0 %
20 %
40 %
60 %
80 %
100 %
<10s
10-30s
30-60s
>60s
16
People are
very patient!
*not mobile
50%
33%
7% 10%

17. 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

18. 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
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19. 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
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20. Vision
Gather
Report
Analyze
FIX
Deploy?
webrtc-internal?
via email?
Hasthisbeen
reportedbefore?
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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21. Free developer account
https://dashboard.callstats.io/register