Varun presented some key numbers from callstats.io WebRTC monitoring and analytics plaform at the WebRTC Conference Expo in Paris December 2015. Browsers, Operating systems, participants, relay types, IPv6, ICE candidates, Setup times, failure reasons, time to failure, churn, RTT distribution, TURN & NAT.

1. Performance in the Wild:
Varun Singh, CEO
18.12.2015
Upperside Paris
One year of measuring
WebRTC service quality
http://geek-and-poke.com/
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2. Analytics for WebRTC
• the first cloud-based monitoring and
management service for WebRTC (audio and
video).
• The team is contributing to the core standards
that enable the WebRTC technology.
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3. Monitoring What?
• Annoyances
• Transport quality
• Per-stream media quality
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4. Overall Metrics
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5. Conference Timeline
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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. Growth
Jan Feb Mar Apr May Jun Jul Aug Sep Oct Nov Dec
purely web endpoints
support:
• TURN,
• VideoBridge
no mobile, yet
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9. Browsers
Chrome Firefox Opera Temasys
95%
5%
0.05%
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10. Browsers
Chrome Firefox Opera Temasys
95%
5%
0.05%
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11. OSes
0
12,5
25
37,5
50
Android Windows Mac Linux
11
49%
34%
13%
4%
using a browser

12. Number of Participants
0
10
20
30
40
50
60
70
80
90
100
2 3 4 5 6-8 >8
12
64%
30%
0,3%3%

13. 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%
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14. IPv6?
0
20
40
60
80
100
IPv4 IPv6
Not all is lost, for
Europe alone
0
20
40
60
80
100
IPv4 IPv6
14
97%
83% 17%

15. how many ICE candidates?
15
0
20
40
60
80
100
0-4 5-8 8-20 >20
host
stun
turn
v4
v6
multi homed
24 candidates: 8%
68%
8%
17%
7%

16. Setup times
0
10
20
30
40
50
<1s 1-2s 2-5s 5-10s >10s
16
47%
23%
14%
7%
9%

17. Failure Reasons~9%
calls fail
5% 95%<1%
~20% of the setup
calls have issues
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18. Time to Failure
0
10
20
30
40
50
<10s
10-30s
30-60s
>60s
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People are
very patient!
perhaps because no mobile
50%33%
7%
10%

19. 0
20
40
60
80
100
no rejoins once twice thrice
Churn* *participant rejoins the
same call repeatedly
Average joins per
participant per conference
76%
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20. Distribution of RTTs
20
0
10
20
30
40
<20m
s
20-50m
s
50-150m
s
150-400m
s
0.4-1s
1-3s
>3s
40%
21%
9%
11%4% 4%
1%

21. Summary
• 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
• Failures: NAT/FW still causes most failures
• Churn: 25% of sessions have a participant
rejoining
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22. A 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: reportError() —> call failed to be setup due to SDP or Firewalls.
• optionally: associateMstWithUserID() —> bridges using multiple media streams in
a single peerconnection may want to correlate mediastream quality across
endpoints.
• optionally: collect and send user feedback
http://www.callstats.io/api/22

23. 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 w o r l d w h e r e r e a l - t i m e
communication is both frictionless and
effortless to set up, operate, and scale.”
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