What does the WebRTC standards have to offer for extracting debug information and how can this be used in your implementation.

1. WebRTC Standards &
Implementation Q&A
Amir
Zmora
TheNewDialTone
Dan
Burnett
StandardsPlay
Alex
Gouaillard
WebRTC by
Dr
Alex
/
Citrix

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7. About Us
• Amir Zmora • Dan Burnett • Alex Gouaillard

8. Save The Date: October 19
Register Now: http://ccst.io/e/webrtcstandards9
Next Session

9. On The Agenda:
Debugging Networking Issues in
WebRTC

10. Debugging networking – within your app
• Standards APIs don't provide post-­execution debugging
• APIs give you live information usable by your app itself to diagnose problems
• Three main areas
• Error events
• Status variables
• Live statistics

11. Error events
• icecandidateerror: will tell you which STUN or TURN server had an error
• error (on data channel)

12. Status variables
• General: pc.{signaling, iceGathering, iceConnection, connection}State
• Related events: {signaling, icegathering, iceconnection, connection)statechange
• Be aware of disconnected and failed in iceConnectionState and/or connectionState
• These can make you aware of a problem, but they don't help much in diagnosing it. They also
can be "slow" to indicate a network problem because they are composite errors
• Per flow: {sender, receiver}.{transport, rtcpTransport}.{state,gatheringState}
• Related events: {state,gatheringstate}change
• failed here also provides awareness of a problem but not diagnosis
• Per flow: {sender, receiver}.{transport,
rtcpTransport}.get{SelectedCandidatePair,{Local,Remote}Parameters}
• gives you the selected candidate pair and the u/p for each end

13. Statistics at the Candidate Pair level
• Counters: Sent/Received values for bytes, requests, responses, retransmissions,
consentRequests, consentResponses (e.g. bytesSent, consentResponsesReceived)
• Calculated current and total Rtt
• Calculated available{Outgoing,Incoming}Bitrate

14. Status variables

15. Ice Candidate Error values (courtesy: Mozilla doc)

16. Making Sense of PeerConnection State Machines

17. ICE
Connection
new
checking
connected
disconnected
failed
Completed
close
new
gathering
complete
ICE
Gathering
stable
Have-­‐remote-­‐offer
stable
Peer
Connection
Trying
to
make
Sense
of
PeerConnection State
Machines
…..
Is
useless
without
the
big
picture
…..
Close

18. DONE
FROZENIN
CALL
CONNECTION
HANDSHAKE
(SDP
O/A)
.
new
checking
connected
disconnected
failed
Completed
close
new
gathering
complete
CALLER
SIG-­‐SERVER
CALLEE
stable
have-­‐local-­‐offer
stable
Close
CHAT
ENTER
WELCOME
OFFER
ANSWER
Create
PC
Add
local
stream(s)
Create
offer
<modify
sdp>
SetLocal(offer)
Sending
offer Create
PC
SetRemote(offer)
addRemoteStream(s)
Add
local
stream(s)
Create
answer
<modify
sdp>
SetLocal(answer)
Send
answer
stable
Have-­‐remote-­‐offer
stable
new
gathering
complete
onIceCandidate
<Filter
candidates>
Send
candidate
<Filter
candidates>
addIceCandidate
onIceCandidate
<Filter
candidates>
Send
candidate
<Filter
candidates>
addIceCandidate
new
checking
connected
disconnected
failed
Completed
close Close
©
Temasys Communications,
pvt,
ltd,
2014
Document
provided
under
CC
BY-­‐NC
4.0
CANDIDATES
Peer
Connection
ICE
Connection
ICE
Gathering
SetRemote(answer)
addRemoteStream
BYE

19. And that was for the “simple” 1:1 case, with trickle ice
• Multiparty?
• Simulcast?
• Object Oriented API?
• …
• Ask Dan or alex (or both) for consulting J

20. Hacks: How to infer infos on the network in the browser
based on ICE
-­ ask the ice gathering to try to generate a relay candidate with transport = tcp and the provided
turn server
=> if the turn server exists (suppose it is, google provides it)
=> if the credential are ok (again, suppose it is)
=> if there is a local port that let tcp through
=> if the turn server is configured to receive TCP on the corresponding port (suppose it is)
THEN
a TCP relay candidate is created during the gathering.
By getting a relay candidate (or not) knowing all the other steps, you infer if you have any tcp
port open.
Rince and repeat for udp, tls dtls, just like the https://test.webrtc.org code does.

21. Ask the browser for more info, traces and log
• Webkit Testing: Tool/Script/run-­webkit-­tests
• Mozilla Mochitests for WebRTC, and extra logs
• https://wiki.mozilla.org/Media/WebRTC/Testing
• https://wiki.mozilla.org/Media/WebRTC/Logging
• Chrome Logging:
• chrome.exe -­-­enable-­logging -­-­vmodule=*/webrtc/*=2,*/libjingle/*=2,*=-­2 -­-­no-­sandbox

22. Some low level tools (chrome canary => no encryption)
Monitoring tools:
• Ganglia + Nagios
Operating system utilities
• Dig for DNS
• Netstat or lsof for socket verification (IP:port)
• S_client (openssl)
Packet sniffers
• Tcpdump / Wireshark.
• If no encryption: ngrep
Debugging mode
• Recompiled/weaponized browsers (not for the faint of heart)
WebRTC and WebSockets
• Mozilla Firefox => about:webrtc
• Google Chrome =>chrome://webrtc-­internals.

23. ?

24. Save The Date: October 19
Register Now: http://ccst.io/e/webrtcstandards9
Next Session

25. Session sponsored by

26. Thank You
Amir
Zmora
TheNewDialTone
Dan
Burnett
StandardsPlay
Alex
Gouaillard
WebRTC by
Dr
Alex