WebRTC Standards from Tim Panton

1. WEBRTC STANDARDS AND
UNDERPINNINGS
Tim Panton, Westhawk Ltd - @steely_glint

2. What is webRTC?
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Haven’t used webRTC
Your laptop is on wifi
Have Chrome/firefox
Browse to:
http://phono.com – click ‘call’

3. What did we just do?
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Placed a video call with no plugins
zero config
just by browsing to a site
encrypted
free
over local wifi

4. WebRTC Definition
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Realtime
voice / video /data
browser based
no plugins
secure
interoperable

5. Standards
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IETF
W3C
Loosely based on pre-existing standards
$100s Millions of IPR - donated
Google, Cisco, Mozilla, Skype, Tropo, ATT, E///,
Lucent etc.

6. Big Picture
HTTP(s) signaling via webserver
Peer to Peer media between Browsers

7. Protocol standards (IETF)
The IETF has responsibility for the wire protocols
in RTCWeb

8. Signaling Standards
None!
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It is up to the javascript in the browser
to do what ever is needed.

9. Media Standards (RIA 2.0)
Many!
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STUN
ICE
TURN
DTLS
SRTP
RTCP
OPUS
ULAW

10. Why so many?
The network environment of a web browser is not the
same as a desk phone.
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Security – hostile lans (coffee shops/hotels)
Variablity – home networks, wifi/3g
Programability – Javascript is dynamically
loadable
This stack of media standards addresses the
differences.

11. NAT
NAT
STUN and ICE address NAT
ICE sends multiple STUN packets down all possible interfaces to try and find a
path. First Bi-directional route found is used.

12. TURN
NAT
NAT
TURN is for when ICE fails to find a viable path
A TURN server in the cloud acts as a packet reflector
Many Telco 3g networks isolate users from each other.

13. MS
NAT
NAT
Bridging via a media server.
If additional services are required – recording, conferences, PSTN interop etc. then
bridging via a media server may be required.

14. DTLS
DTLS is the UDP version of TLS (as used in https etc)
It serves 2 purposes in the WebRTC
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Exchange keys used by the SRTP media encryption
Carry the data channel streams

15. Demo – sharefest.me
Data channel provides Peer 2 Peer data between
browsers.
It can be used for file transfer, game moves etc.
It may also become important in M2M or IOT as a secure
NAT friendly P2P protocol.
Demo show file transfer between 2 browsers by sharing a
URL, but not through that server.

16. SRTP + RTCP
Encrypted version of the classic RTP protocol, with the
RTCP reporting mechanism.
Multiple media streams may be multiplexed over the
same pair of ports – still under discussion.

17. Standard Codecs
Audio
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OPUS
Wideband
Flexible, efficient, loss correcting
Expensive to transcode
ULAW
Narrowband PSTN codec
High bandwidth
Poor in lossy/variable networks

18. Codecs - No video standard yet.
Video
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VP8
H264
Differences are largely commercial and
legal rather than technical, either is plenty
good enough.

19. Standard APIs (W3c)
These are the API’s offered to the javascript
programmer – aka ‘JSEP’

20. Javascript : getUserMedia()
navigator.webkitGetUserMedia({
'audio':true,
'video':true
},
function(stream) {
var url =webkitURL.createObjectURL(stream);
createPeer(stream);
},
function(error) {
});

21. Javascript : RTCPeerConnection
pc = new RTCPeerConnection(configuration,constraints);
pc.onicecandidate = function(evt) {
sendCandyToAlice(evt.candidate);
};
pc.onaddstream = function (event) {
var remotePlay = document.getElementById(”videoTag");
remotePlay.src = webkitURL.createObjectURL(event.stream);
};
pc.addStream(localStream);
pc.createOffer(
function(offer) {
pc.setLocalDescription(offer);
sendOfferToAlice(pc.localDescription.sdp);
},
null, constraints);

22. HTML : <Video> tag
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Extended to accept a webRTC stream as a source
Either a local or remote
See previous slides
Additional mute etc.

23. Javascript:RTCSessionDescription
function onMessageFromAlice(message){
var sd = new RTCSessionDescription(
{'sdp':message,'type':"answer”} );
pc.setRemoteDescription(sd,sessionOk,sessionFail);
}

24. Dissention with SDP as an API
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SDP is emitted by pc.createOffer()
Complex SDP
May be manipulated to select codecs/candidates…
Source of much complaint
Currently ill defined
May be replaced in 2.0 std

25. Other co-operating W3C APIs
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WebGL
Can be used to apply effects to video
WebAudio
Can be used to apply effects to audio
Both will be able to be applied to local or remote
streams

26. Javascript libraries
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Most web coders will use a library
JS Sip
ATT.js
Phono
OpenRTC

27. Demo – WebGL effects
Demo of WebGL doing realtime effects on a webRTC
video stream