WebRTC

WebRTC

• Real Time Communication in the browser
• Peer to peer (WebRTC is the only P2P technology in web browsers)
• Intersection of VoIP and Web
• Released in 2011 by Google as an open-source project
• It is being worked on by both IETF and W3C (technical standards organizations)

WebRTC

Three main JS APIS

• MediaStream (Aquiring audio and video)
• RTCPeerConnection (Does everything.. sets up the connection and sends audio and video, etc)
• RTCDataChannel (low latency arbitrary data)

Where is it available?

 

 

 

 

 

 

 

 

 

 

• There are SDK for mobile, PC and embedded systems

History

• Nov 2011: Chrome releases webRTC
• Jan 2013: Chrome adds webRTC
• Feb 2013: Interoperability
• Sep 2013: Firefox and Chrome add support for the Android versions
• Mar 2014: Opera and Opera for Android support webRTC
• Oct 2014: Microsoft announces ORTC (WebRTC 1.1)

MediaStream

RTCPeerConnection does a lot of work

• Signal Processing
• Codec Handling
• Echo cancellation
• Noise reduction
• P2P Communication (through firewalls and NATs)
• Security
• Bandwidth Management

RTCDataChannel

• It sends data over an existing PeerConnection
• The API is similar to webSockets
• Low latency
• Unreliable or reliable

General process

Signaling

• Signaling needs servers
• Browsers exchange ‘session description’ objects
• What formats are supported, what to send
• Network information for peer to peer setup

Signaling

• Description Protocol format (SDP) looks like this:

v=0
o=- 7614219274584779017 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS
m=audio 1 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 0.0.0.0
a=rtcp:1 IN IP4 0.0.0.0
a=ice-ufrag:W2TGCZw2NZHuwlnf
a=ice-pwd:xdQEccP40E+P0L5qTyzDgfmW
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=mid:audio
a=rtcp-mux
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9c1AHz27dZ9xPI91YNfSlI67/EMkjHHIHORiClQe
a=rtpmap:111 opus/48000/2
…

Signaling

• There is no defined protocol to do the signaling on webRTC
• The decision is left to the developer
• Options: Ajax, Websockets, DataChannel, SIP or XMPP/Jingle

NAT Traversal

• NAT devices are used to alleviate IPv4 address exhaustion by allowing the use of private IP addresses on home and corporate networks behind routers with a single public IP address facing the public Internet.
• The Nat-traversal protocols used by WebRTC are ICE, STUN, and TURN
• Special servers are needed to set the connection.

NAT Traversal

STUN

• Tells me what my public IP address is.
• Simple server, cheap
• With the information, webRTC can perform NAT hole punching

STUN

 

TURN

• STUN works around 80% of the time
• TURN as is used as a fallback media relay
• Data is sent through the TURN server

TURN

ICE

• Tries to find the best path for each call
• It uses STUN if it can
• ICE uses a technique known as “hole punching”

Resources

• WebRTC book: http://webrtcbook.com
• Google demos: https://github.com/googlechrome/webrtc
• webRTC blog: http://bloggeek.me

DEMO