WebRTC - High Level Look - Part III - Servers and Protocols

Abstract Signalling

- This is needed to exchange 'Session Description objects'

- What formats the originator end support, what codec they use, what type of security,

- Network information for peer-to-peer setup 

This singling part can be any, it could be XHR, it could be SIP, XMPP or Websockets etc

The overall architecture is like below 

The overall working is like below 

The app gets the session description from the browser, sends across the clouds to the other side, Once it gets the message back from the other side with the other side's session description, both sides session description is passed down to the webrtc in the broser. WebRTC can then set up and conduct the media. 

An RTP Session description could look like something below. Many apps can alter this to make it custom, but this is a basic information. 

v=0 

o=762434233223 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:W2TGWEQWEWQ

a=ice:pwd:xdQerwerew 

a=extmap:1 urn:ietf:params:rtp=hdrext:ssrc-audio-level

a=mid:audio

a=rtcp-mux

..

…

References: 

http://www.html5rocks.com/en/tutorials/webrtc/basics/