What is WebRTC and How Does It Work?

###This is a summary of a post at innoarchitech

WebRTC stands for Web Real-Time Communications

###Intro

1. A set of plugin-free APIs
2. Leverages multiple standards and protocols such as data streams, STUN/TURN servers, signaling, JSEP, ICE, SIP, SDP, NAT, UDP/TCP, network sockets, and more

###Detail

####Peer-To-Peer Communication

• audio and video communications

• each person’s web browser must agree to begin communication, know how to locate one anohter, bypass security and firewall protections, and transmit all multimedia communications in real-time

• knowing how to locate and establish a network socket connection with another computer’s web browser is crucial

  • you need to bidirectionally transmit multimedia data
  • difficult to make a request to another computer because it is hard to know exactly where it is.
    • you have to send a request by sending your audio/video data while receiving it from the othe rside without going through an external server

####Firewalls and NAT Traversal

• Firewall enforces a set of rules about what data packets will be allowed to enter of leave a network. some sort of a security system

• computer sits behind a firewall and Network Access Translation device (NAT)

  • which is why your computer does not have a static public IP address
  • NAT device translates private IP address from inside a firewall to public-facing IP addresses
    • NAT is needed for security
    • NAT makes your request IP to look different from your actualy IP address

• STUN(Session Traversal Utilities for NAT) and TURN(Traversal Using Relays around NAT) servers allow you to get someone else’s IP to make a call by sending audio/video data

  • a request for your public-facing IP address is first made to a STUN server
    • you tell your peers to send a request to your server via your public-facing IP address

####Signaling, Sessions, and Protocols

• negotiate and establish the network session connection with your peer

  • The initial session negotiation and establishment happens using a signaling/communication protocol specialized in multimedia communications
    • something like Session Initiation Protocol(SIP)
      • the chosen signaling protocol must work with an application layer protocol called the Session Description Protocol (SDP)

• when you try to communicate with someone, you generate a set of Interactive Connectivity Establishment(ICE) protocol candidates

  • the candidates represent a given combination of IP address, port, and transport protocol to be used

  

####Complete Process Summarized

• Each peer first establishes its public-facing IP address by sending a request to STUN

  • Signaling data channels are then dynamically created to detect peers
    • this channel is somewhat like a private room
    • only those who know about the room can send and receive messages
    • you need a unique idenfitier to access it
    • but some protocols do not require a channel since webRTC is flexibile and does not specify the signaling process

• support peer-to-peer negotiations and session establishment

  • once two or more peers are connected to the same channel, the peers are able to communicate and negotiate session information
    • initiating peer sends an offer using a signaling protocol and waits to receive an answer from any receivers that are connected to the given channel
    • Once the answer is received, a process occurs to determine and negotiate the best of the ICE candidates gathered by each peer.

• once the optimal ICE candidates are chosen,

  • things like all of the required metadata, network routing, and media information is agreed
  • the network socket session between the peers is then fully established and active. - local data streams and data channel endpoints are created by each peer, and multimedia data is finally transmitted both ways

• if the process of agreeing on the best ICE candidates fails, the fallback is to use a TURN server as a relay instead of using STUN

  • basically employs a server that acts as an intermediary and relays any stransmitted data between peers.
  • when TURN is used, each peer no longer needs to know how to contact and transmit data to each other.
    • they need to know what public TURN server to send and receive real-time multimedia data during a communication session
  • TURN servers need to be quite robust, have extensive bandwidth and processing capabilities, and handle potentially large amounts of data.

####WebRTC JavaScript APIs

• WebRTC and the processes described are implemented through a set of JavaScript APIs that actually produce and transmit the multimedia data being used for real-time communications.
• The primary WebRTC APIs include, Navigator.getUserMedia (capture audio and video), RTCPeerConnection (create and negotiate peer-to-peer connections), and RTCDataChannel (represents a bidirectional data channel between peers).