WebRTC Introduction & Basics
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WebRTC is an open-source project that enables real-time communication in browsers and mobile applications via simple APIs, supported by major browsers like Google, Mozilla, and Opera. It uses protocols such as STUN and TURN for NAT traversal, with the RTC Peer Connection at its core to manage the connection process, including signaling between caller and callee. The architecture can handle various network conditions and offer optimal communication paths through ICE (Interactive Connectivity Establishment).
WebRTC Introduction & Basics
- 2. WebRTC WebRTC is FOSS that provides browsers and mobile applications with Real Time Communication capabilities via simple API’s It is supported by almost all modern browsers Mobile applications & desktop applications can also embed WebRTC without much hassle It leverages multiple protocols and standards like TURN/STUN(Traversal Using Relay NAT/Session Traversal Utilities for NAT) ICE(Interactive Connectivity Establishment) SIP(Session Initiation Protocol) SDP(Session Description Protocol) It is a project supported by Google, Mozilla & Opera
- 3. WebRTC It can be used for low cost, high quality audio, video & data communication WebRTC itself is written in C++ Available for android and iOS integration without any hassle RTCPeerConnection is the heart of WebRTC, handles lots of things including
- 4. Connection Process Caller creates an offer SDP using some constraints Send this SDP to the callee via some signaling mechanism Callee accepts the offer and generate an answer SDP Answer SDP is sent to caller using some signaling mechanism Meanwhile both caller and callee discovers ICE candidates and exchange them with each others using signaling mechanism Both machines have 2 SDP’s, i.e. local and remote Both machines have ICE candidates from each other WeRTC connects both machines using ICE Candidates Connection is Peer-Peer.
- 5. Signaling Mechanism Signaling is required to transmit following from caller to callee and vice versa Session Description ICE Candidates It can use any protocol & messaging mechanism XMPP has an XEP for this called Jingle You can use WebSockets for this as well. WebRTC doesn’t restrict any protocol Signaling is lightweight, it doesn’t transmit voice, video or data, instead it only transmits SDP’s and Candiates
- 8. NAT Network Address Translation is a process in which a device(usually firewall) assigns a public IP to a computer or group of computers inside a private network. Main purpose of NAT is to limit the number of public IP addresses a network or organization must use. Symmetric NAT also map a specific port network’s public IP, every time a new connection is made a new port will be mapped.
- 9. NAT continued If systems are behind NAT then simply sending candidates doesn’t work If candidates are local IP addresses and ports and both machines resides on different private networks then they won’t be able to connect. We somehow need to know the public IP of the caller and a port mapped to that public IP. Similarly we need to know the same information on second public
- 11. STUN STUN(Session Traversal Utilities for NAT) provides a set of tools for NAT traversal. It actually probes your public IP, bind a port on NAT machine to your local machine It requires a server for NAT traversal
- 12. NAT with STUN
- 13. TURN Sometimes peer-peer connection fails because of some of the following reasons Firewalls blocking peer-peer connections because of custom ports NAT device not allowing ports binding State restrictions based on type of traffic TURN(Traversal Using Relay Network) provide a fallback mechanism if peer-peer connection fails It requires a Relay Server to act as a relay data among different machines Ensures that calling works in almost all of the environments
- 14. NAT with TURN
- 15. ICE(Interactive Connectivity Establishment) ICE is basically a mechanism to find multiple paths to communicate It also provides costs of different paths so that all parties use optimal paths to communicate and user experience is improved Intention is to find best route between caller and callee. It leverages STUN to find routes
- 16. Topologies supported One-One Call Mesh: Small N-way call Star: Medium N-way call MCU(Multipoint Conferencing Unit): N-way call
- 17. Questions?