What is WebRTC
WebRTC is peer-to-peer real-time audio, video and data exchange for your website and application
With WebRTC, browsers and apps learn to talk to each other instead of just to web servers. They can share audio and video streams from your microphone and camera, exchange files and images or just send and receive simple messages the fastest possible way: peer-to-peer.
WebRTC is Great for:
Audio/Video Calling – Enable your app or website to support calling between two visitors.
Online Conferencing – Bring people together with voice and video to collaborate with a web-browser.
Live Presentations – Hold live, online product demos or walk potential customers through your web-app in real-time
Direct File Streaming – Add peer-to-peer file sharing to your social app.
Mesh Networking – Enable messaging between computers or even Internet-of-Things (IoT) devices to exchange data without a centralized server.
More About WebRTC
Peer-to-peer, or P2P, describes a connection from a client device to another client device without the use of servers. It’s your mobile phone directly connecting to your colleague’s laptop at work, or to your friend’s tablet at home. With the new open web-standard, WebRTC and its RTCPeerConnection API, your web browser has learned how to do just that, so that data transport from one web browser to another web browser is now possible.
Optimized for Low Latency
Access to Camera, Microphone and Screen
With the getUserMedia API, WebRTC gives your website access to your users microphone, camera and, with the help of browser extensions, your desktop, to send those streams between connected parties, live. The browser makes sure the user is aware that they are granting access to the microphone and camera, and the user must agree to this before being connected to the service.
File Transfer and Messaging
The RTCDataChannel API allows you to send large chunks of binary data and simple text messages from one peer to another. It’s great for application event message exchange or live transfer of images, songs, movies, documents and other files.
Multi Party Conferencing
WebRTC boasts some surprising performance on the client-side. Using a recent mid-end mobile phone you can already host audio/video conversations with 4 and more people.
Beyond Peer-to-Peer – Adding TURN and MCU
Peer-to-peer connectivity is great for real-time audio/video exchange most of the time, but there are some network situations and more demanding performance use-cases that make for a tough peer-to-peer user experience. This is when you need a Media Relay Service like TURN or an MCU, included in the Temasys Skylink Platform, to help out.
- A TURN service is a server that represents your device as a peer outside of restricted networks, e.g. with restrictive firewalls or proxies in the way. Using a TURN service dramatically increases the situations in which a WebRTC connection can be established.
- Increase the number of participants and overall performance when hosting multi-party audio/video sessions. A Multi-Cast Unit (MCU) reduces the amount of streams required on the peer side and allows for more participants especially on CPU and bandwidth limited mobile devices.