WebRTC vs Websockets: If WebRTC can do Video, Audio, and Data, why do I need Websockets?

WebRTC is designed for high-performance, high quality communication of video, audio and arbitrary data. In other words, for apps exactly like what you describe.

WebRTC apps need a service via which they can exchange network and media metadata, a process known as signaling. However, once signaling has taken place, video/audio/data is streamed directly between clients, avoiding the performance cost of streaming via an intermediary server.

WebSocket on the other hand is designed for bi-directional communication between client and server. It is possible to stream audio and video over WebSocket (see here for example), but the technology and APIs are not inherently designed for efficient, robust streaming in the way that WebRTC is.

As other replies have said, WebSocket can be used for signaling.

I maintain a list of WebRTC resources: strongly recommend you start by looking at the 2013 Google I/O presentation about WebRTC.

WebSockets:

• Ratified IETF standard (6455) with support across all modern browsers and even legacy browsers using web-socket-js polyfill.

• Uses HTTP compatible handshake and default ports making it much easier to use with existing firewall, proxy and web server infrastructure.

• Much simpler browser API. Basically one constructor with a couple of callbacks.

• Client/browser to server only.

• Only supports reliable, in-order transport because it is built On TCP. This means packet drops can delay all subsequent packets.

WebRTC:

• Just beginning to be supported by Chrome and Firefox. MS has proposed an incompatible variant. The DataChannel component is not yet compatible between Firefox and Chrome.

• WebRTC is browser to browser in ideal circumstances but even then almost always requires a signaling server to setup the connections. The most common signaling server solutions right now use WebSockets.

• Transport layer is configurable with application able to choose if connection is in-order and/or reliable.

• Complex and multilayered browser API. There are JS libs to provide a simpler API but these are young and rapidly changing (just like WebRTC itself).

Websockets use TCP protocol.

WebRTC is mainly UDP.

Thus main reason of using WebRTC instead of Websocket is latency. With websocket streaming you will have either high latency or choppy playback with low latency. With WebRTC you may achive low-latency and smooth playback which is crucial stuff for VoIP communications.

Just try to test these technology with a network loss, i.e. 2%. You will see high delays in the Websocket stream.