What is the difference between WebRTC and WebSockets for low level data communication

Question

I am trying to understand the difference between WebRTC and WebSockets so that I can better understand which scenario calls for what. I am curious about the broad idea of two parties (mainly web based, but potentially one being a dedicated server application) talking to each other.

Assumption:

• Clearly in regards to ad-hoc networks, WebRTC wins as it natively supports the ICE protocol/method.

Questions:

• Regarding direct communication between two known parties in-browser, if I am not relying on sending multimedia data, and I am only interested in sending integer data, does WebRTC give me any advantages over webSockets other than data encryption?
• Regarding a dedicated server speaking to a browser based client, which platform gives me an advantage? I would need to code a WebRTC server (is this possible out of browser?), or I would need to code a WebSocket server (a quick google search makes me think this is possible).

Answer 1

There is one significant difference: WebSockets works via TCP, WebRTC works via UDP. In fact, WebRTC is SRTP protocol with some additional features like STUN, ICE, DTLS etc. and internal VoIP features such as Adaptive Jitter Buffer, AEC, AGC etc.

So, WebSockets is designed for reliable communication. It is a good choice if you want to send any data that must be sent reliably.

When you use WebRTC, the transmitted stream is unreliable. Some packets can get lost in the network. It is bad if you send critical data, for example for financial processing, the same issue is ideally suitable when you send audio or video stream where some frames can be lost without any noticeable quality issues.

If you want to send data channel via WebRTC, you should have some forward error correction algorithm to restore data if a data frame was lost in the network.