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

Question

So I m looking to build a chat app that will allow video  audio  and text  I spent some time researching into Websockets and WebRTC to decide which to use  Since there are plenty of video and audio apps with WebRTC  this sounds like a reasonable choice  but are there other things I should consider   Feel free to share your thoughts   Things like    Due to being new WebRTC is available only on some browsers  while WebSockets seems to be in more browsers  Scalability - Websockets uses a server for session and WebRTC seems to be p2p  Multiplexing multiple chatrooms - Used in Google  Hangouts  and I m still viewing demo apps on how to implement  Server - Websockets needs RedisSessionStore or RabbitMQ to scale across multiple machines

User · Answer

Comparing websocket and webrtc is unfair.

Websocket is based on top of TCP. Packet's boundary can be detected from header information of a websocket packet unlike tcp.

Typically, webrtc makes use of websocket. The signalling for webrtc is not defined, it is upto the service provider what kind of signalling he wants to use. It may be SIP, HTTP, JSON or any text / binary message.

The signalling messages can be send / received using websocket.

User · Answer

Security is one aspect you missed    With Websockets the data has to go via a central webserver which typically sees all the traffic and can access it   With WebRTC the data is end-to-end encrypted and does not pass through a server  except sometimes TURN servers are needed  but they have no access to the body of the messages they forward    Depending on your application this may or may not matter   If you are sending large amounts of data  the saving in cloud bandwidth costs due to webRTC s P2P architecture may be worth considering too

User · Answer

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

User · Answer

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

User · Answer

Websocket and WebRTC can be used together  Websocket as a signal channel of WebRTC  and webrtc is a video audio text channel  also WebRTC can be in UDP also in TURN relay  TURN relay support TCP HTTP also HTTPS   Many projects use Websocket and WebRTC together

User · Answer

webRTC or websockets  Why not use both   When building a video audio text chat  webRTC is  definitely a good choice since it uses peer to peer technology and once the connection is up and running  you do not need to pass the communication via a server  unless using TURN    When setting up the webRTC communication you have to involve some sort of signaling mechanism  Websockets could be a good choice here  but webRTC is the way to go for the video audio text info  Chat rooms is accomplished in the signaling    But  as you mention  not every browser supports webRTC  so websockets can sometimes be a good fallback for those browsers

User · Answer

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

User · Answer

Webrtc is a part of peer to peer connection  We all know that before creating peer to peer connection  it requires handshaking process to establish peer to peer connection   And websockets play the role of handshaking process

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

Examples related to websocket

Examples related to webrtc