Is WebRTC right for us?

Question

RTMP

We organize a webinar service for internal tasks. It is necessary to implement the possibility of creating live broadcasts by different users. Is this possible with WevRTC? Is it possible to record a WebRTC signal (audio, video, screen sharing) for later playback. What is the best solution to choose as a media server? Does WebRTC have low translation latency?
Where can I read more about this in Russian?

Answer 1

It is quite suitable for webinars, for videoconferencing - a lot of hemorrhoids with logic.
If you need a record, then we forget about p2p.
It remains either fluxonic or vovza.
The first one is better, because there is support in Russian.
Read here , and here is an example of a publication page.
In short, the web application from the example initiates a connection to the server, captures video and audio from the browser, and starts pouring video into the socket. This requires a secure https connection from the servers, so take care of the certificate.
And then on the server you can already turn the stream as you like - record, transcode, distribute as you like.
Today, there are two common ways to stream video in a browser with low latency:
1. RTMP - good old flash
2. WebRTC - young, raw and nuanced protocol and transport
3. all kinds of exotic, for example, flussonic has its own tricky implementation of the player and protocol, which allows segments (!) To play with low latency and while the browser does not become bad. Set as proto=mse
Under ideal conditions, WebRTC gives a delay of about 200-300ms, in real conditions - about 500ms.
RTMP - up to a second.

Answer 2

Asterisk is good at video and webrtc. The internet is full of documentation.