FirebaseRTC VS nginx-rtmp-module

You might also consider assessing complementary or alternative technologies; WebSocket and HTTP aren’t the only options when it comes to real-time communication, after all. WebRTC is similar to WebSocket, with the key difference being that it’s used to implement peer-to-peer connections without relying on a server. That can be especially helpful for video calls, allowing participants to communicate directly without introducing load to your server.

We use WebRTC to gain access to a user’s camera and microphone using the getUserMedia method. Typically, I would gain access to both of these from the same call. However, our experience requires the camera to flip from facing the environment to facing the user and I noticed that the small period of time the flip occurred (and microphone wasn’t available) contributed to a bit of audio lagging in the final recorded video. This was one of the nastier bugs I faced in development. So, we’ll just access each of these on their own media streams so that the camera can flip independently from the microphone.

Low latency streaming (<500ms): The Video SDK's infrastructure is built with WebbRTC, which helps to deliver secure and ultra-low latency video streams to all audiences at different bandwidths.

Web Real-Time Communication (WebRTC) is a technology developed by Google in 2013 for peer-to-peer communication. WebRTC enables web browsers to capture audio, video, exchange data, and teleconferencing without plugins or intermediaries. WebRTC achieve these through APIs and protocols that interact with one another. WebRTC media streaming when used with SocKet.IO will produce an application that streams media and exchange data instantly. Socket.IO is a library that provides low latency bi-directional communication between client and server. Socket.IO was built on websocket, a communication protocol that provides a full-duplex and low latency communication between server and browser. In this article, readers will learn how to build a video chat application using WebRTC and Socket.IO. This article is for web developers who wish to develop web applications that can stream media between two peers of computers in real-time without installing any plugins.

Possibly you what to look into WebRTC: https://webrtc.org/

Use the language you're comfortable with. Sounds like you're interested in creating a blockchain game. Writing your own simple game engine isn't simple. I would recommend utilizing an existing one for whatever language you want. If you still choose to write your own it can be a valuable lesson in graphical programming which I personally find fun. It's easier to cheat a webpage embedded game written in Javascript than one ported to WebASM in my experience and I've heard good things about WebRTC for embedded multiplayer games.

Have you looked into WebRTC? https://webrtc.org Seems like it supports exactly what you're looking for. SignalR is more for real-time messages, not really for streaming.

P2P is nothing new. It is a long-established means of connecting two or more people directly over a network. Web browsers are very capable of a wide range of P2P connections. Many apps use WebRTC to enhance realtime apps, but it is still an underutilized technology. Even with WebRTC, many apps are designed around the dependence on a central app server with WebRTC performing a user experience enhancement. Web meshes turn this idea on its head: Instead of using P2P connections to enhance the user experience, what if P2P connections were the foundation of the user experience? In other words, what if there was no central server?

The good news is that after reading all I have a better understanding of the Web Realtime Communicate and the big view, not just about small things like sending files. You can read all about WebRTC here

Regarding the first solution, I've looked through some of the source code for the NGINX RTMP Module, and I believe the code I would need to modify lives in the ngx_rtmp_cmd_module.c file: https://github.com/arut/nginx-rtmp-module/blob/master/ngx_rtmp_cmd_module.c

You could look at setting up the RTMP module on nginx and securing it with lets encrypt - https://github.com/arut/nginx-rtmp-module

foobar2000 -> OBS -> nginx server running nginx-rtmp-module

If stuck for a server I've used this in the past. nginx with an RTMP module.

Also linking a config file I commented. Unsure if it'll still work and you can definitely simplify it for OP's needs. It was a proof of concept for a streaming service I was thinking up before realising how much money I don't have, haha

https://github.com/arut/nginx-rtmp-module

https://gist.github.com/cohan/7f676d3f561be62d0550785c015f00...

If you can only output something like RTMP, then your options are a bit more limited. Set up an RTMP server somewhere (e.g. https://github.com/arut/nginx-rtmp-module), and connect with another client that will soak up the RTMP into a file - I think ffmpeg should be able to do that, but I've had the most luck with GStreamer. Again, avoid transcoding it on the way into a file, both ffmpeg and GStreamer can be set up to never touch the encoded media (just copy/pass it along).

Are you using the rtmp nginx module?

FROM buildpack-deps:stretch ENV NGINX_VERSION nginx-1.18.0 ENV NGINX_RTMP_MODULE_VERSION 1.2.1 RUN apt-get update && \ apt-get install -y ca-certificates openssl libssl-dev && \ rm -rf /var/lib/apt/lists/* RUN mkdir -p /tmp/build/nginx && \ cd /tmp/build/nginx && \ wget -O ${NGINX_VERSION}.tar.gz https://nginx.org/download/${NGINX_VERSION}.tar.gz && \ tar -zxf ${NGINX_VERSION}.tar.gz RUN mkdir -p /tmp/build/nginx-rtmp-module && \ cd /tmp/build/nginx-rtmp-module && \ wget -O nginx-rtmp-module-${NGINX_RTMP_MODULE_VERSION}.tar.gz https://github.com/arut/nginx-rtmp-module/archive/v${NGINX_RTMP_MODULE_VERSION}.tar.gz && \ tar -zxf nginx-rtmp-module-${NGINX_RTMP_MODULE_VERSION}.tar.gz && \ cd nginx-rtmp-module-${NGINX_RTMP_MODULE_VERSION} RUN cd /tmp/build/nginx/${NGINX_VERSION} && \ ./configure \ --sbin-path=/usr/local/sbin/nginx \ --conf-path=/etc/nginx/nginx.conf \ --error-log-path=/var/log/nginx/error.log \ --pid-path=/var/run/nginx/nginx.pid \ --lock-path=/var/lock/nginx/nginx.lock \ --http-log-path=/var/log/nginx/access.log \ --http-client-body-temp-path=/tmp/nginx-client-body \ --with-http_ssl_module \ --with-threads \ --with-ipv6 \ --add-module=/tmp/build/nginx-rtmp-module/nginx-rtmp-module-${NGINX_RTMP_MODULE_VERSION} && \ make -j $(getconf _NPROCESSORS_ONLN) && \ make install && \ mkdir /var/lock/nginx && \ rm -rf /tmp/build RUN ln -sf /dev/stdout /var/log/nginx/access.log && \ ln -sf /dev/stderr /var/log/nginx/error.log COPY nginx.conf /etc/nginx/nginx.conf EXPOSE 1935 CMD ["nginx", "-g", "daemon off;"]

But, if it's something you want to do, and want to do it inexpensively you want to use Nginx-RTMP - http://nginx-rtmp.blogspot.com/ - this is a free version of AntMedia Server without the $70 per month fee.