srt VS FirebaseRTC

Is receiver side also using mobile internet? You could try to setup this on separate server and both sender and receiver connnect to it. https://github.com/Haivision/srt/blob/master/docs/apps/srt-live-transmit.md

Here is one example how to feed SRT stream to ffmpeg. Try streaming directly to this via OBS first and confirm it works before stacking more software between endpoints. https://github.com/Haivision/srt/issues/506

SRT, or Secure Reliable Transport, is a type of streaming protocol that provides enhanced security and reliability for video streaming. SRT is becoming increasingly popular among broadcasters and streamers including industry stalwarts such as ESPN because of its ability to deliver high-quality content over challenging network conditions and for its ability to make contribution and stream ingestion easy. SRT streams provide improved security, low latency, and flexibility and is supported by a global community of developers all contributing to the open-source project. Because of the power of SRT streams, Dolby.io Real-Time Streaming has decided to launch support with an SRT open beta program.

https://github.com/Haivision/srt/discussions/2220 I think this is probably the basis of what i did...

If it turns out to be an SRT/socket issue (rather than an audio issue) you can pump FFmpeg into https://github.com/Haivision/srt srt-live-transmit, and let that do the SRT. It is a very long time since I played with srt-live-transmit, but it did behave differently to both FFmpeg and TSDuck's SRT impementations when there were input issues. Obviously, putting more components in the chain may be disadvantageous, but component abstraction can help when debugging.

SRT is an open-source protocol developed by streaming technology provider Haivision. Known for its security, reliability, compatibility, and low-latency streaming it is the preferred protocol for members of the SRT Alliance. This protocol does not rely on a single codec, which allows developers to pair it with any audio and video codecs they desire.

I've also looked at SRT, which to all intents and purposes ought to work, but doesn't?

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