My primary intention is to setup a VoIP session between 2 users A & B; Here the raw audio / video media bytes are fetched from A's browser are played in B's browser and vice versa.
The reason is that, when the user C & D are added into this call, we need not have to create a P2P mesh network which limits the performance.
Tried recording media with getUserMedia() and playback, but it is not real time. It also gives a bad user experience. (However, haven't experimented yet with videos of small chunks as 200 ms)
Is there any approach where I can get the raw bytes of the media and play it on other browser? Currently I have a server in between which can connect to both peers if required.
Any online examples or libraries are welcome.
Have already asked 2 questions in this regard with 100-100 bounties, but not much of use:
- How to use libsrtp or similar library to decrypt/encrypt the WebRTC data stream?
- How to integrate part of WebRTC as a static / dynamic library with the existing C++ code?
Related: How to stream, live video playing on my browser to browser of another user?
If i understand you well is you're looking on how to have more than two users on the session right? without using mesh topology
thats possible and configurable as well by means that some maybe active speaker or everyone is active speaker not only receiver whatever configuration you choose but to me it seems that you're asking for video conferencing there are couple of tools for this the best one i might recommend is
mediasoupits a SFU as selective fowarding unit mediasoup