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What's the minimum latency broadcasting one WebRTC stream to many clients real-time?

Can you share more details about the pipeline generating the stream? There're many factors that can increase latency.

Well so far it's a setup where the broadcaster is an iPhone and the listeners are iPhones as well. Strictly one-way. Using the default iOS and WebRTC libraries. The critical factor is that all listeners get the same data at the same time with as little latency as possible. I have been experimenting with a one-to-many setup using Janus, but I'm not sure how well that will scale up.

Each listener iphone will almost never get each stream chunk at the same millisecond, but it can be close enough most of the time, depending on your use case.

I understand that after sending the packet it still has to deal with network speeds, but the idea is that there is no buffer just mirroring all of incoming video real-time.

My experience with Janus is that it will not scale at least for the sole reason of using poll not epoll for the I/O (http://www.kegel.com/c10k.html#nb.poll).

When I tried to use it for the last time (1-2 years ago) it also had severe stability issues but maybe that improved over time.

Maybe drop me an email at info@membraneframework.org so we can discuss the matter in greater detail?

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