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Mediasoup – WebRTC Video Conferencing

mediasoup.org

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Re: Mediasoup – WebRTC Video Conferencing

#31

Post by ibc »

sandGorgon wrote: post #6

any comparisons with jitsi ? really would like to understand how these technologies stack against each other. jitsi has been production tested far longer i suppose, through its freely available videoconferencing service https://meet.jit.si/

mediasoup co-author here.

Comparing Jitsi with mediasoup is like comparing Netflix (backend + apps) with Express.js + libcurl.

Jitsi developers may replace their RTC core internals (including the SFU) with mediasoup + mediasoup-client and you wouldn't even realize of it. Hope this helps.

Re: Mediasoup – WebRTC Video Conferencing

#32

Post by ibc »

fulafel wrote: post #11
supermatt wrote: post #9

Earlier quoted context omitted.

Yes, that is the "full mesh" that I mention in my comment... This platform - and jitsi/janus/zoom/whatever - all use an SFU for more than a handful of participants.

But thats very doable. And no need for the unaccelerated ffmpeg wasm afaics, and no need to use a "sfu". I don't know what Mediasoup does, was just commenting on what webrtc can do re the "the current state of webrtc" subject.

mediasoup co-author here.

So you don't know what mediasoup does but you assure that "no need to use a SFU". Too much free time to comment maybe?

e2ee makes sense in one-to-one and N-to-N scenarios. It's just more complex in N-to-N scenarios because you DO need a centralized server so each participant just sends its audio/video once (to the server) and the server distributes it to others given whichever application policy/logic. That's a SFU, and that's what mediasoup does.

Re: Mediasoup – WebRTC Video Conferencing

#33

Post by fulafel »

ibc wrote: post #32
fulafel wrote: post #11

Earlier quoted context omitted.

But thats very doable. And no need for the unaccelerated ffmpeg wasm afaics, and no need to use a "sfu". I don't know what Mediasoup does, was just commenting on what webrtc can do re the "the current state of webrtc" subject.

mediasoup co-author here. So you don't know what mediasoup does but you assure that "no need to use a SFU". Too much free time to comment maybe? e2ee makes sense in one-to-one and N-to-N scenarios. It's just more complex in N-to-N scenarios because you DO need a centralized server so each participant just sends its audio/video once (to the server) and the server distributes it to others given whichever application po…

We may be talking past each other, I'm talking about webrtc in general as the subthread was discussinh that. Think about a limited user count screen sharing or vr app with a/v comms built in for example.

Re: Mediasoup – WebRTC Video Conferencing

#34

Post by imtringued »

Can this be used as an alternative to Kurento? I use an ffmpeg command to send an RTP stream to Kurento which then broadcasts that to all clients that connect via WebRTC. I've had to rewrite the code that interacts with Kurento twice already. It doesn't feel like a rock solid solution. e.g. when you accidentally switch audio and video ports in the ffmpeg command the entire kurento instance crashes. I can't expose Kurento to untrusted instances otherwise there is a risk of DOS. I don't feel like running a whole Kurento instance per ffmpeg command is a sustainable solution.

It might not be obvious but I am not using WebRTC for video conferencing. My use case is basically regular live streaming but with latencies in the 100ms range.

Edit: I just compared the docs https://raw.githubusercontent.com/versatica/mediasoup/v3/art... vs https://doc-kurento.readthedocs.io/en/6.13.0/features/kurent... This is exactly what I need!

Re: Mediasoup – WebRTC Video Conferencing

#35

Post by manthideaal »

To understand the terms: webrtc, stun, turn, mesh, sfu, mcu, ice and trickle ice, there is (1). 15 minutes to understand what is all this about. What about IPv6 stun and turn?, it seems other people asked the same idea I thought: (2) bout all of this, one is the answer is: As IPv6 takes over the complexity of new networks, STUN and ICE will become irrelevant. I think that with the surge in video conferences and rtc, ipv6 with take off.

In my very humble opinion, I would suggest to reserve some address space in IPv6 for rtc, so that a peer is able to adopt a new special ip reserved for rtc. Nothing new under the sun, in 2014 someone comment along this line of thought (2) and (3).

So what are we waiting for?

(1) https://webrtcglossary.com/ (2) https://www.quora.com/Will-the-IPv6-result-in-the-death-of-S...

(3) 2014, AshleysBrain, https://news.ycombinator.com/item?id=7496986 I think the solution is IPv6. Once every device on the Internet is uniquely addressable again, we can do away with these NAT hacks and two endpoints should be able to reliably connect to each other again, no matter where they are. Of course, that's assuming we don't get more short-sighted engineering that breaks things again...

Re: Mediasoup – WebRTC Video Conferencing

#37

Post by ibc »

Hi, mediasoup co-author here.

TL'DR': Pornhub uses mediasoup.

I've read many comments here asking about "how mediasoup is different than XXX" or about "mobile apps". I think the Overview in the website should be self explanatory, I'll just paste a fragment here:

https://mediasoup.org/documentation/overview/

--------------------- Design goals of mediasoup and its client side libraries:

- Be a SFU (Selective Forwarding Unit). - Support both WebRTC and plain RTP input and output. - Be a Node.js module in server side. - Be a tiny JavaScript and C++ libraries in client side. - Be minimalist: just handle the media layer. - Be signaling agnostic: do not mandate any signaling protocol. - Be super low level API. - Support all existing WebRTC endpoints. - Enable integration with well known multimedia libraries/tools.

Use cases:

- Group video chat applications. - One-to-many (or few-to-many) broadcasting applications in real-time. - RTP streaming.

Re: Mediasoup – WebRTC Video Conferencing

#38

Post by ibc »

pkstn wrote: post #36

Does it do TURN/STUN?

mediasoup is a SFU that must be deployed in a reachable server, so STUN is not needed at all. You may need a TURN server if a client has a restrictive firewall that blocks UDP. mediasoup is not a TURN server but you can deploy a TURN server (i.e. coturn) in your backend.

Re: Mediasoup – WebRTC Video Conferencing

#39

Post by bryanrasmussen »

ibc wrote: post #37

Hi, mediasoup co-author here. TL'DR': Pornhub uses mediasoup. I've read many comments here asking about "how mediasoup is different than XXX" or about "mobile apps". I think the Overview in the website should be self explanatory, I'll just paste a fragment here: https://mediasoup.org/documentation/overview/ --------------------- Design goals of mediasoup and its client side libraries: - Be a SFU (Selective Forwarding…

I read the whole thing. I am really confused as to why out of that you chose "Pornhub uses mediasoup" as the TLDR?

Re: Mediasoup – WebRTC Video Conferencing

#40

Post by supermatt »

manthideaal wrote: post #35

To understand the terms: webrtc, stun, turn, mesh, sfu, mcu, ice and trickle ice, there is (1). 15 minutes to understand what is all this about. What about IPv6 stun and turn?, it seems other people asked the same idea I thought: (2) bout all of this, one is the answer is: As IPv6 takes over the complexity of new networks, STUN and ICE will become irrelevant. I think that with the surge in video conferences and rtc,…

addressable !== routeable !== reachable.

IPv6 will certainly REDUCE the need for STUN, but there are still (many) cases where you don't want to be "reachable by default", in which case you need a stable reference for negotiating routing and reachability (e.g. STUN).

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