WebRTC - scalable live stream broadcasting / multicasting

Question

PROBLEM:

WebRTC gives us peer-to-peer video/audio connections. It is perfect for p2p calls, hangouts. But what about broadcasting (one-to-many, for example, 1-to-10000)?

Lets say we have a broadcaster "B" and two attendees "A1", "A2". Of course it seems to be solvable: we just connect B with A1 and then B with A2. So B sends video/audio stream directly to A1 and another stream to A2. B sends streams twice.

Now lets imagine there are 10000 attendees: A1, A2, ..., A10000. It means B must send 10000 streams. Each stream is ~40KB/s which means B needs 400MB/s outgoing internet speed to maintain this broadcast. Unacceptable.

ORIGINAL QUESTION (OBSOLETE)

Is it possible somehow to solve this, so B sends only one stream on some server and attendees just pull this stream from this server? Yes, this means the outgoing speed on this server must be high, but I can maintain it.

Or maybe this means ruining WebRTC idea?

NOTES

Flash is not working for my needs as per poor UX for end customers.

SOLUTION (NOT REALLY)

26.05.2015 - There is no such a solution for scalable broadcasting for WebRTC at the moment, where you do not use media-servers at all. There are server-side solutions as well as hybrid (p2p + server-side depending on different conditions) on the market.

There are some promising techs though like https://github.com/muaz-khan/WebRTC-Scalable-Broadcast but they need to answer those possible issues: latency, overall network connection stability, scalability formula (they are not infinite-scalable probably).

SUGGESTIONS

1. Decrease CPU/Bandwidth by tweaking both audio and video codecs;
2. Get a media server.

Solution 1

As it was pretty much covered here, what you are trying to do here is not possible with plain, old-fashionned WebRTC (strictly peer-to-peer). Because as it was said earlier, WebRTC connections renegotiate encryption keys to encrypt data, for each session. So your broadcaster (B) will indeed need to upload its stream as many times as there are attendees.

However, there is a quite simple solution, which works very well: I have tested it, it is called a WebRTC gateway. Janus is a good example. It is completely open source (github repo here).

This works as follows: your broadcaster contacts the gateway (Janus) which speaks WebRTC. So there is a key negotiation: B transmits securely (encrypted streams) to Janus.

Now, when attendees connect, they connect to Janus, again: WebRTC negotiation, secured keys, etc. From now on, Janus will emit back the streams to each attendees.

This works well because the broadcaster (B) only uploads its stream once, to Janus. Now Janus decodes the data using its own key and have access to the raw data (that it, RTP packets) and can emit back those packets to each attendee (Janus takes care of encryption for you). And since you put Janus on a server, it has a great upload bandwidth, so you will be able to stream to many peer.

So yes, it does involve a server, but that server speaks WebRTC, and you "own" it: you implement the Janus part so you don't have to worry about data corruption or man in the middle. Well unless your server is compromised, of course. But there is so much you can do.

To show you how easy it is to use, in Janus, you have a function called incoming_rtp() (and incoming_rtcp()) that you can call, which gives you a pointer to the rt(c)p packets. You can then send it to each attendee (they are stored in sessions that Janus makes very easy to use). Look here for one implementation of the incoming_rtp() function, a couple of lines below you can see how to transmit the packets to all attendees and here you can see the actual function to relay an rtp packet.

It all works pretty well, the documentation is fairly easy to read and understand. I suggest you start with the "echotest" example, it is the simplest and you can understand the inner workings of Janus. I suggest you edit the echo test file to make your own, because there is a lot of redundant code to write, so you might as well start from a complete file.

Have fun! Hope I helped.

Solution 2

As @MuazKhan noted above:

https://github.com/muaz-khan/WebRTC-Scalable-Broadcast

works in chrome, and no audio-broadcast yet, but it seems to be a 1st Solution.

A Scalable WebRTC peer-to-peer broadcasting demo.

This module simply initializes socket.io and configures it in a way that single broadcast can be relayed over unlimited users without any bandwidth/CPU usage issues. Everything happens peer-to-peer!

This should definitely be possible to complete.
Others are also able to achieve this: http://www.streamroot.io/

Solution 3

AFAIK the only current implementation of this that is relevant and mature is Adobe Flash Player, which has supported p2p multicast for peer to peer video broadcasting since version 10.1.

http://tomkrcha.com/?p=1526.

Solution 4

"Scalable" broadcasting is not possible on the Internet, because the IP UDP multicasting is not allowed there. But in theory it's possible on a LAN.
The problem with Websockets is that you don't have access to RAW UDP by design and it won't be allowed.
The problem with WebRTC is that it's data channels use a form of SRTP, where each session has own encryption key. So unless somebody "invents" or an API allows a way to share one session key between all clients, the multicast is useless.

Solution 5

There is the solution of peer-assisted delivery, meaning the approach is hybrid. Both server and peers help distribute the resource. That's the approach peer5.com and peercdn.com have taken.

If we're talking specifically about live broadcast it'll look something like this:

1. Broadcaster sends the live video to a server.
2. The server saves the video (usually also transcodes it to all the relevant formats).
3. A metadata about this live stream is being created, compatible with HLS or HDS or MPEG_DASH
4. Consumers browse to the relevant live stream there the player gets the metadata and knows which chunks of the video to get next.
5. At the same time the consumer is being connected to other consumers (via WebRTC)
6. Then the player downloads the relevant chunk either directly from the server or from peers.

Following such a model can save up to ~90% of the server's bandwidth depending on bitrate of the live stream and the collaborative uplink of the viewers.

disclaimer: the author is working at Peer5

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Comments

[user2003356]

What about the stream speed in server side solution? Please share.

[user2003356]

Server side solution means? What did you used? It would be helpful for my research. Please share. Thanks.

[igorpavlov]

Server side solution means Opentok by Tokbox. I do not advertize them, there are tons of such solutions on the market, but I stick with this one. It is working like a media server. P.S. What do you mean by multi-party communication? I don't get it.

[igorpavlov]

Thank you. I do know about peer5 and find it a pretty attractive solution. However, the purpose of this question was to find a solution absolutely server-less (only STUN/TURN allowed).

[igorpavlov]

Do you use any kind of server side software kinda MCU?

[flavioribeiro]

I'm actually using some server side componentes from rtcio people: github.com/rtc-io

[igorpavlov]

It looks like you use their components for signaling. How about server side video streaming? Or you solution is absolutely P2P?

[flavioribeiro]

sorry for the long delay in answering you @igorpavlov, I'm using EvoStream to segment the videos and I'm looping a video source and pointing to EvoStream using Elemental encoder.

[igorpavlov]

It is a media servers provider. More efficient? Probably. Is it what I am looking for? No.

[Ramil Amerzyanov]

@igorpavlov could you give list of companies who provide server side webrtc? I know only Flashphoner and Opentok. Thanks

[igorpavlov]

People do not kill the technology. The technology is killing itself by providing very poor UX, especially when allowing mic/camera. That's where getusermedia wins.

[Benjamin Trent]

I would be curious to see if this would actually scale. There are sure to be scaling issues with latency on HUGE groups(1000+) but if there are only 5-10 I would imagine it would work quite nicely but some fancy foot work would be needed if somebody in the middle of the peer "chain" leaves and reconnecting all the subsequent peers if it is just a single chain would be a HUGE over head. May be better to use a binary/ternary tree structure.

[igorpavlov]

This architecture works in case a) absence of latency, b) stable and fast internet connection, c) minimum connection time. So it definitely works via LAN. Can be helpful for broadcasting realtime video/audio through your company employees if they sit in the same building, for example.

[igorpavlov]

Moreover, the architecture, where every participant rebroadcasts to 2 people and each participant receives streams from 2 people as well reduces the problem with reconnections in case of peer disconnection from chain. B (broadcaster) sends to A1 (attendee 1) and A2. A1 sends to A2 and A3. A2 sends to A3 and A4. If lets say A2 disconnects, then A3 still receives from A1 (so you need just quickly switch streams), A4 still receives from A3, then the whole chain architecture is being recovered in the background and it almost solves the connection time issue.

[Tracker1]

And A1's computer crashes, meaning everyone downstream is SOL.

[igorpavlov]

Not exactly true. A2 will swap to B, A3 to A2 and B, A4 to A3 and A2 etc. And you can process 2 streams to see which one is the best quality. But With every peer the time lag will increase as the video transmission is not really "real-time". And this method relies on the assumption that at least 50% of peers in average have good connection (which is statistically not true)

[Ben]

Is it true to say this doesn't work in Safari currently (or any browser that doesn't support WebRTC?). Does anyone know of a hybrid solution where you broadcast from the Browser to server using WebRTC then transcode the video to HLS/HDS (Or even RTMP) to fit into a traditional broadcast system?

[rubo77]

but chats already work with WebRTC, so everyone sees every message, so one-to many should work for video also somehow

[igorpavlov]

This stuff seems a little bit outofdate for me. Any updates or thoughts on this idea?

[igorpavlov]

Also, does it solve latency problems anyhow? For example, Peer1 talks to Peer5 and Peer2 eventually loses connection. Or is this architecture good for LAN only?

[igorpavlov]

Also, is Streamroot similar to Peer5?

[rubo77]

Apart from the bad ux, would this be the solution? Server less?

[nschoe]

@Ben yes it doesn't work with browsers that don't support WebRTC. Back in the days (when I write this) Safari was clearly not supporting this. Today however I have not checked. But I still think they don't support WebRTC (to be confirmed, though). As for using it in a hybrid system, this is totally possible, actually this is what I did for the company I worked at ; as you said, I broadcast from the browser to the server and from there, I built and plugged a GStreamer pipeline to transcode the feed.You can do this too!

[ishandutta2007]

any idea about jitsi ? is jitisi also the same ?

[Navigateur]

@nschoe Is this any better than using websocket to do the same?

[Dirk V]

You're actually explaining how an SFU (Selective Forwarding Unit) works. You can do the same with mediasoup

[Dirk V]

@rubo77 The data sent with text message is nothing at all compared to the rate and amount of data sent with video streams. So chats can easily contain waayy more users at once

[Muhammad Umer]

what are some other alternatives, what do you google?