I am wondering what the benefit of a WebRTC based P2P system would be?
Is the performance improved?
Is the security stronger?
... as opposed to a traditional vanilla P2P solution in, fx, Java.
The benefit of WebRTC/RTCWeb (first is the implementation and w3c API, second is the ietf standard) is in its name: Web. It does not have anything special, it's just an standard inspired by SIP whose implementation is shared by all the main browsers. The benefit is that you don't have to code a client for it because it's already embedded in the browser next to GetUserMedia which enables you to easily capture video and audio streams from the devices.
Performance is determined mostly by the codecs and they use open source codecs you could use in your own implementation. It is also secure, but you can make any p2p connection secure through encryption.
It is just an standard meant to make web developers' lifes easier when dealing with p2p real time voice, video and data streaming.
Related
I'm using WebRTC in a sort of non-conventional way.
I have multiple streams generated by several 'broadcasting' peers being sent to a collection of several 'receiving' peer.
I intend to use an SFU media server (maybe Jitsi or Kurento)
It is very critical that these streams are presented at the receiving peers in a synchronized fashion.
What are the methods I can use for synchronization? Usually this isn't an issue with WebRTC because there is not usually a consistent clock between peers, but in my case there is a common clock for all the stream sources.
The only ways I can imagine doing it are:
Not worry about it and hope that WebRTC's low latency will cause everything to be in sync.
Somehow encoding timestamp metadata in the WebRTC stream frames, and somehow synchronizing display with javascript in the browser.
Using a tool like GStreamer that can perform video synchronization, mix the streams into a single stream and forward that to the media server (and thus to the receiving clients). I don't have a good idea of how I would actually perform the synchronization though.
Any thoughts and advice would be appreciated.
The only OTT system capable of synchronisation of low latency streams available (when writing this text), is the SYE system made by Net Insight. They are able to synchronise any device down to single digit millisecond in low latency mode.
They do not provide any open source that I know of but you can check it out by downloading a app that uses it.
Primetime
The game starts 20:00 CET every day, download it on several phones/tablets to verify the sync part.
However there are other systems that can synchronise playback that I found.
HibbTV
HibbTV seams to focus on more IPTV replacement solutions as I interpret the solution. They do not seam to target the wild west of internet. I might be wrong please correct me then.
W3C MULTI-DEVICE TIMING COMMUNITY GROUP
Spoke to the researchers a while back. They can synchronise playback but they target collaborative viewing. The low latency part is not part of the scope as I understand it.
Then when it comes to WebRTC, LHLS, MPEG-DASH CMAF and all other solutions they have no sense of time so it will not be possible to render the same video frame on different devices using various access technologies such as 4G, WiFi or cable or even if the devices uses the same technology because the rendering is buffer controlled not time controlled.
/Anders
I am trying to build a website and mobile app (iOS, Android) for the internet radio station.
Website users broadcast their music or radio and mobile users will just listen radio stations and chat with other listeners.
I searched a week and make a prototype with Wowza engine (using HLS and RTMP) and SHOUTcast server on Amazon EC2.
Using HLS has a delay with 5 seconds, but RTMP and SHOUTcast has 2 second delay.
With this result I think I should choose RTMP or SHOUTcast.
But I am not sure RTMP and SHOUTcast are the best protocol. :(
What protocol should I choose?
Do I need to provide a various protocol to cover all platform?
This is a very broad question. Let's start with the distribution protocol.
Streaming Protocol
HLS has the advantage of allowing users to get the stream in the bitrate that is best for their connection. Clients can scale up/down seamlessly without stopping playback. This is particularly important for video, but for audio even mobile clients are capable of playing 128kbit streams in most areas. If you intend to have a variety of bitrates available and want to change quality mid-stream, then HLS is a good protocol for you.
The downside of HLS is compatibility. iOS supports it, but that's about it. Android has HLS support but it is still buggy. (Maybe in another year or two once all the Android 3.0 folks are gone, this won't be as much of an issue.) JWPlayer has some hacks to make HLS work in Flash for desktop users.
I wouldn't bother with RTMP unless you're only concerned with Flash users.
Pure progressive streaming with HTTP is the route I almost always choose to go. Everything can play it. (Even my Palm Pilot's default media player from 12 years ago.) It's simple to implement and well understood.
SHOUTcast is effectively HTTP, but a poorly implemented version that has compatibility issues, particularly on mobile devices. It has a non-standard status line in its response which breaks a lot of clients. Icecast is a good alternative, and is what I would recommend for production use today. As another option, I have created my own streaming service called AudioPump which is HTTP as well, and has been specifically built to fix compatibility with oddball mobile clients, such as native Android players on old hardware. It isn't generally available yet, but you can contact me at brad#audiopump.co if you want to try it.
Latency
You mentioned a latency of 2 seconds being desirable. If you're getting 2-second latency with SHOUTcast, something is wrong. You don't want latency that low, particularly if you're streaming to mobile clients. I usually start with a 20-second buffer at a minimum, which is flushed to the client as fast as it can receive it. This enables immediate starting of the stream playback (as it fills up the client-side buffer so it can begin decoding) while providing some protection against buffer underruns due to network conditions. It's not uncommon for mobile users to walk around the corner of a building and lose their nice signal quality. You want your stream to survive that as best as possible, so if you have already sent the data to cover the drop out, the user doesn't have to know or care that their connection became mediocre for a short period of time.
If you do require low latency, you're looking at the wrong technology entirely. For low latency, check out WebRTC.
You certainly can tweak your traditional internet radio setup to reduce latency, but rarely is that a good idea.
Codec
Codec choice is what will dictate your compatibility more than anything else. MP3 is easily the most compatible, and AAC isn't far behind. If you go with AAC, you get better quality audio for a given bitrate. Most folks use this to reduce their bandwidth bill.
There are licensing fees with MP3, and there may be with AAC depending on what you're using for a codec. Check with a lawyer. I am not one, and the licensing is extremely complicated.
Other codecs include Vorbis and Opus. If you can use Opus, do so as the licensing is wide open and you get good quality for the bandwidth. Client compatibility here though is the killer of Opus. (Maybe in a few years it will be better.) Vorbis is a mediocre codec, but is free and clear.
On the extreme end, I have some stations doing their streaming in FLAC. This is lossless audio quality, but you're paying for 8x the bandwidth as you would with a medium quality MP3 station. FLAC over HTTP streaming compatibility is not code at the moment, but it works alright in VLC.
It is very common to support multiple codecs for your streams. Depending on your budget, if you can't do that, you're best off with MP3.
Finally on encoding, don't go from a lossy codec to another lossy codec if you can help it. Try to get the output stream as close to the input as possible. If you re-encode audio, you lose quality every time.
Recording from Browser
You mentioned users streaming from a browser. I built something like this a couple years ago with the Web Audio API where the audio is captured and then encoded and sent off to Icecast/SHOUTcast servers. Check it out here: http://demo.audiopump.co:3000/ A brief explanation of how it works is here: https://stackoverflow.com/a/20850467/362536
Anyway, I hope this helps you get started.
Streaming straight audio/mpeg (mp3 packets) has worked everywhere I've tried.
If you are developing an APP then go with AAC, if you are simply playing via web browser then you need a HTML5 Implimentation which is MP3. All custom protocols like RTMP or SHOUTcast requires additional UI to be built. There are some third party players available in open source world. You can either use them or stick to HTML5 MP3/OGG as most people now days are using chrome browser or other HTML5 complaint browsers.
I need to know if TideKit will be able to stream live video and audio from device cameras and microphones. The Android and IOS APIs allow for this. I think Flex can do it. I asked about this on the Twitter page but I'm looking for a more definitive answer. The one I got was "TideKit is a development, not a streaming platform but you could develop an app for that! That’s where TideKit comes into play" which doesn't fully answer the question.
The goal is to stream video from Android & IOS cameras and audio from the device microphones to a media streaming server such as Flash Media Server or a Wowza streaming server using either RTMP or HTTP streaming from the app to the server. That or it would work if the stream were sent live in any other way to a server socket and then encoded for redistribution via a streaming server.
They key here though is "live" rather than having to waiting for a video or audio file to become complete before sending it off to the server. I know it's possible with the APIs and I really hope TideKit will be able to do this because no other platform similar to TideKit (and there are MANY) can do this besides Flex. I've poured through countless SDK documents. If TideKit can do this it will attract a lot more customers.
Eagerly awaiting a response,
Thanks
#xendi Thank you for your question. TideKit is an app development platform. You can use it for any type of app development for mobile, desktop and web. We've purposefully kept the core of TideKit small. This is to ensure its core is extremely stable and that most functionality can can come through modules.
Out of the box, TideKit has core AV functionality on all platforms. Extension of this functionality is through TideKit modules that have operating system implementations or from pure JavaScript modules. There are almost 100,000 modules of pure JavaScript functionality now available to you through existing repositories including NPM, Bower and Component that can simply be consumed in CommonJS.
When a TideKit or JavaScript module is installed it offers its APIs. This extends the APIs with those already available. Either way those APIs become available to you in JavaScript.
You already have access to camera with TideKit. The rest is handling the streaming protocol, ie RTSP, RTMP, HTTP etc. So there are a few ways to accomplish what you want with TideKit.
Using a TideKit module that supports the streaming protocols by interacting with its APIs in JavaScript.
Using a pure JavaScript solution from a repository together with TideKit that supports the protocols.
Writing your own TideKit module that ties together with APIs of the operating systems.
Writing the solution in pure javascript using TideKit's camera and network APIs.
TideKit is new and has not yet formally launched. We are currently in a reservations mode. We will be delivering it first to those with reservations and it will be gradually rolled out. Demos are currently being prepared to demonstrate the speed and low barrier to development. When TideKit formally launches, I would check for the availability of modules at that point (for both TideKit and JavaScript implementations). Note that not all possible functionality in TideKit modules will be available with the launch. New modules will be releasing over time.
As an aside, TideKit also supports WebRTC in HTML5 so this could work together with TideKit's other capabilities for interesting possibilities.
I'm looking to write my own multiplayer game server (most likely in nodejs) and I was wondering what protocol I should be using to transfer data? Are Datagrams the norm to send information (i realize they don't confirm delivery like HTTP, that can written on top of the protocol)? Any suggestions of performant proven systems would be a real help.
I guess I'm looking for successful techniques in handling the data transfer quickly and effectively (maintaining state on the server and scaling are a separate issue that I have a solid understanding of).
I'm looking to initially support desktop/mobile games (MacOS, iOS, and Android).
For all your protocol needs take a look at socketIO.
Basically your best solution is to rely on websockets which are TCP sockets. socketIO is just a nice cross-browser compliant abstraction.
Either you use standard long pulling techniques or html5 websockets. There is no access to UDP for browser <-> server.
There is a technology that is called RTMFP that Adobe introduced in the latest version of Flash (Flash 10). It allows you to do P2P connection and transfer data directly from a client to an other client without passing by the server. On top of that, it's using UDP to transfer data. I believe that this was originally designed to do video and audio streaming, but you can use it to pass data around.
However the main downside on this technology is the mobile since most of them don't support Flash. In this case you can use Socket.IO and use the server as a router of information as a fallback.
If you want to build your application in Javascript, you can still use it by bridging the functionnality to Javascript. If you want to take a look at a simple version of a bridge, you can take a look at this github project (I am the author).
As a C++ developer of Massive Multiplayer games for 10 years, I can tell you that most of your more advanced games, such as ones in which I was involved (Legends Of Kesmai, Magic: The Gathering Online, Airwarrior II, AVP, NTN Triva) to name a few, TCP is used for most communication simply because you need an ACK / NACK to be sure the data was received from the client. That is not to say UDP doesn't have it's place. In Legends we wrote the the protocol code to use UDP for out of bandwidth delivery of data which wasn't imperative that it be received in proper packet order and complete. Use UDP when you want to do things like update graphic files in the background while the user is playing, etc. This type of delivery is often used for such purposes and allows your TCP packets to arrive as required by your server.
Is it possible to create a real time game with node.js that requires twitch reflexes. How high is the latency? How low can it realistically go?
It is possible to make a real-time game in node.js as you could with any other language/framework.
The problem here would be what kind of server and client you would use.
Using the http server feature for such game would be a bad idea and very difficult, but you could use the TCP server (now called net server) as you would in any other language.
The client would be on some platform where you can use sockets, like Flash, Java applets or desktop software.
Please notice that even with using a TCP socket server you might have problems with latency for a twitch game, but this is outside the area related to this question and more about games and networking.
PS: You could use web sockets since they should theoretically work like TCP sockets but there isn't yet a good support for them in the current modern browsers.
EDIT:
It seems I haven't explained myself correctly, you can make a browser accessible game like you said, you just need to use a protocol that allows you to quickly send data back and forth in real time.
If you want a "pure" browser game without any third party plugins the only way, like I said before, is using JavaScript with websockets which is not well supported yet by the major browsers. (You could use a Flash bridge and still have your game in JavaScript though.)
Using a third party plugin you have Flash and Java (besides the numerous less known plugins like unity and so on). Both have TCP sockets (not sure about UDP) and can be made to connect to a node.js net server (with some security limitations). Most people would say for you to go with Flash since there is a bigger support but Apple doesn't like it so no Flash in iPhone/iPad/iPod Touch or on other miscellaneous mobile devices (that support Java instead).
So yeah... good luck with this.
EDIT 2:
Websocket support in browsers is now pretty decent so I recommend it for realtime games if you want to use the browser as a client.
HTTP servers are typically optimised for throughput/bandwidth over latency. node.js is unlikely to be an exception, and HTTP is intrinsically poor for low latency anyway due to the structure of the protocol.
One informal benchmark using node.js supports this, showing latencies of hundreds of milliseconds. By comparison most twitch games support latencies of no more than 30 or 40ms, ideally less.
Therefore I'd recommend dropping the twitch aspect if you can't drop HTTP.
It's possible, but it depends on how much data must be transmitted between server and client and how fast (speaking of latency). Take a look at Sousaball by Creationix, for example.
Also, if you plan to use websockets, take a look at Socket.IO library by learnboost. It uses websockets when available and falls back to comet in other cases.