I've been experimenting with WebRTC and its ability to stream video/audio using p2p, however I've found that it's not able to stream any resolutions > 720p. Using getUserMedia() I'm able to capture the local webcam in 1080p, but it seems to fail at encoding/decoding in 1080p (I'm not sure which it is) and the client-side displays the video in 1280x720.
There are a few bug reports regarding this issue: Here and here.
Is there a way I can capture the webcam using getUserMedia(), encode the video myself in 1080p and still use WebRTC for its peer-to-peer capabilities?
Basically I'm trying to achieve the same sort of thing shown in this video. That is, live streaming a video and sharing it with other viewers so there's significantly less server/broadcaster bandwidth used. WebRTC seemed the obvious choice for this.
Are there any alternative solutions, besides waiting for the WebRTC bug to be fixed? Thanks.
This issue was resolved on the tickets you posted:
https://code.google.com/p/webrtc/issues/detail?id=1750
Which points to this url for verification.
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I've doing Javascript programming for some time but it's always been related to data updating, saving, manipulating, etc.
I have no idea how something like an in-browser audio player gets audio (especially live, streaming audio) from the internet and plays it out of my computer speakers.
How does this happen in Javascript?
For example, how does a website deliver live audio to my speakers using Javascript? http://player.streamtheworld.com/liveplayer.php?callsign=WVIEAM
The live audio is not much different from pre-recorded audio... it's just played back as it's received, and when live it's encoded as it's recorded.
In browsers these days, the most basic form of streaming audio is a simple <audio> tag. By changing the src attribute from a file to a stream, you're up and running:
<audio src="http://cdn.audiopump.co/waug/main_mp3_256k" />
The browser doesn't know or care in this case that the audio is a live stream. All it knows is that there's some media data that it's fetching via HTTP, and playing back while it comes in.
If your browser compatibility is good, it would be preferable to use the MediaSource API, giving you more control (such as switching to a different quality stream mid-stream, like in HLS) and ensuring that the browser doesn't try to cache what is effectively an inifinitely sized file.
For example, how does a website deliver live audio to my speakers using Javascript? http://player.streamtheworld.com/liveplayer.php?callsign=WVIEAM
This particular site is ran by Triton Digital, and they still use Flash. Many sites still do this as a holdover from a time when HTML5 audio was not widely supported. There is little reason to do this today.
Other reasons to use Flash include incompatible server protocols. If your streaming server is using RTMP, you're stuck with Flash as browsers don't speak RTMP.
There used to be an issue with streaming AAC in-browser due to browsers not properly handling AAC wrapped in ADTS. (This encapsulation is required for streaming AAC in most situations.) Most browsers have resolved this, but I suspect that this is the reason Triton Digital is still using their Flash solution. By using Flash, they can play AAC/ADTS streams.
So far, I have found red5, but I can't get it to run (no video arrives at the server side), so I was looking for a flash-based getUserMedia and found: https://github.com/addyosmani/getUserMedia.js . But how do I get the video on the server? IE10 doesn't support webRTC, which http://lynckia.com/licode/ is built on.
getUserMedia is a WebRTC API. It doesn't exist on IE10.
Your alternatives are
Go with a Flash based solution (red5, Wowza, etc)
Use a plugin for WebRTC on IE (check out this one: https://bloggeek.me/temasys-free-webrtc-plugin/
Use Ziggeo (they should be able to use WebRTC or Flash automatically for you, taking care of all relevant transcoding and format changes necessary to playback the recorded stream). CameraTag (virtually the same at first glance) was also suggested in another answer, and likely, there are more.
I'm a bit at my wits end here.
I want to stream a live video broadcast to a web browser.
Currently I use ffmpeg to stream a directshow live source as a webm stream to node.js which then forwards the stream to the http request from the <video> element. So far everything works.
live source -> ffmpeg -> POST [webm] -> node.js -> GET [webm] -> video tag
My problem is that the source clock and the web clients clock doesn't exactly match each other (not that surprising). For video this is not a problem, dropping or duplicating a frame every now and then is not noticeable. However, with audio it is another issue. From what I've been able to figure out so far Chrome (or any other browser) does not perform any form of audio resampling compensation (e.g. swr_set_compensation from ffmpeg) to compensate this mismatch. Instead I get quite audible audio distortions (a loud beep) when the playback buffer runs out of samples.
My question is whether it is possible to achieve proper playback (with audio) of a live source in a web browser?
I haven't tried using silverlight or flash for playback yet. Would that possibly work better?
Live media (audio or/and video) streaming to a web browser has been possible for a couple of years though it is still making progress as of today. It is the next big thing for media on the web and many platforms like Youtube are already on board.
A typical live media streaming scenario is:
audio/video feed > transcoding > streaming > player
At each step you have several technological possibilities available. However I should already mention here that the road to live media streaming is paved with proprietary technologies.
audio/video feed: either raw or very lightly compressed media format and cannot be uploaded as such to the Internet. You need to transcode it. You may have to use a grabbing device like a PCI Express card or USB/thunderbolt device to get your cam onto a computer.
transcoding: you have software (ffmpeg, Flash media live encoder, Wirecast) or hardware solutions (streamingmedia.com has a wealth of information on the subject like here). H264/AAC is the current media professional standard and streams are often transcoded to multiple renditions (bitrate) to suit different network conditions.
streaming: you most likely need to target multiple devices to deliver your live stream. Not all devices support the same streaming protocol. HLS works on Apple devices and Android > 4.1. HDS or RTMP works in Flash, Smooth streaming in Silverlight. You cannot reach all devices with one protocol so in this case you would need a streaming server like Wowza or Red5. A streaming server take as an input a transcoded live stream and prepare it for cross device delivery while sustaining a massive number of simultaneous connections (over a thousand is not uncommon nowadays). It can also add functionalities like DVR or DRM. As of today the effort is around HTTP adaptive bitrate delivery. Large companies add CDN support for global delivery.
player: to display your live stream with various options like custom layout, closed captions, ads, chat module and more. Flash has been leading the market up until now for live media streaming on desktop. You can use HTML5 video for iOS and Android where HLS is supported.
Coming in fast is MPEG DASH and it works live with HTML5 video. There is a JS lib that supports live. I have tested it and it works though I may not use it for a production case scenario just yet as it is still a bit clunky (on demand support is better) and browser support is narrow at the moment (As of 8/30/13, Desktop Chrome, Desktop Internet Explorer 11, and Mobile Chrome Beta for Android are the only browsers supported).
I cannot comment much on your solution because I have not used node.js for streaming but it sounds like an interesting effort. A typical solution I would use relating to your case:
Device > ffmpeg (H264/AAC) > Wowza > Hybrid player (Flash + HTML5).
Instead of Wowza you could use Red5 (free/open source - but not much activity as of late). You can also look into Nginx RTMP module which supports HLS and MPEG DASH on top of RTMP.
For flash I use Strobe from Adobe which support live streaming and is easy to set up and a fallback to HTML5 where flash is not supported. I use SWFObject lib to detect flash support and feed a HLS URL to an HTML5 video tag for mobile devices. You can use RTSP for Android < 4.1 and other mobile devices.
Another thing I should mention is real time communications. For video/audio conferencing you could have a look at WebRTC. Those 2 articles should get you on the right track. Here and here. WebRTC will work great for one to few, one to one, few to few. If you need to support more concurrent connections you can have a look at Licode or tokbox.
I had a someone ask me if it were possible to record a person's voice inside of a web-browser - similar to what can be done on many operating systems. Is this possible through javascript or flash? Are there any programs that let you do this, and have the wav/mp3 of the recorded audio uploaded to the server? I'd prefer the solution to not use Flash if it can be avoided.
Thanks
You could use WAMI-Recorder. It still uses flash, but is designed to work with Javascript and data can be sent with HTTP POST. However, there's no way to access the microphone with Javascript alone.
At the moment, Chrome is the only browser that supports x-webkit-speech which is what they use to allow users to use voice to search Google on their desktop. HTML5 has getUserMedia, but support for that is also minimal and if your project is for mass consumption then it's unlikely to be a suitable.
Hey check out this post where i replied to a guy with a similar question as you.
How do I embed a Flash audio recorder in my site
I have done extensive research into client side only microphone recording and could not come up with any solution other than flash. If you follow my correct answer in the thread i posted, you can record audio from a microphone on the client side and then you can modify the code yourself and upload the audio to your server, its not that hard just a simple POST call.
If you do find a non-flash way to record let us know!
What is the fastest way to stream live video using JavaScript? Is WebSockets over TCP a fast enough protocol to stream a video of, say, 30fps?
Is WebSockets over TCP a fast enough protocol to stream a video of, say, 30fps?
Yes.. it is, take a look at this project. Websockets can easily handle HD videostreaming.. However, you should go for Adaptive Streaming. I explain here how you could implement it.
Currently we're working on a webbased instant messaging application with chat, filesharing and video/webcam support. With some bits and tricks we got streaming media through websockets (used HTML5 Media Capture to get the stream from our webcams).
You need to build a stream API and a Media Stream Transceiver to control the related media processing and transport.
The Media Source Extensions has been proposed which would allow for Adaptive Bitrate Streaming implementations.
To answer the question:
What is the fastest way to stream live video using JavaScript? Is
WebSockets over TCP a fast enough protocol to stream a video of, say,
30fps?
Yes, Websocket can be used to transmit over 30 fps and even 60 fps.
The main issue with Websocket is that it is low-level and you have to deal with may other issues than just transmitting video chunks. All in all it's a great transport for video and also audio.
It's definitely conceivable but I am not sure we're there yet. In the meantime, I'd recommend using something like Silverlight with IIS Smooth Streaming. Silverlight is plugin-based, but it works on Windows/OSX/Linux. Some day the HTML5 <video> element will be the way to go, but that will lack support for a little while.