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
Related
My primary intention is to setup a VoIP session between 2 users A & B; Here the raw audio / video media bytes are fetched from A's browser are played in B's browser and vice versa.
The reason is that, when the user C & D are added into this call, we need not have to create a P2P mesh network which limits the performance.
Tried recording media with getUserMedia() and playback, but it is not real time. It also gives a bad user experience. (However, haven't experimented yet with videos of small chunks as 200 ms)
Is there any approach where I can get the raw bytes of the media and play it on other browser? Currently I have a server in between which can connect to both peers if required.
Any online examples or libraries are welcome.
Have already asked 2 questions in this regard with 100-100 bounties, but not much of use:
How to use libsrtp or similar library to decrypt/encrypt the WebRTC data stream?
How to integrate part of WebRTC as a static / dynamic library with the existing C++ code?
Related: How to stream, live video playing on my browser to browser of another user?
If i understand you well is you're looking on how to have more than two users on the session right? without using mesh topology
thats possible and configurable as well by means that some maybe active speaker or everyone is active speaker not only receiver whatever configuration you choose but to me it seems that you're asking for video conferencing
there are couple of tools for this the best one i might recommend is mediasoup its a SFU as selective fowarding unit mediasoup
I don't know if I understand correctly, but it is not likely that you will get raw video data and play it on the browser, it will just kill your bandwith and performance because the raw data is huge.
You need to use the compressed data ( media codec ex.H264 ) and you need a protocol to send and receive it. If you are looking for sub-second latency than webrtc is your best choice in here already. If you have a server in between, distribute your media through that server instead of Mesh. Check this out for webrtc network topologies:
https://antmedia.io/webrtc-servers/
I am working on a project for large group broadcasting in WebRTC since it needs to work on iOS and Android devices, I am using Kurento, and iOSWEBRTC cordvoa plugin to build this I am curious if anyone can help improve my plan, or if there is a easier way to achieve this.
We need to have a video/audio conference with 5 people per room, however we need to be able to show that video to large audiences. Now my idea would be use Kurento as a middle-man and capture the streams into .webm files for live playback as the conference is going on.
Is there a better way to achieve this? And how would I playback the webm file as it is being recorded, it needs to update and continue playing as more video is sent, basically a live stream copy of the camera.
I am unsure if I am going the best route but I figured that would reduce the bandwidth from my original idea, I originally was thinking of making it like this:
5 person conference for broadcasters X number of viewers then downloaded those streams however I realize the upload bandwidth requirement would be crazy high, that is why I settled on this idea. Additionally the viewers do not have to see real time like the broadcasters. They need to be able to see and communicate with each other at the same time and the viewers can be a few seconds behind.
TL;DR:
Trying to make a 5 person video conference with video/audio capturing to then live stream it to viewers players. This would allow avoiding of PeerConnection bandwidth limitations. Would this work or am I forgetting something?
You'll need to look into using an SFU or MCU. An MCU is very costly, but multiplexes video streams and sends down a single video stream to all peers, and can also record that stream. An SFU is a single point of receipt of all streams, and selectively forwards them to clients. It could record off individual streams and then you could do post-processing to make a single recording out of the multiple recorded streams. A mesh network of connections really doesn't work for this use case.
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 am working on a project which will involve http live media streaming from a variety of devices like android phones/tablets, iphone, ipad, browser,etc. It will be a 2 way communication for all the devices with multiple devices connected to a conversation. I have implemented it partially i.e. one way by capturing audio from android phone(native app) and streaming to a web browser(HTML5 app) with a PHP server using ffmpeg and cvlc. I wanted to know of the best way to go ahead about it. Like, if there are any standards to be followed. Also what kind of a server should I be using? I don't want to use any streaming servers like Red5. I would like to implement the streaming logic similar to Http LiveStreaming by apple. I have come across MPEG-DASH that seems to be a standard for http streaming. I still have to look deeper into it. I was also thinking of using NodeJS for its popularity with streaming. Another worry was how do I go about capturing of media from devices? As in, should I use the native capability of the devices to convert media into an mp4 or any container that it supports and then stream it to the server or capture audio and images for a particular period of time and then send it to server and create a common output(I am not really sure of this idea). The separate capture is basically for simplifying the process of video streaming from the server end to any device. I was also thinking if I could completely bypass the server in any cases like a phone to phone or phone to tablet connection.
I just wanted to be sure of the things I will be using/implementing so that I wouldn't have to make drastic changes later on. Any help is deeply appreciated. Thank you.
I'm evaluating building an application which, simplifying the requirements, records from a microphone equipped small computer (eg: a Raspberry PI) and streams the digitalized sound over wireless connection in almost realtime to a server on the same LAN (No Internet involved). Ideally, the server application would record different streams from various wifi microphones and mix them together..
I'm currently looking into obtain a pretty good quality out of this, comparable somehow to a 128Kb stereo MP3.
At this point, I'm still evaluating options here, so I'm also looking to see your opinion on the feasibility of this.. if you think it's doable, what libraries, APIs, protocols would you use? Consider that this will be likely deployed on Linux based embedded computers (for the wifi mic part) and Linux based servers.
Thanks for your help.
I listen often Shoutcast on the iPad. This sounds pretty good to me. I do not know the kb/s rate there, I think they stream mp3. So I do not think this would be a big issue if you can live with the quality loss which comes with mp3. The bigger issue might be, how good your wireless connection is. When your network is pretty busy, there are more errors and lower speed. It also depends on the wireless standard and the hardware you are using. You may think about buffering, too.