Heyy Everyone,
We are designing a television module. In the current architecture, we have 2 independent devices, each running linux over Atom processor. We have a requirement to stream live transport stream from one device to another via a network. I tried looking for streaming softwares running on linux, which are capable of live streaming Transport Streams, but could not find any.
Any suggestions will be appreciated.
--
Sen
Try the following:
ffmpeg
Gstreamer
VLC
Live555
I'm using "Nginx" with "ffmpeg" on "linux" server.
There are many tools for this process. but newly I using srs.
Nginx
SRS
for send "RTMP" stream please use "Vmix" or "OBS".
For live streaming, both #dipan-mehta and #fakhredin-gholamizadeh give lots of open-source softwares, which is stable and widely used.
I want to share about more use scenarios about live streaming, first is about WebRTC. Although WebRTC is design for video conference, but it's also OK to use in live streaming, especially the realtime latency(<1s). Let's take a view from end-to-end latency:
HLS or DASH: About 3~10s latency, and LLHLS maybe more lower about 3~5s. Note that all the file-based protocols(HLS, HDS, DASH, CMAF, etc) is impossible to get 1s latency.
RTMP or HTTP-FLV: About 1-3s latency, works quit perfect for most of live streaming use scenarios.
SRT or WebRTC: About 0.2~1s latency, because they are UDP streaming protocol, and without cumulative latency caused by jitter of network.
Note that SRT is used in publisher systems, to replace RTMP, it's not support by player(H5) now.
In future, the live streaming protocol maybe QUIC or WebTransport, which is now RFC, and SRS is plan to support it.
As a live streaming server, SRS supports almost all of these protocols:
RTMP, HLS, HTTP-FLV: Stable.
WebRTC: Stable.
DASH, SRT: Experimental.
HDS: Deprecated.
LLHLS, CMAF, QUIC, WebTransport: In plan.
Note: Please see the latest Features.
Apart of protocols, it's also important to support DVR and Cluster.
Related
I have a Linode server and need to broadcast one to-many audio (they can hear but can not talk back) to a group of three to five people. I looked at WebRTC and the Janus server but it seems complete overkill. Using commercial applications like Skype, Discord etc. results in low audio quality and it is mono. Best possible audio quality and low latency (on a par with that of Skype, Discord etc.) is essential.
Any pointers would be greatly appreciated.
I can recommend building such system based on Icecast streaming. It's an old proven technology which has a latency close to real-time.
You could use any set of Icecast-enabled tools for that.
As example, here's what you an do with tools by our company:
Larix Broadcaster mobile app allows streaming in audio-only
mode.
Nimble Streamer software media server can get Larix' input and
produce Icecast stream. You can use any Icecast-enabled here
instead.
SLDP Player can play Icecast produced by Nimble
Streamer or any other Icecast-enabled server.
That can also be built with other companies products, so you can pick the right tools yourself.
A super simple setup would be to just use command line tool called ffmpeg (it also has an api) see doc at https://trac.ffmpeg.org/wiki/ffserver
Where your source audio lives just launch either the ffmpeg or ffserver
ffserver -f /etc/ffserver.conf
in that config put location of source audio and output url it will publish to ... then your client receivers can use ffplay with
ffplay <stream URL>
ffmpeg is a free open source industry workhorse for audio/video manipulation ... its the underlying technology several more visable tools like vlc use under the covers
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 am working on a RTSP RTP-over-TCP H264 streaming application from a live HW-based encoder.
I am sure , there is somewhere a example code, please any reference, web.
Most of the hardware appliances I have had to deal with (mostly IP cameras) use Live555. It is quite straightforward - you set up the needed tracks and then provide NAL units and the framework takes care of all the rest. They have a nice set of test programs that describe the API - at least I was able to make a RTSP server for a Texas Instruments Pandaboard SoC, and there are lots of info on Live555 on Stackoverflow.
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.