Currently, I'm finding a lib able to stream video from multiple sources through one RTP Stream (one connection). Anbody have sugesstion on it?
Actually, I figured out that Opal 3.8 is VoIP lib, supported RTP/H264. But I don't know whether it can support mux/demux rtp media from one stream? If no, can you give me some suggesstion?
Thanks,
There are a few RTP stacks around and which one you use depends on which language you are going to be developing in, pjmedia is a good cross-platform one.
RTP streams can only carry media from a single source so you won't be able to multiplex multiple video streams into a single RTP stream, see Synchronization source (SSRC) on page 9 of the RTP RFC. What you could do is have two separate RTP streams (different SSRC's) being sent from the same socket which would mean you're mutliplexing them as far as the network is concerned. If you actually want to combine multiple video streams into a single RTP stream then you need to mix them which is a whole different kettle of fish.
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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'd like to be able to capture the audio from the audio card of my computer and to dispatch it with WebRTC. However, I am not sure if it's possible or not to have access to the audio directly produced by my computer.
According to this repo https://github.com/niklasenbom/RecordingApp/blob/master/app.js there is a system audio stuff but not sure if it's what I'm looking for.
Thanks,
You can do it by using NAudio. Actually I did the same project myself and will put it in GitHub in a few weeks and update this answer. You can configure the frequency etc. and use it's OnDataAvailable event to dispatch the sound to registered clients.
Right now my goal is to grab a streaming video from an IP surveillance camera and display it on a web page.
The camera allows to encode the streaming either in h264 or mjpeg, and transmits it by the RTSP protocol.
The streaming has to be available for several kinds of devices (mainly computers, android smartphones and iphones).
According to my findings it seems like the best option for doing that (in terms of latency) is to transmit the frames of the video through a websocket:
http://phoboslab.org/log/2013/09/html5-live-video-streaming-via-websockets.
Almost all the implementations of this mechanism I've found are based on mjpeg since it's easier to get the video frames.
There's also a h264 player: https://github.com/131/h264-live-player, based on https://github.com/mbebenita/Broadway, which I didn't manage to run ( I would appreciate any help in that respect).
Now the first question is: it is worth trying to work with h264 (since it saves a lot of bandwidth). Or would the h264 decode process probably introduce too much latency?
I would also like to ask if anyone knows a better solution that the one I'm trying to implement.
Finally, where I say "additional information" I mean that I might want to include some additional data associated with some video frames. (something like subtitles or telemetry data).
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 have written an application that receives media files from a central server and plays those files according to a playlist. All works well.
A client has contacted us and wants to use our application to play some audio files as presentations in a kiosk-style application. So far, so good, our application can handle this no problems.
He has requested as a potential feature that we would have a number of headphone sockets at the front of the kiosk. Each headphone socket would play the same audio presentation in a different language.
I have come up with the idea of encoding a single audio file with the presentation in multiple languages, and each language in a different channel. We would then require a sound card that could decode each channel and output it on a different headphone socket.
Thing is, while I'm think the theory is sound, I have absolutely no idea whether this is feasible and what would be required to pull it off.
Any ideas?!
As a side-note: the application uses Media Player as the underlying component to handle the playback of audio and video. I'd appreciate any help as to the software we could use to generate the multi-channel audio stream and the hardware (USB sound card would be fine) that we could use to decode the stream.
Thanks!
You need to use multiple files not channels, its going to be way easier that way.
Instead of using Media Player use DirectShow (on .NET you have DirectShow.NET), In DirectShow you have the notation of Multiple files on the same graph.
You will be able to control to which audio device play which files, and your Play, Pause, Stop commands will be preformed on all files without you need to worry about syncing.
There are many samples on how to build media player like with DiectShow, extending them to use multiple files should be really easy.
For HW take a look at this (USB with 8 output channels)
I think with Shay's hardware you've got a complete solution:
Encode a 7.1 file with a different mono voice track on each channel.
Use the 8 channel output device in 7.1 mode, with a different headset in each port, and you've got it. Or, if you only have 6 languages, a 5.1 file would work. Many PC's have 5.1 outputs built in, you'd only need 3 splitters to break out the left and right channels from each jack.
You can do the encoding with Windows Media Encoder, or other pro audio tool.