I am not able to compile the WebRTCTest application in the antmedia github repository. This is the application I am trying to compile.
I need to test live stream performance over webRTC. The setup is running Linux OS and I am using gstreamer to capture the video frames.
Following is the use case:
Stream live video frames being captured from the camera over to antmedia using WebRTC protocol. However, I tried Compiling the WebRTCTest application given in the above link, but I got a lot of compilation issues mainly due to the header file names in the webRTC code source.
How can I publish the video frames being captured from the camera to antmedia via webRTC?
Also does antmedia provide any webRTC based SDK for linux platform (C language based)?
Actually WebRTCTest has not been supported for more than 1 year. But there is another WebRTC Test Tool that is developed by Ant Media team to test Ant Media Server. It is java based and publishes video files. You can still use it on Linux and publish your web cam with small changes.
https://gitlab.com/Ant-Media/webrtc-test
you should use https://resources.antmedia.io/docs/load-testing as a testing tool.
But if you need to publish or play only with gstreamer , I have wrote a gstreamer program in C which uses webrtc to send and receive stream from and to ant media server https://github.com/USAMAWIZARD/AntMedia-Gstreamer-Webrtc. It is tested with AMS version Enterprise Edition 2.5.1
Related
I have an IP Camera with built in RTSP URL but it's not having a public IP. I want to stream it with as low latency as possible.
Is there any way that I can play my IP camera with WebRTC using Ant Media Server?
There are two ways to do this with Ant Media Server.
You can add the RTSP URL for your camera to Ant Media Server as a stream source. You can do this as documented here or you can use this REST method by providing the source URL. With this solution you will have video/audio data on the server side and you can record or re-stream it.
The second way is more innovative and but it is applicable for the camera which has own processors (mostly ARM) on it. In this solution you can run the Embedded SDK software on your camera's processor. This software can capture video/audio data from the camera and feed that data to the WebRTC peer. In this case data is sent directly to the WebRTC peer without passing through the server.
You can find for about Embedded SDK here.
You can find reference project executable files for different architectures here.
Lastly if you want to modify and build this reference project by your self you
will need SDK libraries. You can login to antmedia.io and download the SDK libraries.
I'm trying to build a live video streaming application from a usb camera to an application running on a remote desktop. I've researched protocols like RTMP, RTSP, WebRTC. According to my understanding I can't use webRTC since it's only compatible in the browser and I'm not building my application for a browser here. Please help me choose the right protocol and also the media server.
You can, and many applications do, use WebRTC outside the browser. WebRTC implementations are available for many different platforms including iOS, Android and embedded systems.
You can even use Headless Chrome if you want to use the Chrome APIs without the visual parts of the browser.
Short version:
I need an in-browser solution to deliver the webcam and mic streams to a server.
Long version:
I'm trying to create a live streaming application. So far I've only managed to figure out this workflow:
Client creates stream (some transcoder is probably required here)
Client sends(publishes?) stream to server (basically hosts an RTMP/other stream that should be accessible by my server)
Server transcodes, transrates, etc. and publishes the stream to a CDN
Viewers watch published stream
Ideally, I'd like a browser-based solution that requires minimal setup from the client's end (a Flash plugin download might be acceptable) and streams the webcam and mic inputs to the server. I'm either unaware of the precise keywords or am looking for the wrong thing, but I can't find an apt solution.
Solutions that involve using ffmpeg or vlc to publish a stream aren't really what I'm looking for, since they require additional download and setup, and aren't restricted to just webcam and mic inputs. WebRTC probably won't serve the same quality but if all else fails, I think it can get the job done, at least for some browsers.
I'm using Ubuntu for development and have just activated a trial license for Wowza streaming server and cloud.
Is ffmpeg/vlc et. al. the only way out? Or is there something that can do the job in a single browser tab?
If you go the RTMP way, Adobe Flash Player supports H.264 encoding directly. Since you mentioned Wowza you can find an example and complete source code (including the fla) in the examples directory. There's also a demo here. There are many other open-source Flash capture plugins.
You can also use the aforementioned Flash recorder without Wowza. In this case you'll need a RTMP server, a notable example being the Nginx RTMP module which supports recording (to flv) and also offers callbacks that allow you to launch the transcoding once the recording is done.
With WebRTC you can record (getUserMedia, MediaStreamRecorder) small media chunks and send them to the server where they will get concatenated or using the peer-to-peer communications features of WebRTC (RTCPeerConnection). For a detailed overview see my answer here.
In both cases you'll have issues with devices/browsers that don't support Flash or WebRTC, eg. iPhones, Safari. Plus getUserMedia doesn't capture the same format across all browsers: Firefox audio/video in WebM and Chrome audio in wav and video in WebM.
For mobile devices you'll probably have to write apps.
In WebRTC, I always see the implementation about peer-to-peer and how to get video streaming from one client to another client. How about server-to-client?
Is it possible for WebRTC to streaming video file from server-to-client?
(I am thinking about using WebRTC Native C++ API to create my own server application to connect to the current implementation on chrome or firefox browser client application.)
OK, if it is possible, will it be faster than many current video streaming services?
Yes it is possible as the server can be one of the peers in that peer-to-peer session.
If you respect the protocols and send the video in SRTP packets using VP8, the browser will play it. To help you build these components on other applications or servers, you can check this page and this project as a guide.
Now, comparing WebRTC with other streaming services... It will depend on several variables like the Codec or the protocol. But, for instance, comparing WebRTC (SRTP over UDP with VP8 Codec) against Flash (RTMP over TCP with H264 Codec), I would say that WebRTC wins.
The player will be Flash Player against the native <video> tag.
The transport would be TCP against UDP.
But of course, everything depends on what you are sending to the client.
I have written some apps and plugins using the native WebRTC API, and there isn't a lot of information out there yet, but here are a few useful resources to get you started:
QT Example: http://research.edm.uhasselt.be/jori/qtwebrtc
Native to Browser example: http://sourcey.com/webrtc-native-to-browser-video-streaming-example/
I started with the WebRTC Native C++ to Browser Video Streaming Example but it doesnot build anymore with the actual WebRTC Native Code.
Then I made modifications merging into a standalone process :
management of the peerConnection (the peerconnection_server)
access to Video4Linux capture (the peerconnection_client).
Removing the stream from browser to the WebRTC Native C++ client give a simple solution to access throught a WebRTC browser to a Video4Linux device that is available from GitHub webrtc-streamer.
Live Demo
We are attempting to replace MJPEGs with Webrtc for our server software and have a prototype module for doing this using a smattering of components tied to the Openwebrtc project. It has been an absolute bear to do, and we have frequent ICE negotiation errors (even over a simple LAN), but it mostly works.
We also built a prototype with the Google Webrtc module, but it had many dependencies. I find it easier to work with the Openwebrtc modules because Google's stuff is so tightly tied to general peer-to-peer scenarios on the browser.
I compiled the following from scratch:
libnice 0.1.14
gstreamer-sctp-1.0
usrsctp
Then I have to interact with libnice a bit directly to gather candidates. Also have to write out the SDP files by hand. But the amount of control--being able to control the source of the pipeline--makes it worthwhile. The resulting pipeline (with two clients off one server source) is below:
Of course. I'm writting a program using native WebRTC api which can join the conference as a peer and record both video and audio.
see: How to stream audio from browser to WebRTC native C++ application
and you can definitely streaming media from native app.
I'm sure you can use dummy_audio_file to streaming audio from local file, and you can find a way to access the video streaming progress by your own.
Yes it is. We have developed an load test tool to publish and play for Ant Media Server. This tool can broadcast media file. We used the same native WebRTC library used in Ant Media Server.
Sure it's possible, it allows covert live streaming to WebRTC, for example:
OBS/FFmpeg ---RTMP---> Server ---WebRTC--> Chrome/Client
For this scenario, it allows the ultra low latency live streaming, about 600~800ms, to play the live streaming by WebRTC. Please take a look at this demo.
Does any one knows about RTSP streaming using WOWza Server ?
I want to play it on a MPMoviePlayer controller in iOS6 but it shows not enough buffer to keep it up. My webservice urls work fine because I have also checked them using a browser but I can't find anything about RTSP streaming.
Does any one have any tutorials about RTSP streaming on iPhone using WOWza Server ?
rtsp streaming is not possible on iPhone, iPad and iPod touch. Please refer to the following link.
http://www.wowza.com/forums/content.php?62
MPMoviePlayer only supports HTTP live streaming. to have RTSP working on iPhone, you must implement your client on the iOS.
There is library live555 which implements for you, but you must integrate it with the code. Also decoding of the stream must be implemented in software, by you or a 3rd party library.
Wowza support re-restreaming the content as HLS, if this is your wowza server than there are easy instructions on the www.wowza.com site, if its not perhaps they are streaming HLS.
If you have to use rtsp, there are several good players available on the app store, or you can build your own using lib 555 as mentioned or one of our open or closed sourced frameworks.
https://github.com/mooncatventures-group/AVDemoPlay2L