I am running kurento-media-server with this: https://github.com/lulop-k/kurento-rtsp2webrtc
example. OK, I am getting rtsp stream but it appears with delay about 3-4 seconds. As I saw in this post: How to disable video encoding In Kurento Media Server?
that can be because encoding running on the Kurento media sever. The problem is that I cannot install the openh264-gst-plugins-bad-1.5 because I am running 32 bit Ubuntu so I have to build it by myself. Currently I succeeded to build all packages but not this one! So as I saw in the google group it may be another way to distribute the rtsp media. Can you explain how to distribute media with WebRTC instead of HTTP streaming?
Following this https://github.com/Kurento/kms-elements/pull/3 article I recompiled the kms-elements and now I have no delay at all!!!
Related
I am using CWP Pro (Control Web Panel)
I have selected webserver = Apache + Nginx
I want to install RTMP and want to live stream on my website with obs studio.
My queries are =
Do I need to install NGINX even if I am using Apache+Nginx server ?
Maximum tutorials / search results are showing NGINX + RTMP installation guide. Do I need to install NGINX too ? Or only RTMP module ?
After installing RTMP, I have created url for streaming (e.g. rtmp://my_ip_address/live/stream_key), and added it in OBS studio. Started OBS streaming. But I am stuck at Code To Embed this live streaming in my html page of my website. How Can I EMbed it with video player lie video.js or other suggestions ?
Please consider about this solution in two parts:
CWP, the admin control dashboard, to manage your system and live streams.
Media System, the live streaming system, to publish by OBS, to play the live stream by some proper protocols.
Generally, there are some HTTP Callback and HTTP-API between the two system, so it's better to deploy and build them separately.
For Media System, the generally workflow is:
Generate the live stream URL by your CWP system, like the RTMP url you mentioned.
Use encoder, OBS as such, to publish the RTMP stream. RTMP is the widely used protocol by encoder, SRT is an optional, WebRTC is also able to publish live stream now, see this post.
Depends on your scenarios, H5 or Mobile, use some players to play the live stream. Well, it's complex, but RTMP definitely doesn't work, please use HLS/HTTP-FLV/DASH/WebRTC, see this post.
There are some commercial solutions too, which does the same things.
I have an iOS application that displays an rtsp stream from an IP camera on the local network, I would like to restream it to an external server in real time (Wowza to be specific) the server will take care of converting rtsp to HLS so that the users can view the live broadcast on their devices.
On a computer it would be pretty straight forward:
ffmpeg [input-options] -i [input-file] [output-options] [output-stream-URI]
But I need to do it programmatically on iOS, and I'm not really sure if it's even possible. Anyone?
If you already have a server (where Wowza is installed) a good idea would be to run FFMPEG there. You can trigger that from mobile (with a script requests that executes ffmpeg command).
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