I currently have a video chat app working on web(Flash) and android via Adobe AIR, it uses Adobe Media Server (RTMP) as backend for video streaming and shared objects, my question is, if there is another server or solution that provides many to many live video broadcast maybe using H.264 codec from android and iOS, have some sort of user list and room list stored in a database or similar, I want to move away from Adobe as it has many limitations on mobile devices.
Live video is crucial in 1 to many broadcasts that will have hundreds of viewers at the same time.
Thanks for reading!
Ulex.fr created an RTMP connector for Asterisk (the free PBX platform).
Used with the Asterisk Vonference application, it allows you to create conference rooms for 1 to many configuration, with audio and video. The only one limitation is the power of your server. You can plan a scalable architecure in order to broadcast one video to many (many could be unlimited). We developp a specific protocol to connect and manage the connection based on the telephony events. I think we already done a direct RTMP connection that skip this protocol too.
All the project done by ulex.fr is free, OpenSource and GPL.
Get the full project here : https://github.com/voximal/asterisk-rtmp
(a live demo is available)
We already develop an RTMP stack for android with video (using the camera), this allows you to create your own application without using AIR.
You can check Adobe Cirrus, it's still in the beta stage (actually IMHO Adobe forgot about it), but it works on web, desktop and mobile too. Check this Video Phone example, it can handle chat applications without a problem.
http://labs.adobe.com/technologies/cirrus/samples/
You could take a look at Red5 Media Server, which is an open source solution. There are other options like the Wowza's solutions on AWS, but they come a higher cost...
Ok as today, we have decided that we can manage the users,rooms and messages via Google Firebase Real Time Database, and the live video stream using ANT Media Server
Related
We have mobile application that historically has used RTSP streaming to allow a user to watch a live stream, which currently is published via Wowza Streaming Engine. We have had a need to lower stream latency, so have gravitated towards WebRTC to achieve this.
The problem is that there seems to be a lack of documentation, or examples regarding the implementation of a react-native WebRTC viewer which connects to a remote stream.
Does anyone out there have any documentation, or code examples for this kind of implementation?
I do note there is a react-native-webrtc library, however, all examples demonstrate connecting two peers on mobile phones with their video cameras i.e. Like facetime. We are after an example demonstrating someone on a phone connecting to a remote streaming server with a video feed.
Cheers,
If you want a webrtc client to connect to a server you need a server doing webrtc with the proper signaling that fit your need. Webrtc don't care which signaling you use, so you have to choose it or choose a the platform you need.
There are a lot of different media server, or library that support webrtc in server side all having there specific signaling(ex: Freeswitch, Kurento etc), or no signaling (ex: Mediasoup). Few will have a react native version as Media Streaming is not really something in the javascript/UI side but you can do something with the webrtc react-native lib.
Twillio has a lot of supported platform and could be a good start if you search a ready to use solution.
Suppose I have 2 peers exchanging video with webRTC. Now I need both of the streams to be saved as video files in the central server. Is is possible to do it realtime? (Storing/Uploading the video from peers is not an option).
I thought of making a 3 node webRTC connection, with the 3rd node being the server. This way, I can screen record the 3rd node's stream or save it using some other way. But I am not sure about the reliability/feasibility of the implementation.
This is for a mobile application, and I would avoid any method that involves uploading/saving.
PS: I'm using Agora.io for the purpose of video-conference.
in my opinion
you can do it like the record demo:https://webrtc.github.io/samples/src/content/getusermedia/record/.
record each stream to blobs and push them to your server with websocket.
then convert the blobs to a webm file or just add in a video
Agora doesn't offer on-premise recording out of the box but they do provide thee code for you to be able to launch your own on-premise recording using your own server. Agora has the code and instructions to deploy on GitHub: https://github.com/AgoraIO/Basic-Recording
The way it works, once you have set up the Agora Recording SDK, the client would trigger the recording to start, via user interaction (button tap) or some other event (i.e. peer-joined or stream-subscribed) this will trigger the recording service to join the channel and record the streams. _The service outputs the video file once recording has stopped.
you need a WebRTC media server.
WebRTC media servers makes it possible to support more complex
scenarios WebRTC media servers are servers that act as WebRTC clients
but run on the server side. They are termination points for the media
where we’d like to take action. Popular tasks done on WebRTC media
servers include:
Group calling Recording Broadcast and live streaming Gateway to other
networks/protocols Server-side machine learning Cloud rendering
(gaming or 3D) The adventurous and strong hearted will go and develop
their own WebRTC media server. Most would pick a commercial service or
an open source one. For the latter, check out these tips for choosing
WebRTC open source media server framework.
In many cases, the thing developers are looking for is support for
group calling, something that almost always requires a media server.
In that case, you need to decide if you’d go with the classing (and
now somewhat old) MCU mixing model or with the more accepted and
modern SFU routing model. You will also need to think a lot about the
sizing of your WebRTC media server.
For recording WebRTC sessions, you can either do that on the client
side or the server side. In both cases you’ll be needing a server, but
what that server is and how it works will be very different in each
case.
If it is broadcasting you’re after, then you need to think about the
broadcast size of your WebRTC session.
link:https://bloggeek.me/webrtc-server/
I am building a web-based project which has webcam one-way broadcasting part .(A user can open its own cam and some viewers can join its room to only view and listen).
So i have decided to use Kurento Media Server(KMS) because of not having any experince with flash.
My questions in my head:
Do i need anything extra beside KMS to make a user broadcast webcam?
Can Kurento provide me the live streaming to webpage?
And What is the difference using Red5 or Kurento?
Thanks in advance
Do i need anything extra beside KMS to make a user broadcast webcam?
You'll probably need a TURN server for users that have some port limitations
Can Kurento provide me the live streaming to webpage?
Sure! Check the tutorials and the documentation for a full list of features.
And What is the difference using Red5 or Kurento?
Kurento is more than just a media server. It is a pluggable platform that offers computer vision and augmented reality capabilities, on top of video and audio streaming, recording and playing. It also offers WebRTC out of the box, which is something Red5 can't do as of today.
Disclaimer: I'm part of the Kurento team.
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.