What I am trying to do:
I have a live stream coming from my raspberry pi and I would like to view it on a webpage being served from a server which consumes the stream from the r_pi. What would be the best way to accomplish this?
Also It is preferred to have low latency.
I have looked at:
https://gist.github.com/vo/a0cc9313861888ad5180f442a4b7bf48
which works but there is a large latency and doesn't work on chrome. Any ideas?
This also is an option:
https://blog.miguelgrinberg.com/post/video-streaming-with-flask
But it uses mjpeg. I am curious if I can directly consume a stream either on the server or on the client.
I guess I can do client side consumption by having the server feed the client the html page directly (although I don't know the implications of having a gstream pushing video to a tcpserver [tcpserversink]).
Could I just take the stream convert it to a webm and feed this webm to the client?
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 multiple Raspberry Pi Devices with the native camera in my home and office (PUBLISHERS). - Publisher(Pi) they are on a local network behind a firewall/router and connected to the internet.
I have an EC2 webserver (BROKER). It is publicly accessible over a public IP Address.
I have an Android App on my phone. It has internet connectivity through a 4G Network. (SUBSCRIBER/CONSUMER/CLIENT)
I am trying to view the live feed of each of the raspberry cameras on my Android app. The problem is more conceptual than technical. I am unable to decide what should be the right approach and most efficient way to achieve this in terms of costs and latency.
Approaches, I have figured out based on my research on this:-
Approach 1:
1. Stream the camera in RTSP / RTMP in the pi device via raspvid/ffmpeg
2. Have a code in the pi device that reads the RTSP stream saves it to AWS S3
3. Have a middleware that transcodes the RTSP stream and saves it in a format accessible to mobile app via S3 url
Approach 2:
1. Stream the camera in RTSP / RTMP in the pi device via raspvid/ffmpeg
2. Have a code in the pi device that reads the RTSP stream pushes it to a remote frame gathering (ImageZMQ) server. EC2 can be used here.
3. Have a middleware that transcodes the frames to an RTSP stream and saves it in a format on S3 that is accessible to the mobile app via pubicly accessible S3 URL
Approach 3:
1. Stream the camera in WebRTC format by launching a web browser.
2. Send the stream to a media server like Kurento. EC2 can be used here.
3. Generate a unique webrtc pubicly accessible url to each stream
4. Access the webrtc video via mobile app
Approach 4:
1. Stream the camera in RTSP / RTMP in the pi device via raspvid/ffmpeg
2. Grab the stream via Amazon Kinesis client installed on the devices.
3. Publish the Kinesis stream to AWS Cloud
4. Have a Lambda store to transcode it and store it in S3
5. Have the mobile app access the video stream via publicly accessible S3 url
Approach 5: - (Fairly complex involving STUN/TURN Servers to bypass NAT)
1. Stream the camera in RTSP / RTMP in the pi device via raspvid/ffmpeg
2. Grab the stream and send it a to mediaserver like gstreamer. EC2 can be used here.
3. Use a live555 proxy or ngnix RTMP module. EC2 can be used here.
4. Generate a unique publicly accessible link for each device but running on the same port
5. Have the mobile app access the video stream via the device link
I am open to any video format as long as I am not using any third-party commercial solution like wowza, antmedia, dataplicity, aws kinesis. The most important constraint I have is all my devices are headless and I can only access them via ssh. As such I excluded any such option that involves manual setup or interacting with desktop interface of the PUBLISHERS(Pis). I can create scripts to automate all of this.
End goal is I wish to have public urls for each of Raspberry PI cams but all running on the same socket/port number like this:-
rtsp://cam1-frontdesk.mycompany.com:554/
rtsp://cam2-backoffice.mycompany.com:554/
rtsp://cam3-home.mycompany.com:554/
rtsp://cam4-club.mycompany.com:554/
Basically, with raspvid/ffmpeg you have a simple IP camera. So any architecture applicable in this case would work for you. As example, take a look at this architecture where you install Nimble Streamer on your AWS machine, then process that stream there and get URL for playback (HLS or any other suitable protocol). That URL can be played in any hardware/software player upon your choice and be inserted into any web player as well.
So it's your Approach 3 which HLS instead of WerRTC.
Which solution is appropriate depends mostly on whether you're viewing the video in a native application (e.g. VLC) and what you mean by "live" -- typically, "live streaming" uses HLS, which typically adds at least 5 and often closer to 30 seconds of latency as it downloads and plays sequences of short video files.
If you can tolerate the latency, HLS is the simplest solution.
If you want something real-time (< 0.300 seconds of latency) and are viewing the video via a native app, RTSP is the simplest solution.
If you would like something real-time and would like to view it in the web browser, Broadway.js, Media Source Extensions (MSE), and WebRTC are the three available solutions. Broadway.js is limited to H.264 Baseline, and only performs decently with GPU-accelerated canvas support -- not supported on all browsers. MSE is likewise not supported on all browsers. WebRTC has the best support, but is also the most complex of the three.
For real-time video from a Raspberry Pi that works in any browser, take a look at Alohacam.io (full disclosure: I am the author).
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'm trying to sent some audio stream from my browser to some server(udp, also try websockets).
I'm recording audio stream with webrtc , but I have problems with transmitting data from a nodeJS client to the my server.
Any idea? is it possible to send audio stream to the server using webrtc(openwebrtc)?
To get audio from the browser to the server, you have a few different possibilities.
Web Sockets
Simply send the audio data over a binary web socket to your server. You can use the Web Audio API with a ScriptProcessorNode to capture raw PCM and send it losslessly. Or, you can use the MediaRecorder to record the MediaStream and encode it with a codec like Opus, which you can then stream over the Web Socket.
There is a sample for doing this with video over on Facebook's GitHub repo. Streaming audio only is conceptually the same thing, so you should be able to adapt the example.
HTTP (future)
In the near future, you'll be able to use a WritableStream as the request body with the Fetch API, allowing you to make a normal HTTP PUT with a stream source from a browser. This is essentially the same as what you would do with a Web Socket, just without the Web Socket layer.
WebRTC (data channel)
With a WebRTC connection and the server as a "peer", you can open a data channel and send that exact same PCM or encoded audio that you would have sent over Web Sockets or HTTP.
There's a ton of complexity added to this with no real benefit. Don't use this method.
WebRTC (media streams)
WebRTC calls support direct handling of MediaStreams. You can attach a stream and let the WebRTC stack take care of negotiating a codec, adapting for bandwidth changes, dropping data that doesn't arrive, maintaining synchronization, and negotiating connectivity around restrictive firewall environments. While this makes things easier on the surface, that's a lot of complexity as well. There aren't any packages for Node.js that expose the MediaStreams to you, so you're stuck dealing with other software... none of it as easy to integrate as it could be.
Most folks going this route will execute gstreamer as an RTP server to handle the media component. I'm not convinced this is the best way, but it's the best way I know of at the moment.
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.