I have one website say social network hosted on BlueHost. THen i need a RTMP hosting server to enable live video chat. Bluehost doesn't provide it. Is it possible i can host the video chat part on another service that runs RTMP and connect the two sites?
Yes, you can freely embed information from another server to your main site.
Depending on implementation, you probably have to host the Flash stream player on the same server that is hosting the RTMP. This is because the same origin policy controls what servers the Flash applet can connect to.
The easiest solution may be to use an iframe to embed the video chat.
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/
Is there a simple guide from where I can start creating a stun / turn and signaling server ?
I spend over a week searching for those things and couldn't find any guide where I can say:
okay, I am on the right track now - this is clear.
So far, everything is so abstract without any examples.
This is what I'm trying to achieve: a simple video stream on my local network where I'll have a server with installed usb camera on it, and an application on my iis which will connect to the usb camera and stream it to the clients, and every time when a client opens the application, will see the video stream from the server camera.
Note: since I want to use it on my local network do I really need a stun/turn server, or is there a guide that shows how to avoid it ?
Media streamed over dedicated servers HTTP/HTTPS rarely needs a NAT traversal solution. Instead, just have your web server with camera attached, on the public Internet or behind your NAT with port-forwarding enabled.
There are LOTS of streaming media solutions available as open source, free downloads, or commercially sold. A good list is here:
https://en.wikipedia.org/wiki/List_of_streaming_media_systems
I am planning a software application where the user will be able to select a given media channel from a list of RTMP streams available on one or more media servers on the internet. The list should ideally be dynamically created through some kind of service that knows about the available and active channels.
My question is: Would this be possible through some kind of protocol between the service and the media server. I understand that RTMP by itself doesn't allow this. A therefore assume that some outbound mechanism will be required.
No...
...there is no native application discovery in RTMP. If you'd like this kind of functionality you'll need to program some sort of discovery service for which ever streaming server you are running.