I'm beginner developer.
And I don't speak engilsh very well. sorry
I want to broadcast live video from iPhone camera like iphone video call.
In this case, which do I choice better, hls or dss.
so, what's the functional difference HLS and DSS.
Can hls broadcast live video from iphone camera to another iphone?
Darwin Streaming Server is for RTSP streaming. HLS is a streaming technology based on using HTTP server for hosting the content.
iPhone to iPhone video isn't well served by either technology. It's possible to use an iPhone camera to capture video, upload it to a server, package it for HLS and serve it to the client viewers. This is very high latency (around 10-30 seconds), so it's likely not suitable for you.
If you want 1-to-1 messaging, you're probably better off using a real-time system like RTP, which is what's used by FaceTime and video calling programs.
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.
We are about to start a stream project and we are considering options right now, one options we are considering is we use RTMP to stream in mobile Android (or iOS), broadcast in the backend media servers (either Antmedia or Janus) and stream/play it in mobile device using RTMP, But for web users they will stream it thru WebRTC (as RTMP support only works in flash).
What is the advantage of such approach, or are there any pros and cons of such approach?
This is an alternative to the full WebRTC approach wherein mobile devices broadcast and publish WebRTC to media servers, played and streamed in WebRTC for both mobile and web users.
Any advantage/disadvantage of both approach?
(Sorry kinda new to the streaming world and such questions are raised by managers)
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).
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.
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