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)
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/
In essence, I'd like to create an online radio where users can upload music to be played at specific times. Is webRTC capable of this or would I be better served going with something like Icecast?
WebRTC is about peer2peer communications.
If users would upload their media on to your server, then you would need to use some WebRTC-compatible media streaming software (such as Wowza, for example) for serving the media via WebRTC; in other words, the server would have to act as a WebRTC peer.
For the described task WebRTC is not the case, on my opinion. Icecast & Co would be better suitable for the task. Basically, I believe that it can be built using just HTML5 (and JavaScript, probably).
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.
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.