I'm trying to develop a peer-to-peer desktop app with Electron and WebRTC which transfers only JSON data betwen peers. I ran into many libraries such as PeerJs, node-crt and electron-webrtc but I'm not sure what's the best way to properly integrate that, any ideas? thanks
Personaly, i have chosen https://github.com/andyet/SimpleWebRTC, which is an API pretty easy to setup. This is not relative to Electron, this is open source, there is no API key needed and it works pretty well!
But to transfer JSON data you can just use websockets, because you will need it with webRTC for signalling anyway... (ok this is not a p2p solution)
For those who come across this post now, I would recommend https://github.com/feross/simple-peer, as it provides a simpler abstractionon top of webRTC and is actively maintained. It appears that as of now, SimpleWebRTC has been deprecated.
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I’m planning to build a video conference app. (NodeJS + React Native)
Requirements
One to One Video Conference ( 2 Speakers )
Video / Audio Recording of both the participants.
Store the recorded stream in an S3 bucket and watch the videos directly from it.
Live Streaming (Future Goals, but not at the moment)
Strategies tried so far:
Tried Twilio and Agora, but it wasn’t feasible due to pricing.
Mediasoup (SFU - inspired from dogehouse) was another option, but it’s relatively new and the development time takes much longer.
So I have come to a conclusion to start with Peer to Peer using WebRTC with React Native and record videos on a virtual server by connecting as a ghost participant. ( 2 Speakers + 1 Ghost Participant)
Need some strategies to implement WebRTC recording at the server. (Recordings are a bit crucial, so I don’t want to depend on the client)
Should I go with Puppeteer on server, join as ghost participant and record whenever a room is created, If yes - Is it possible to run multiple instances of puppeteer? Because at times, multiple room recordings might happen, so it needs to record concurrently. Need to confirm the scalability.
Look into Kurento / Jitsi
Any other options?
Great, if you could help me out! Cheers!!
As a developer evangelist for Agora, I want to say thanks for considering Agora. With regard to the pricing, while Agora offers a generous free-tier (10k min/ month) this is meant for development usage and once your project is deployed into production, costs will scale similar to hosting infrastructure (like AWS/GCP).
As with any project, to cover costs you will need to have some monetization strategy or have some free credits to grow the business. Similar to other platforms Agora has a start-ups program for qualified startups.
All that being said to answer your question about approach, I can tell you that the ghost client approach should work, Agora's cloud recording uses similar logic. With regard to scalability you could run multiple puppeteer instances.
You can take a look at html5 videocall web application on GitHub for inspiration.
As it uses Wowza SE as relay for scaling and reliability, streams can be recorded server side with FFmpeg. FFmpeg can input one or multiple streams, mix/transcoded and output to a local or external destination.
More advanced setups like PaidVideochat - Turnkey Videochat Site on WordPress support mixing multiple streams from conferencing/calls in same video file.
Using a relay streaming server is also great for scaling to multiple viewers.
The Galene SFU has native support for server-side recording (disclosure, I'm the main author). However, it is a fairly young project, which might be a problem for you.
I am trying to build an application which works this way: I as a user want to start a call with another user. The way I want the connection be made is by random. So it will connect to one of the many clients out there by random. Also when other clients try to make a call, it should connect to another random client and so on. I want those phone calls be made via application(such as WhatsApp) not as a phone call.
Now, the question is; is Twilio a good approach for this purpose?
If yes can you tell me which of their feature would fit my app the best?
Thanks for any suggestions!
Twilio developer evangelist here.
I can answer that Twilio would be a good approach for you to do this within your own application. I'd recommend using Twilio Video to build this as it allows cross platform communication via audio or video (in your case, you may not need the video, but this will give you the best audio quality).
As an example, my colleague Dominik built a video roulette application. It is the case that the interface was built in JavaScript for the web, but the idea would be the same for a native app. The code for the server side part of the application should give some insight into how to connect random pairings.
It's also possible to integrate Twilio Video with CallKit and Connection Services so that you can make outbound calls to other devices that ring like a real incoming call.
I want to develop a video chat application, between Web browser and an Android device. As far as I know I have two prominent options, WebRTC and RMTP. I have tested out WebRTC and for Web app it was quite convenient to use, so I am inclined to use it. However, I should consider all my options, since I know little about Android development.
Do I have any reason to choose RMTP over WebRTC in the following use case:
Simple 1 to 1 video chat
Between Android application and Web browser ( just Chrome and Firefox is fine)
Recording and storing the call
Or neither has a clear advantage over the other in this simple case? For peer discovery I have a separate application server.
For a 1:1 video chat, there is no reason whatsoever to use RMTP.
RMTP is good (and even that is debatable in 2015) for streaming - a case where one end is producing the content and many on the other end are consuming it.
For something bidirectional, you should just pick WebRTC - its codecs are better, its availability is better and its technology is better.
I'm trying to create a virtual classroom. Since I'm not familiar with the web conferencing (or conferencing) terminology, I'm not sure if I'm understanding WebRTC's capabilities as I should.
I've looked in the examples for WebRTC, and all that I've found seem to be peer-to-peer connections. As I understand it, peer-to-peer connections are between two entities. However, virtual classrooms are different as far as I know; they require all parties to be connected to each other, so that when one user speaks/types, all users hear her.
Is such a thing possible with WebRTC? If so, what is it called and how can I read more about it?
Check out the open source Big Blue Button project (http://bigbluebutton.org/). They're currently Flash based but are actively moving towards webRTC. Rumor has it they'll be using Kurento as their MCU. They also have open source mobile (Android/iOS) application code.
According to http://www.html5rocks.com/en/tutorials/webrtc/infrastructure/, such a thing is possible:
Beyond one-to-one: multi-party WebRTC
You may also want to take a look at Justin Uberti's proposed IETF standard for
a REST API for access to TURN Services.
It's easy to imagine use cases for media streaming that go beyond a simple
one-to-one call: for example, video conferencing between a group of colleagues,
or a public event with one speaker and hundreds (or millions) of viewers.
A WebRTC app can use multiple RTCPeerConnections so to that every endpoint
connects to every other endpoint in a mesh configuration. This is the approach
taken by apps such as talky.io, and works remarkably well for a small handful
of peers. Beyond that, processing and bandwidth consumption becomes excessive,
especially for mobile clients.
Maybe you can try searching in the webrtc google group
hope this helps
I want to create audio, video and text messagtes chat. Is it possible using WebRTC? Or it only allow audio and video chats?
One side of my app will be implemented using browser. An other one - using C++ native API.
Does anyone have examples in native C++ API and/or javascript?
The WebRTC specification is still very much in flux, but there's a DataChannel API in the spec that is implemented in an early form in both Firefox and Chrome. DataChannels are intended to allow you to send arbitrary bytes between peers, and the spec provides for both reliable (TCP-like) and unreliable (UDP-like) channels.
I am not sure if WebRTC allows for text chatting. I was able to successfully create an Android Application that performed all of this, but only with the combination of Google's Libjingle and WebRTC libraries. Within the Libjingle library there are several example programs/pieces of code that demonstrate the library's functionality. The call example in Libjingle sounds very similar to what you are wanting to do, and is what I built my Android application out of. The only thing is I have not yet ported it to an web browser, so I am not sure if Libjingle will work with that.
I have begun looking into it, and I have found some people on the WebRTC discussion group that have developed a very nice Multi-user video chat application for a web browser that is built using WebRTC. It is capable of video (along with voice) communications as well as text chatting. I do not know if this matters, but it all occurs within a single interface (meaning that it does not seem to allow for separated/singular form communications -- text only, voice only, video only). I am sure that it would not be too difficult to separate them all out if you wanted/needed. They have posted all of their code onto GitHub and seem to be actively updating and improving it.