This is a broad question - are there any solutions to WebRTC Video & PSTN integration ? The requirements are:
Multi-party WebRTC video conference (SFU or MCU, not peer to peer)
Ability to join the conference via PSTN end points (telephones) - obviously with audio fallback
Prefer paid service (like Tokbox or Twilio) rather than roll-your-own solution
We are currently using TokBox, however, it does not provide a PSTN integration. Since the call signalling is entirely hidden under the TokBox API, it seems unlikely that we could add (some kind of) WebRTC to PSTN gateway and make it work. Twilio has a video offering but it's actually in a very infant stage right now (peer to peer only, it seems with a limit of 4 participants).
Since we a Web App company and not a infrastructure company, I'd prefer a solution that handles the infrastructure part (like TokBox and Twilio do), but am open to other solutions as well, if that's what it'll take.
Thank you.
Avinash,
Twilio Video does not currently support PSTN integration. And there is currently that 4 participant limit regarding video chat.
This product is still a beta and constantly evolving so I'd suggest this group to you for keeping up with the updates.
Related
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'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 am new to WebRTC and vLine API. I need to implement a Group Video Chat just like Google Hangout using vLine API and PHP. Please suggest if any example or solution available or how to go about it?
Thanks
Tanjum
We have a PHP example on GitHub: https://github.com/vline/vline-php-example (More examples available here).
With the current API you can implement group chat with a mesh topology where each user establishes a connection with the other users (using person.startMedia() on each person you want to include in the conference). Due to both bandwidth and CPU, this won't scale well beyond four people or so.
We have a better conferencing solution in development (that won the Best Conferencing Award at the WebRTC Expo), but it's currently only available to a select group of beta testers.
I searched about this on google but could not find any suitable answer so posting here for help.
I want to implement video streaming with multiple participants connected. While google this topic I found that WebRTC provide similar functionality but I want to make sure whether WebRTC can support all my requirements.
I want to build an application that should support large number of participants in conference (around 10000).
I want to implement facility like one participant is broadcasting its video and audio streams and other are just listening to their stream.
Also when prompted only one participant will be able to communicate with broadcaster which will be managed by one participant (a administrator). Administrator will decide who can communicate with broadcaster.
Is same can be possible with any other WebAPI ?? I found OpenTok, but not confident if it provide any feature of moderation in conference (i.e. feature of having an Administrator who manages stuff)
Did anybody worked on similar concept or having any information related to this.
Let me know if I am not clear of any further details are required.
Any help would be useful,
Thanks in anticipation
Hardik - I am Product Manager at TokBox, the makers of the OpenTok platform. Good news: TokBox can fulfill virtually all of your requirements, but with a few caveats.
TokBox has been building a video chat/conferencing platform for years, long before WebRTC even existed in fact. In that time we have supported many customers with almost your exact requirements on OpenTok, a platform that is based on Flash (Major League Baseball is one such customer). Building applications on this architecture has the added advantage of solving virtually all of the interop issues that exist when connecting people using different devices and browsers. It is based on Flash however, which technically doesn't meet your WebRTC requirement. So you know, there's that.
WebRTC is where it's at though, which is why we created OpenTok for WebRTC in 2012. It was a complete rewrite of the platform that not only provides higher quality video, but also gives developers more hooks and far more control over how exactly they integrate video and audio chat into their primary customer experience.
Currently in beta (as of this writing in June 2013) are two new components in our WebRTC infrastructure. The first we refer to as Mantis, which solves many of the challenges associated with hosting large multi-party calls. The other is Cloud Raptor, which gives developers access to a stream of events stemming from a WebRTC session, and through which developers can issue events and commands of their own. Raptor is what enables you for example to moderate calls, boot participants, and control whose audio and video streams are broadcast to all the other participants.
So, TokBox has what you need. In the short term we can help you get up and running using OpenTok pretty quickly. Then we can discuss with you how to get you onto OpenTok for WebRTC and into our Mantis and Raptor beta program.