Closed. This question needs to be more focused. It is not currently accepting answers.
Want to improve this question? Update the question so it focuses on one problem only by editing this post.
Closed 8 years ago.
Improve this question
I would like to build a service that allows a user to listen to a call live from their browser.
I have some experience with Asterisk and this seems to be flexible enough to do what I have described.
Node.js sounds good because it is purported to handle concurrency well, and, I like JavaScript.
In the browser I figure that the HTML5 audio tag, since it handles playing from a streaming source, would be fine to play the sound.
A colleague of mine worked together a demo of this concept using Icecast, but was not able to finish it. There were also significant latency isssues.
My question is this:
How should I go about getting started on this?
Any help is appreciated!
Update:
I found a presentation discussing implementing SIP on top of WebSockets via a SIP proxy on the backend:
http://sip-on-the-web.aliax.net/
Once I have this up and running, the next step would be implementing the streaming. It seems like I should be able to proxy the audio output that would normally go to the sip client, through a secondary server that then streams it to the browser. I wonder why this couldn't be done all in memory? Then there is no need to write and read the file as the call proceeds.
If you're willing to wait for Asterisk 11, we're currently working on implementing support for WebSockets directly in Asterisk. More on it here:
Asterisk 11 WebRTC/RTCWeb Support
I'll just quote Kevin here, since he summarizes it better then I can:
"Today, the in-progress development branches have support for the WebSocket transport protocol (used for communicating signaling messages between the browser and Asterisk), SIP over WebSocket (currently being standardized by the IETF) and ICE/STUN/TURN (media handling mechanisms for NAT traversal and connection setup security). In addition, there’s a new Jingle/Google Talk/Google Voice channel driver, and we plan to support Jingle over WebSocket as well. At this point, we don’t have a quite complete solution (a new Canary build of the Google Chrome browser is needed with a few small changes), but each of the pieces has been tested and we’re anxious to see it all work together. We’d like to thank Iñaki and José from the SIP-on-the-Web project for providing us their JavaScript SIP stack to use during our testing, and we’ll probably be testing with the PhonoSDK as well for Jingle support."
This seems a nice guide
Remote call-center solution using Node.js
I've built a similar solution here. In this post I'm talking a little about it:
http://www.igorescobar.com/blog/2014/08/13/working-with-asterisk-and-node-js/
I built a call center solution using Node.js (Express/Socket.io), Javascript, HTML5 and CSS3.
I think trying to stream an audio file while it is being recorded will have extreme latency issues that you will not be able to get around. If you want to get real-time listening to a phone conversation I would suggest looking into Phono. It is a JQuery plugin that turns your web browser into a phone. Then you would just have the listener conferenced in to the conversation with it on mute.
If you don't mind the latency (caused by buffering of the Icecast stream), Asterisk is able to stream to Icecast (configure Asterisk's Ices application).
If you can't tolerate the latency, you'll need a browser-based SIP client. Unfortunately, there's not many of them that aren't locked to someone else's phone system. You might try red5phone (http://code.google.com/p/red5phone/) but that requires that you set up a Red5 server.
Related
I'm painfuly new to webrtc world. Have learned the essentials but when I want to build my own very simple app, I understand I'm missing a lot to start develope.
What I want?
Build a receiver audio node server that receive the audio stream and do with it something
(my main goal is sending the stream to another SIP node after adding some audio background effects, but that can wait for another time)
What the heck is my problem?
I don't know from where to start and what is the best way for me to do it right... I'm kinda lost..
There are already built webrtc servers like Kurento or Janus from one hand,
and (It sound better choice for me, but I'm not sure) some node base webrtc servers like node-webrtc, easyrtc, electron-rtc to the other hand
I also looked at the native API and it is a bit complicated and it is written in CPP that I do not master (but can learn if you guys tell me it is the best way to go)
What choice should I choose for my purpose?
I'm fairly good programmer and can cope with any lang, but still wish the easiest and best option for my purpose.
Please, if anyone was at the same phase as I'm now and passed it,
Or if you know how to help me and set me on the right track.
Please help
As you can see, I'm lost and beg for guidance.
Thank you heros of the world wide web!!
If you want someone using a browser to speak with a SIP user (or another webrtc user), then one of the easiest paths is to setup a Freeswitch server and use its 'verto' webrtc js library. See http://evoluxbr.github.io/verto-docs
The doc above will show you how to make a webrtc call to Freeswitch. You will still need to find a way for the 2nd caller to find what call/session to connect to. Ask about that on the Freeswitch users mailing list.
We've made an application where two people can video chat with each other using TokBox, but are running into a lot of technical issues surrounding WebRTC and TokBox itself. I know that Twilio recently launched a Javascript version for their video service, but both TokBox and Twilio seem to be aiming for larger scale publish/subscribe operations. It also isn't as far along as TokBox.
Are there other services out there that can do web video 1 on 1's? Perhaps some that don't use WebRTC and therefore don't have the problems we are facing?
I can't help but to think back to ChatRoulette and similar apps.
If what you need is an application that needs to run within the context of a browser, then WebRTC is your only choice when it comes to the technology to use. There's just nothing else there now that Flash is officially dead.
If you need it to run purely inside a packaged PC/mobile application, then you can use something other than WebRTC, but I don't really see the need for that.
When using real time video technologies, one aspect to look at closely is the quality of the network you are using. The questions I usually ask myself are things like does Skype/Hangouts/FaceTime run any better? If the answer is "yes they do", then the problem is in the implementation you have done/used. If the answer is "no, they are just as bad" then you probably can't do a lot better either.
For alternatives, you can check out the vendors listed in this WebRTC Develoepr Tools Landscape: https://bloggeek.me/webrtc-developer-tools-landscape/
I don't know what you mean with "a lot of technical issues surrounding WebRTC and Tokbox itself", but I do know Tokbox handles millions of 1:1 streaming minutes every day, without issues, and it can even handle sessions with 1 publisher and 3000 subscribers at the same time, so, maybe the technical issues are not there, but in another place...
Closed. This question needs to be more focused. It is not currently accepting answers.
Want to improve this question? Update the question so it focuses on one problem only by editing this post.
Closed 7 years ago.
Improve this question
I am developing a video conferencing application for education purposes that uses WebRTC. It needs to be done in a star topology as it connects up to 20 participants.
Conceptually it is easy to understand, but I don’t know how to start, as I do not have any examples.
All clients will connect to a server using WebRTC, and the server will mix the video streams in a specific layout and send it back to all clients. Here are my questions/difficulties:
How to implement the server part? What’s the best technology (e.g. NodeJS)? Are there simple examples of a star topology application like that?
How can we start writing the MCU code? Are there examples? Or is it easier to customize an open source MCU like Licode/Lynckia?
How can I estimate the right AWS EC2 instance type that we will use as the MCU server?
How can I estimate the data transfer cost (the size, in GB/TBs) which will be transferred in 1h of conference?
Thanks a lot in advance,
Carlos
My two cents on your various doubts:
Personally, I prefer NodeJS, but from what I have seen, application server does not play much of a role in WebRTC communication other than passing messages between peers/ media servers, so go with a technology you are comfortable with.
That said, for examples, you can check out kurento's Tutorials in both Java and Node.js, Licode example(using NodeJS) and Jitsi Meet in Java.
Yes, I think going with existing MCU is good idea, better one is SFU, difference being SFU justs forwards streams not mixes them, mixing streams is a costly process thus MCU needs to have high processing power. SFUs are comparitively light, all you need is a good bandwidth for the server.
About last two points, not much idea, depends on your use case, what is video resolution of streams, how many people, you needs to run some tests and guage it.
simulcast is another interesting idea, unfortunately I believe it is still in development.
We build solution based on NodeJS on WebRTC. With this technology is one big problem - everyone is sending video stream to everyone. If you have 20 participants then every computer sends and recives video 19 streams.
We created restriction "up to 4 participants" per room and it works fine.
So in my opinion if you have enought time - you can mix two technologies (up to 4-5 users) WebRTC to save server time, and something different for bigger meetings.
Closed. This question does not meet Stack Overflow guidelines. It is not currently accepting answers.
We don’t allow questions seeking recommendations for books, tools, software libraries, and more. You can edit the question so it can be answered with facts and citations.
Closed 1 year ago.
Improve this question
Is there any open source alternative to talky.io? where client code and all server side code is available.
I just double checked and it seems the priologic team are keeping tawk.com code behind a paywall, just like talky.io.
In the webRTC ecosystem, vendors either focus on
an application, and the Backend is hidden (think Skype, or bistri, even though bistri pivoted to propose an API as well.),
a PaaS and
then both the sever code is hidden and you won't have an
application, at best a demo, because they do not focus on any
vertical and do no implement business logic (think AWS in general, or tokbox, temasys for webRTC)
propose consulting/app development and will provide a complete
application most often open source, but keeping some key components
(priologic: mobile SDK + app, &yet: app, algoworks, ...) behind a
paywall. They usually team up with a Paas and or Hardware vendor to
provide more compelte/scalable solution to their client
(priologic/oracle, ...).
It is very unlikely, and I wouldn't know of any, that there were a vendor that would provide a full solution open source. It is still too complicated to have a non commercial entity provide one. The ones which do provide a full solution do so for a limited scope.
In any case, an application is always focussed on a use case. Even though the underlying infrastructure might be the same, and the BE/client API might be the same, an app for contact center, and app for social dating, and an app for conferencing will be quite different because they implement a quite different business logic and address quite different market. It is reasonable NOT to expect a full stack, but to have only the top most layer left to implement.
I put a list of vendors and products there, but it s a little bit raw. So here is a recipe to build a free/open-source solution, and then where to look to upgrade:
mandatory: open source signaling server (easyrtc, signalmaster, peerjs-server, rtc.io, ...)
mandatory: BE API (easyrtc, simplertc, peerjs, rtc.io respectively)
optional but highly recommended: add the free turn server rfc5766, or the most advanced version "coTurn". Some of the open source server and library propose examples or how to to support this TURN server by yourself.
optional: a client API that brings you closer to your use case,
optional: a free plugin to support IE and Safari (temasys free plugin),
optional: a media server if you need to host many-to-many calls or conferences (MCU or SFU) (licode, meteecho's janus, medooze, kurento, jitsi's videobridge)
optional: a SIP gateway to connect to VoIP and/or pone by extension (PSTN).
and ... that's about as far as you can go with open source / free libs today. You might hit a scalability problem quite fast depending on your traction.
A next step would be to get hosted servers, but it's not free anymore.
Separate servers:
ICE/TURN/STUN: See xirsys/twilio for hosted solution,
Media server: see dialogic, radysis, for hardware and meedoze, Jitsi, acano, pexip, openclove for software/hosted solutions,
Full Paas including all of the above:
tokbox (beware of streamed minutes billing if you have large conferences, has recording and some features temasys does not have yet)
temasys
some of the media server vendors also market themselves as PaaS. I have not tested, so I can't comment or recommend.
If you want to connect to SIP/phone, you will need different vendors as hither temasys nor tokbox provide interoperability today.
You could have a look at jitsi https://jitsi.org/, which is an opensource solution for private communication and also serves as a video conference tool for the browser.
You could try Subrosa (latin for "under the rose"). According to https://subrosa.io/source: "The Subrosa client and server are both open source and licensed under GPLv3."
Would be better if the server component was GNU AGPL 3.0, to make sure anyone running a server makes their code changes available for re-use, but at least both ends are free code.
Closed. This question needs to be more focused. It is not currently accepting answers.
Want to improve this question? Update the question so it focuses on one problem only by editing this post.
Closed 8 years ago.
Improve this question
I am trying to understand what is the right tools for creating an app with text, audio and video exchange. As server-side tool i want to use node.js.
There is a lots of examples which shows you client-side code, but what about server-side? I know, that for WebRTC server is needed only for signaling purposes, but i cant find a descriptive giude helping me to find out how to say:"Hey, here is this guy, he wants to talk to you, so here's his IP" or something simmilar.
How to make sure that i am establishing direct conection? How to establish the most performant kind of connection? I know that there is some NAT traversal protocols, but how i can explicitly use/enable them?
Can i create distributed Skype-like network by connecting many-to-many peers and having some signaling/auth servers? Or maybe use some peers as servers for signaling only?
All the above can be obtained with software such as Flash Media Server (or Red5 for open source). If you want to use Node.JS, you will need to either create your Node services (message queue, media server), or use some already available, and have Node.JS handle the interaction between them. So all of these will be needed:
Node web service(s) with web sockets
Node / other message broker (mq)
Node / other media server (FMS, Red5)
Optional, a caching service for multiple Node web services (Redis)
You can choose Flash, it has great support for RTMFP/RTMP. If you really want WebRTC, you will have to create a STUN node service for p2p discovery, which is connected to the caching service to handle authorizations.
RTMFP is an option, webrtc too. Most performant depends on how you define performance: quality? latency? how should it be biased? If you want low latency, go for p2p. If you want recording capabilities, either rtmp or a node webrtc relay.
Yes, but you will most likely need a team to do that :)
Found almost all of the answers :)
There are no any restrictions for server-side tools, so you can use whatever you want. Node.js fits great. For communication purposes WebSockets or XHR can be used, but there is no restrictions for any others.
In the process of creating connections browser will generate some events with all needed data. You just need to send it to both sides and process it there. There is also offer/answer system, so connection can be made only with agreement from both sides.
Browser will try to establish best connection possible by default. If it's can't be done it will fallback to TURN, which translates data through server.
It will be possibe when DataChannels will be implemented.