I have FreeSwitch working with SIP Clients for
Extension to Extension Call
Extension to PSTN / Gateway Call
PSTN/DID to Extension Call
I have configured WebRTC with SIPML5 clients and it is working on following scenarios
Extension to Extension Call
for Extension to PSTN/ Gateway FreeSwitch routes call to TRUNK but its not connected.
How can I get it working ?? What is missing parameter?? I think something at bridge is required to
FS Console logs are available here
http://pastebin.com/Ye0jw37x
Related
I am trying to implement WebRTC p2p communication for my business model and I am using Kandy (https://www.kandy.io/) service for the same
I have tried to implement the Video call feature. I have used the code available in the tutorial and have hosted the files over HTTPS, the call gets established but I don't find any communication happening (Both audio and video) , I tried the same code in Codepen the call works fine there. I don't undertstand what the issue is ?
Is there any standards that are to be met for the SSL or any server requirements are present to implement WebRTC ?
I want to create my own video chat application. I use the WebRTC framework. I read a few tutorials and each of theme assumes that signalling channel exists. How to implement my own signalling channel?
Since signalling is not defined for the WebRTC standard at the moment, it leaves you a few options. Check out this article for more info the following articles:
Signalling Options for WebRTC Applications
Choosing your signalling protocol
1.SIP over WebSockets
Companies like JSSIP offer a SIP signalling framework over Javascript. The advantage here is that it's interoperable with the usual VoIP structures.
JSSIP
SIPJS
SIPML5
2.The WebRTC Data Channel
Uncharted territory but viable!
Tutorial by Pusher
3.XMPP
If you take this route, it is probably either because you have an existing XMPP installation
Jingle
4.JSON over COMET or WebSockets
My favourite! WebRTC signalling shouldn't be done any other way than the Web way.
Matrix
Firebase
I hope this helps!
You can make a Node.js WebSocket server or other WebSocket server to broker the connection. Here is a simple guide that gets the first client talking to the server. An alternative is PeerJS, which can handle the signaling and alleviate most of the complexity of setting up the WebRTC call.
With serverless options available, vanilla HTTP AJAX options may not be bad for scalability and costs.
Create a plain HTTP(s) API exchanging information using JSON.
Is there any API to invoke a Java adapter from a HTTP adapter in IBM MobileFirst Platform Foundation 7.0?
I have tried with WL.Server.invokeHttp(input), but received a 404 response. What should be the value for 'path' in the input?
It is currently not possible to directly call a Java adapter from a JavaScript adapter.
You are encouraged to submit a feature request: https://developer.ibm.com/mobilefirstplatform/help
Alternatives:
Implement a JavaScript adapter that invokes Java code. See the tutorial Using Java in JavaScript adapters
Use another JavaScript adater - set the Java adapter as the backend it connects to and call it from the requesting JavaScript adapter
Rough steps to follow:
You have JavaScript adapter A
This adapter calls a procedure in another JavaScript adapter, adapter B (adapter mashup)
JavaScript adapter B will set in its XML the Java adapter properties as the backend (localhost, MFP server port)
Requests from this adapter should then be sent to /the-server-context-root/adapters/the-java-adapter/*
The Java adapter should have its security disabled, otherwise an access token would need to be forwarded in an header from the client in the request
As you can see the second option is less trivial to implement, and is less supported. I would investigate option 1...
I want to develop a Video Chat application. So I started to check the SimpleWebRTC JS Framework. When I share my screen, it requires HTTPS mode. If I hosted the files in SSL domain. But the Socket js (included file) is in an insecure domain.
http://signaling.simplewebrtc.com:8888/socket.io/1/?t=139165345345345
If I change the protocol into HTTPS then it doesn't return response.
Please advice.
You can host the socket.io.js library, which is the simpleWebRTC signalling server yourself on a node server, and point that as your signalling server.
That worked for me.
I'm working on a video conferencing app.
We have a server running erlyVideo for internal video streaming.
We have a rest service on our web server for our erlyVideo server to call to validate session keys from our external site.
What we're stuck on is how to hook into new connections to pass the session key to the REST service to decide whether or not to terminate the connection.
Any thoughts?
What's wrong with using authentication described in their docs?
http://erlyvideo.org/authorization
You need to use apps_rewrite_play module:
{rtmp_handlers, [...., {apps_rewrite_play, "http://rest-service"},...]},