SIP.js | A simple , intuitive , and powerful JavaScript signaling | TCP library

I want to import a module for one of my .js files and call my .js file with Vaadin 14. Unfortunately I struggle with importing it and a ReferenceError is thrown in the browser.

My view:

Good evening Stack Overflow! I really need help for a project of mine where I'm using sip.js and a VoIP to make real calls to a phone number.

The Goal

I want to allow the user to record the audio and microphone and save the data on a server (in base64 encoding or as a file). So I after the conversation can hear the conversation again and use it for what ever my purpose (employee training) was.

The Problem

I can't get the sound of the person speaking, which comes through and -HTML tag (working with the sip.js plugin). As of now I haven't found any way to successfully save the sound streaming through this audio tag.

What I've done so far

I've successfully figured out how to record the audio of the microphone using a plugin called AudioRecorder which allows me to record the audio through the microphone and saving it. I slightly changed the code so it got saved encoded as base64. This all work as expected, though I only get the audio of my own voice, and not the person I'm talking with.

Because I succeed to record the audio of my own voice I looked into the AudioRecorder plugin and tried to reverse the plugin to record from a audio tag. I found the "createMediaStreamSource" function inside AudioRecorder which I wanted to work with the -tag which did not work (as I suspected, because the -tag in it self isn't a stream (of which i understand).

The Code

I'm basically using the sip.js plugin to establish a call to a phone number by using below code (just using an example, matching my code, because my raw code contains some added values which doesn't need to be showed here):

I am trying to implement Sip.js on my project. I have created a session variable and stored the session in that , but now I need to track an event on that session variable to attach audio on DOM. How do I add a event listener on a variable.

Here is how my code look like:

I use freeswitch as sip server also use pem from freeswitch folder (/usr/local/freeswitch/certs) .

On client part i use SIP.js client looks like :

I'm developing a mobile app that must implement Twilio Programmable Video. I have developed a backend with Twilio SDK 5.24.2 in Laravel PHP Framework and that works well.

In the mobile app, I have installed the twilio-video 2.0.0-beta4 package through npm and developed a page based in the example in "video-quickstart-js-1.x". The problem is when Twilio tries to connect to the room, the error in the console is as follows (at the end of the log):

How to Identify Online users in WebRTC app? Iam using Sip signalling and Sip.js. Iam unable to find proper documentation or on web. I thought to send invite request to all contacts and if one responds,that one is online,BUT for sending request back from client 2 we need to create peerconnection or data Channel, that wil be against Proper technique..so pls suggest me something Many begginers may be facing this problem so I thought this will teah me plus benefit others.

So I am using jssip 3.2.10 to make calls on a React project.

The server is setup on Asterisk and CentOS.

I can make calls where the call receiver hears me well, but I can't hear their audio, nor the waiting (traditional) beep noises it should make until the call is picked up.

It does work with some sipml5/asterisk udp online tests so I feel it's on my clients side issue. I tested it on Chrome and Firefox (both latest, with the same results).

My setup

I have a helper to connect called sip.js:

Has anyone seen that error, literally:

"Uncaught DOMException: Failed to construct 'RTCPeerConnection': Cannot create so many PeerConnections"?

I guess it's happens because I didn't removed connections (maybe not, maybe it's because I didn't provide more memory or something). But from where? Where do they accumulate?

I didn't found answer for the question, literally, nowhere.

I was tried to solve that problem a couple days. By the way, no such thing as a stupid question, right?

UPD: Forgot to add that I using sip.js library.