little howto?

[verbavolant]

Hi!

• I've installed multiparty-meeting-sip, made a config.js (sip with ws, works)
• where i call the nodejs server at port 4443? from anywhere or from the server? If i call from a browser, i can see a login to asterisk from this browser, but i can't call
• i've set the sip-header for calling the correct room

Sorry but i can't understand what exactly mm-sip does!

[havfo]

There is lack of documentation in this repository, this has only been a POC so far. What you need to do is the following:

• Set config in public/config/config.js

  • uri, password and wsServers is the login of the SIP account you have set up on the Asterisk server
  • multipartyHostName and multipartyPort is the address and port to the multiparty-meeting server you want to call in to using SIP

• Configure Asterisk to accept calls from wherever you want to be able to receive calls from. Insert the header X-Room and pass the call to the account you have configured in the previous step.

• Start a browser and go to the site where you have hosted this repository.

The browser will then register to the Asterisk and wait for calls to come in. Based on the X-Room header it will then join the corresponding multiparty-meeting room and work as a bridge between SIP and multiparty-meeting, mixing audio and video. You will probably want to automate this browser, and run it headless, but this is not something that has been done yet.

[rtcexpert]

• edmeet-sip

  • change your config first as per above comment.
    Set config in public/config/config.js

    - `uri`, `password` and `wsServers` is the login of the SIP account you have set up on the Asterisk server
    - `multipartyHostName` and `multipartyPort` is the address and port to the multiparty-meeting server you want to call in to using SIP
    

  • yarn install

  • yarn start

  it will run on the port.

  For Test: You can run this on client side on browser.

  • It will register to your kamailio or SIP infra with WSS (it's your simple webrtc endpoint. you can use any of your webrtc endpoint).

• Now when ever you want to join the meeting.

  • You can init the call to your extension, That you register on edmeet-sip with X-Room.
  • It will connect to your room on multiplary-meeting and bridge the Audio/Video of SIP and Mediasoup.

• For server side deploy.
  You can use bellow sample code.

const express = require('express');
  const puppeteer = require('puppeteer');
  const app = express();
  async function ssr(url) {
    const browser = await puppeteer.launch({headless: true, args: ['--no-sandbox', '--disable-setuid-sandbox', 
"--enable-extensions",  "--enable-usermedia-screen-capturing", 
"--enable-experimental-extension-apis", "--enable-experimental-web-platform-features"]});
    const page = await browser.newPage();
    await page.goto(url, {waitUntil: 'networkidle0'});
    const html = await page.content(); // serialized HTML of page DOM.
    page.on('console', async msg => console[msg._type](
      ...await Promise.all(msg.args().map(arg => arg.jsonValue()))

  }

  setTimeout(async () => {
    await ssr("YOUR_IP:PORT")
  }, 1 * 1000);

  app.listen(8984, () => console.log('Server started. Press Ctrl+C to quit'));

[Alex-YddeR]

How can multiple SIP endpoints/clients can connect here with edumeet.