sip-to-webrtc

README

sip-to-webrtc

SIP is an example of accepting inbounding SIP traffic (Invites) and bridging it with WebRTC. This is the most common way to connect phone calls with your WebRTC application. This is possible because of the excellent emiago/sipgo library.

This example demonstrates accepting SIP audio and playing it in the browser. If you wish to implement multiple participants in a call you will need to have audio mixing. See zaf/g711 for decoding + encoding. After you have decoded sum all the samples and then encode it.

Instructions

Open sip-to-webrtc example page

jsfiddle.net you should see a audio player, two text-areas and a 'Start Session' button

Run sip-to-webrtc with your browsers SessionDescription as stdin

In the jsfiddle the top textarea is your browser, copy that and:

Linux/macOS

Run echo $BROWSER_SDP | sip-to-webrtc

Windows

1. Paste the SessionDescription into a file.
2. Run sip-to-webrtc < my_file

Input sip-to-webrtc's SessionDescription into your browser

Copy the text that sip-to-webrtc just emitted and copy into second text area

Hit 'Start Session' to connect

If a WebRTC session was successfully established you will get log messages about ICEConnectionState going to connected. In your browser and terminal.

Browser

checking
connected

Terminal

Connection State has changed checking
Connection State has changed connected

Starting SIP Listener

If everything worked now it is time to make a SIP Invite.

Make your phone call

sip-to-webrtc is now listening on :5060 and will accept all invites. When an Invite has been accepted you will see a log message like this.

Accepting SIP Invite: From: "+15550100001" <sip:+15550100001@192.168.1.93>;tag=nc8uzmZUHUTbqH0v

Done

You should hear the audio of the phone call in your browser.

Congrats, you have used Pion WebRTC! Now start building something cool

Documentation

Overview

sip demonstrates how to bridge SIP traffic and WebRTC