Web Softphone (WebRTC)

RemoSIM features a browser-based softphone powered by SIP.js and WebRTC. You can place and receive calls through your home physical SIM cards directly from any modern web browser without installing third-party desktop software.

1. Interactive Web Softphone Client

Below is the client interface. Enter your Asterisk server’s WebSocket URL and extension credentials to establish a secure real-time WebRTC audio connection:

RemoSIM WebPhone

Ready to Connect

SIP Connection Settings

DIAL NUMBER

*43

2. WebRTC Requirements & NAT Traversal

WebRTC establishes peer-to-peer audio streams between your web browser and the Asterisk PBX. Because residential connections and cellular networks operate behind NAT (Network Address Translation), STUN/TURN servers are required to guarantee two-way audio.

What is a STUN / TURN Server?

STUN (Session Traversal Utilities for NAT): Discovers your public IP address and port mapping.
TURN (Traversal Using Relays around NAT): Acts as an encrypted audio relay when strict symmetric NATs or corporate firewalls block direct UDP media streams.

Recommended TURN Providers

If you experience one-way audio (you cannot hear the caller or they cannot hear you), configure a free TURN service like Open Relay by Metered or deploy your own self-hosted eturnal / coturn relay daemon.

3. Server Configuration for WebRTC

To enable SIP over WebSocket on your Asterisk server, verify that the following configurations are active in your FreePBX / Asterisk settings:

Transport: Ensure the ws (WebSocket) and wss (Secure WebSocket) transports are enabled in sip_general_additional.conf or FreePBX SIP Settings.
Audio Codecs: Enable opus, ulaw, and alaw for browser compatibility.
SSL Certificate: Modern web browsers mandate that WebRTC microphone access only works over HTTPS. Your WebSocket URL must use wss:// with a valid TLS certificate (e.g. via Let’s Encrypt).

4. Testing Your Audio with the Echo Test

Once registered, dial *43 on your keypad and press Call:

You will hear an automated prompt welcoming you to the Asterisk Echo Test.
Speak into your microphone. If you immediately hear your voice echoed back with minimal latency, your WebRTC audio path, codecs, and NAT traversal are configured properly.

Native Smartphone Softphones

Prefer using an app on your iPhone or Android device? Check our Mobile Clients Guide for detailed SessionTalk and Zoiper configurations.