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aghasuhail96/asterisk22-webrtc

By aghasuhail96

•Updated 3 months ago

A production-ready Asterisk 22 with complete support of WebRTC, PJSIP, SRTP, ARI, and audio codec.

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aghasuhail96/asterisk22-webrtc repository overview

⁠Asterisk 22 WebRTC Ready Docker Image šŸš€

A production-ready Asterisk 22 Docker image with complete WebRTC, PJSIP, SRTP, ARI, and audio codec support pre-configured.

This image is built for developers and teams who need a ready-to-use Asterisk environment without spending hours fixing missing modules, codec issues, WebSocket configuration, and WebRTC setup.

⁠✨ Features

āœ… Asterisk 22.x
āœ… Docker ready
āœ… WebRTC support
āœ… PJSIP support
āœ… Secure WebSocket (WSS) support
āœ… UDP SIP support
āœ… RTP media support
āœ… SRTP enabled
āœ… Opus codec support
āœ… G.711 (ulaw/alaw) support
āœ… G.722 support
āœ… GSM codec support
āœ… MP3 recording format support
āœ… MixMonitor support
āœ… ARI (Asterisk REST Interface) enabled
āœ… HTTP & HTTPS server enabled
āœ… WebSocket media support
āœ… Ready for SIP.js / JsSIP / browser based softphones

⁠Included Modules

This image includes commonly required Asterisk modules:

  • res_srtp
  • res_http_websocket
  • res_pjsip_transport_websocket
  • ARI modules
  • Opus codec support
  • MP3 format support
  • MixMonitor
  • Queue support
  • Conference support
  • Recording support

⁠WebRTC Ready

The image is prepared for browser-based calling.

Compatible with:

  • SIP.js
  • JsSIP
  • WebRTC softphones
  • Custom browser dialers

⁠UDP / SIP Ready

Standard SIP communication is fully supported.

⁠Included Configuration Files

This image includes Asterisk configuration files.

Configuration repository:
GitHub: soh-kaz/asterisk22-webrtc⁠

Configuration examples include:

config/
ā”œā”€ā”€ pjsip.conf
ā”œā”€ā”€ extensions.conf
ā”œā”€ā”€ http.conf
ā”œā”€ā”€ rtp.conf
ā”œā”€ā”€ ari.conf
ā”œā”€ā”€ logger.conf
└── other Asterisk configs

⁠Quick Start

Pull the image:

docker pull agahsuhail96/asterisk22-webrtc:latest

Generate TLS Certificate:

openssl genrsa -out ca.key 2048
openssl req -new -x509 -key ca.key -out ca.crt -days 3650 \
-subj "/CN=Asterisk CA"
openssl genrsa -out asterisk.key 2048
openssl req -new -key asterisk.key -out asterisk.csr \
-subj "/CN=localhost"
openssl x509 -req \
-in asterisk.csr \
-CA ca.crt \
-CAkey ca.key \
-CAcreateserial \
-out asterisk.crt \
-days 3650

⁠Use Cases

This image can be used for:

  • VoIP applications
  • Call centers
  • CRM integrations
  • AI voice agents
  • WebRTC calling platforms
  • SIP gateways
  • Automated dialing systems
  • Voice automation projects

⁠Why this image?

Setting up Asterisk with WebRTC often requires manual configuration of:

  • Missing codecs
  • WebSocket support
  • SRTP libraries
  • PJSIP WebRTC settings
  • Docker networking
  • RTP ports
  • Build dependencies

This image provides a ready foundation so developers can focus on building their applications instead of spending time on Asterisk setup.

⁠Contributions & Collaboration

Suggestions, improvements, and collaborations are welcome.

Feel free to open issues, submit PRs, or discuss ideas.

GitHub: soh-kaz⁠ Docker: aghasuhail96⁠

⁠License

Use freely for personal and commercial projects.

Always verify your own production security requirements before deployment.

Tag summary

Content type

Image

Digest

sha256:d5f2d4d58…

Size

558.4 MB

Last updated

3 months ago

docker pull aghasuhail96/asterisk22-webrtc