SIPMediaGW

SIPMediaGW is an open-source set of components that allows traditional meeting rooms with conferencing systems to join any modern platform (like Jitsi, BigBlueButton, LiveKit — and even Google Meet & Microsoft Teams). <br><br><br> <br><br><br> The room connector is compatible with all video devices supporting the SIP protocol. It has been tested with major devices from Polycom, Cisco, Huawei, and Aver.

<img src="docs/architecture.png">

Features

• Audio and video support
• Inbound and outbound calls
• Encrypted SIP and RTP traffic
• Autoscaling logic for Cloud deployment
• Content sharing via BFCP (Binary Floor Control Protocol)
• Streaming capabilities via RTMP (Real-Time Messaging Protocol)

Get started

Installation

• Development environment

Configuration

• SIPMediaGW
• Kamailio
• Coturn
• SIPCAPTURE: HOMER, HEP

Testing

Once the services are up and running, you can join a conference from your preferred SIP softphone. Refer to the testing section for more information.

Troubleshooting

The logs are handled by syslog of the host machine:

tail -f /var/log/syslog | grep mediagw

Inspect Kamailio database:

docker run -it --network=host --entrypoint mysql mysql -h 127.0.0.1 -u root -pdbrootpw kamailio -e "SELECT username, locked, to_stop FROM location"

For troubleshooting/monitoring purposes, real-time packet capture and visualization tools can be deployed as follows:

docker compose -f deploy/docker-compose.yml up -d --force-recreate heplify_server homer_webapp

NOTE: Homer and SIP Capture tools are automatically deployed with the Development environment.

Documentation

See the Documentation index for details.

SIPMediaGW relies on several open-source projects such as Coturn, Kamailio, Homer, Baresip, FFmpeg, Pulseaudio, ALSA, Video4Linux, Fluxbox.

License

This project is licensed under the Apache 2.0 License. See the LICENSE file for details.