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 Teams).
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.
- 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)
- SIPMediaGW
- Kamailio
- Coturn
- SIPCAPTURE: HOMER, HEP
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.
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.
See the Documentation index for details.
SIPMediaGW relies on several open-source projects such as Coturn, Kamailio, Homer, Baresip, FFmpeg, Pulseaudio, ALSA, Video4Linux, Fluxbox.
This project is licensed under the Apache 2.0 License. See the LICENSE file for details.