HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring

Setup for a WEBRTC client and Kamailio server to call SIP clients

Ansible scripts to install and configure SIP3

Barebone Opensource Powered SBC

FreePBX container (Asterisk 16; OpenPBX 15 with Backup and IVR modules installed)

Dockerfile for freeswitch

How I replaced Skype with Twilio to make phone calls from my computer

openduplex uses speech-to-text, artificial intelligence and text-to-speech, to call businesses and make appointments for you

Dockerized FreePBX 16 with Asterisk 16, PHP 7.4, dedicated MySQL database, Hashicorp Vault integration and data persistence

The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic.

Skype for Business/Lync PSTN gateway

Kamailio in docker container with TLS enabled using Let's Encrypt.

OpenIMSs open source environment for real life development of IMS based 4G/5G/NR voice/video/data/RCS/IM services

Docker container for running Kamailio as part of a WebRTC Gateway

Jami (formerly Ring (formerly SFLphone)) is a SIP compatible softphone for GNU/Linux.

🐋 Minimal FreeSWITCH Docker images

How To Use Router as Sip Server (Asterisk openwrt) ...

Code samples to Get started quickly with Symbl's Voice SDK and APIs: Node.js, JavaScript, WebSockets, & PSTN.

IP SLA is a project to control the SLA of VoIP circuits

Notes for daily study