Sip2ban.github.io
Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS
| Project Name | Stars | Downloads | Repos Using This | Packages Using This | Most Recent Commit | Total Releases | Latest Release | Open Issues | License | Language |
|---|---|---|---|---|---|---|---|---|---|---|
| Gosip | 449 | 3 | 4 months ago | 4 | June 28, 2020 | 19 | apache-2.0 | Go | ||
| Public Switched Telecommunications Network Unleashed | ||||||||||
| Homer App | 186 | 6 months ago | 6 | April 21, 2021 | 23 | agpl-3.0 | Go | |||
| HOMER 7.x Front-End and API Server | ||||||||||
| Asterisk Opus | 72 | 9 years ago | 11 | gpl-2.0 | ||||||
| Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration | ||||||||||
| Asterisk Config | 46 | 2 years ago | 30 | June 08, 2022 | 4 | other | Go | |||
| Kubernetes dynamic configuration engine for Asterisk | ||||||||||
| Freepbx | 37 | 2 years ago | 7 | mit | Shell | |||||
| FreePBX container (Asterisk 16; OpenPBX 15 with Backup and IVR modules installed) | ||||||||||
| Asterisk Dialogflow Rtp Audioserver | 27 | a year ago | 12 | mit | JavaScript | |||||
| Docker Asterisk | 10 | 7 years ago | Shell | |||||||
| My playground to dockerize asterisk configuration | ||||||||||
| Tg2sip | 9 | 3 years ago | n,ull | gpl-2.0 | C++ | |||||
| Telegram <-> SIP voice gateway | ||||||||||
| Sip2ban.github.io | 9 | 9 years ago | gpl-2.0 | Perl | ||||||
| Avoided SIP and RTP attackers in Asterisk, FreeSwitch and OpenSIPS | ||||||||||
| Dana Tsg Rtp Stt Audioserver | 5 | 4 years ago | 3 | mit | JavaScript | |||||
| An AudioServer that takes audio from Asterisk via UDP and sends it to Google's Speech To Text Engine | ||||||||||
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