kousik19 / SIP
☆38Updated 4 years ago
Alternatives and similar repositories for SIP:
Users that are interested in SIP are comparing it to the libraries listed below
- ☆40Updated 4 years ago
- Stream Asterisk audio over Websockets☆174Updated 8 months ago
- A fully featured browser based WebRTC SIP phone for Asterisk☆564Updated 5 months ago
- Speech Recognition in Asterisk with Vosk Server☆113Updated 10 months ago
- The Asterisk Documentation Project.☆37Updated this week
- Minimalist Windows / OSx / Linux SIP Softphone with API Control☆137Updated last month
- SIP softphone☆170Updated last month
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆169Updated 10 months ago
- Asterisk external speech to text application☆59Updated last year
- Integrate ChatGPT into Asterisk☆37Updated last year
- OpenSource Freeswitch & Kamailio Billing, rating and Routing Software☆107Updated last year
- Vicidial open source telephony platform based on Asterisk☆82Updated 12 years ago
- Modern and flexible SIP/VoIP cli tool☆342Updated last month
- SIPREC recording server based on drachtio and rtpengine☆89Updated 9 months ago
- Barebone Opensource Powered SBC☆108Updated 8 months ago
- OpenSIPS AI Voice Connector Community Edition Platform☆35Updated 3 weeks ago
- OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder☆166Updated 2 months ago
- ARI examples in Python and JavaScript.☆107Updated 9 years ago
- Simple bidirectional audio protocol☆84Updated last month
- ☆77Updated last month
- Asterisk configuration GUI☆155Updated 2 weeks ago
- Demo of scalable Asterisk on Kubernetes☆170Updated last year
- Docker image providing Asterisk PBX☆224Updated 4 months ago
- Pure python VoIP/SIP/RTP library. Currently supports PCMA, PCMU, and telephone-event☆259Updated last month
- Normally, IP Camera streams video as RTSP protocol. But browser is unable to render this format, so we need to convert it as HLS format s…☆98Updated 2 years ago
- OTF React SIP.JS Phone☆23Updated last year
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆208Updated this week
- Dashboard for Queues in Asterisk and FreeSWITCH. app_queue panel for Asterisk and mod_callcenter in FreeSWITCH. Get news -> http://eepu…☆184Updated last year
- IVOZ Provider - Multitenant solution for VoIP telephony providers☆206Updated this week
- Sip To WhatsApp Gateway for Converting Sip Voice Protocol RTP Audio to WhatsApp Voice Call Protocol,☆31Updated last year