drachtio / docker-drachtio-freeswitch-mrfLinks
Dockerfile for creating a minimal Freeswitch image for use with drachtio-mrf
☆20Updated 10 months ago
Alternatives and similar repositories for docker-drachtio-freeswitch-mrf
Users that are interested in docker-drachtio-freeswitch-mrf are comparing it to the libraries listed below
Sorting:
- Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf☆55Updated last week
- Core telephony feature server for the jambones platform☆71Updated this week
- nodejs client for accessing rtpengine via ng protocol☆23Updated 2 years ago
- Freeswitch ASR module☆11Updated 3 months ago
- SIPREC recording server based on drachtio and rtpengine☆95Updated last year
- A SIP call processing server that can be controlled via nodejs applications☆288Updated 2 months ago
- OpenSIPS AI Voice Connector Community Edition Platform☆61Updated last month
- WebRTC SIP based VoIP client software (+chrome extension)☆109Updated 10 months ago
- Classes of Telephony for the Python's diagrams package☆11Updated 3 years ago
- Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration☆36Updated 7 years ago
- Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.☆44Updated 3 years ago
- Node.js client and server for FreeSwitch Event Socket☆138Updated 9 months ago
- Ari-proxy connects Asterisk, an open source communication server, to the Apache Kafka distributed streaming platform.☆62Updated last week
- Dockerfile for creating a minimal Freeswitch image for use with drachtio-mrf☆10Updated last year
- Asterisk Management Interface (AMI) to Web-socket proxy☆88Updated 2 years ago
- drachtio signaling resource framework☆193Updated this week
- React SIP user agent☆55Updated 5 years ago
- SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…☆178Updated last year
- Simple RTPEngine Speech-to-Text Recording Spooler☆16Updated last year
- UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services☆217Updated last week
- Complete VoIP Solution on kubernetes based on Kamailio/freeSWITCH☆56Updated 2 years ago
- REST API for sharing IP addresses sending unwanted SIP traffic☆62Updated last year
- Call API is a front-end layer for managing advanced SIP call flows. It listens for WebSocket connections and talks JSON-RPC 2.0 over them…☆55Updated 3 weeks ago
- SIP outbound proxy based on drachtio and freeswitch that includes siprec client functionality☆20Updated last year
- Kubernetes dynamic configuration engine for Asterisk☆68Updated 3 years ago
- Javascript library to build a web-broswer softphone☆100Updated 2 months ago
- HOMER 7 Docker Images☆96Updated last year
- Easier web calling by providing a layer of abstraction around SIP.js☆65Updated last year
- SIP Phone WebRTC for your browser☆60Updated 4 years ago
- Pipe arbitrary data rows (logs, events, cdrs, esl, etc) to HEP Server (HOMER)☆25Updated last year