Alternatives and similar repositories for jester

Users that are interested in jester are comparing it to the libraries listed below

• davehorton / drachtio-fs-load-balancing-proxy
  Load-balancing SIP proxy for Freeswitch
  ☆26Updated 9 years ago
• sippy / rtp_cluster
  RTP Cluster is a front-end for multiple RTPproxies
  ☆41Updated last year
• sipcapture / sipgrep
  SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console
  ☆169Updated 2 months ago
• voxserv / freeswitch_conf_minimal
  Minimalistic FreeSWITCH configuration as a start for new projects
  ☆82Updated 8 years ago
• sangoma / switchy
  async FreeSWITCH cluster control
  ☆74Updated last year
• seanbright / mod_amd
  FreeSWITCH mod_amd
  ☆42Updated last year
• gnostix / freeswitch-monitoring
  Project to monitor FreeSwitch status and health
  ☆42Updated 8 years ago
• voxserv / voip_qos_probe
  Voice quality probe for end-to-end measurement of jitter and packet loss
  ☆41Updated 10 years ago
• xadhoom / mod_bcg729
  FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications
  ☆176Updated this week
• drachtio / drachtio-rtpengine-webrtcproxy
  Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)
  ☆43Updated 2 years ago
• pascomnet / asterisk_sbr
  Asterisk app_queue on steroids. Use a lua script to augment the queue strategy.
  ☆44Updated 3 years ago
• sipcapture / captagent
  100% Open-Source Packet Capture Agent for HEP
  ☆173Updated 3 months ago
• sipwise / sems
  SIP Express Media Server, very fast and flexible SIP media server
  ☆68Updated this week
• sipcapture / HEP
  HEP-EEP: Extensible Encapsulation Protocol (Specs & Technical Docs)
  ☆48Updated last month
• voxserv / freeswitch-perf-dialer
  FreeSWITCH dialer program for VoIP performance tests
  ☆46Updated 7 years ago
• ghrst / Kamailio-HA
  This repository contains Ansible playbooks and related files for an Active-Passive Kamailio auto-deployment using Pacemaker and Corosync.
  ☆55Updated 6 years ago
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆36Updated 7 years ago
• OpenSIPS / call-api
  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…
  ☆54Updated last year
• ianblenke / docker-rtpengine
  A docker SIPWise rtpengine first-class citizen implementation
  ☆30Updated 6 years ago
• mojolingo / sippy_cup
  Create SIP load test scenarios the easy way
  ☆223Updated 4 years ago
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆94Updated last year
• tic-ull / func_redis
  ☆16Updated 8 years ago
• florentchauveau / kamailio_exporter
  Prometheus exporter for Kamailio SIP server
  ☆60Updated last year
• caruizdiaz / kamailio-ws
  Kamailio scripts to call from websocket UA to classic UA, and vice versa.
  ☆41Updated 10 years ago
• 2600hz-archive / kazoo-configs
  Configurations for Kamailio, FreeSWITCH and all other components of KAZOO
  ☆19Updated 8 years ago
• sipcapture / homer-api
  HOMER 5: Back-End (API) DEPRICATED - use sipcapture/homer-app
  ☆27Updated 5 years ago
• asipto / secsipidx
  Secure SIP Identity Extensions - IETF STIR/SHAKEN - CLI/REST API tool and C library
  ☆50Updated 2 weeks ago
• OpenSIPS / opensips-ai-voice-connector-ce
  OpenSIPS AI Voice Connector Community Edition Platform
  ☆53Updated this week
• seanbright / asterisk-opus
  Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration
  ☆35Updated 8 years ago
• AGProjects / mediaproxy
  Media relay for RTP/RTCP and UDP streams
  ☆46Updated 3 months ago