Alternatives and similar repositories for esl

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

• englercj / node-esl
  FreeSWITCH ESL implementation for Node.js; implements the full Event Socket Library specified in: http://wiki.freeswitch.org/wiki/Esl
  ☆173Updated last year
• drachtio / drachtio-server
  A SIP call processing server that can be controlled via nodejs applications
  ☆283Updated this week
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆92Updated last year
• havfo / WEBRTC-to-SIP
  Setup for a WEBRTC client and Kamailio server to call SIP clients
  ☆295Updated last year
• sems-server / sems
  Sip Express Media Server
  ☆170Updated 2 months ago
• asipto / siremis-php
  Web Admin Interface for Kamailio
  ☆103Updated 4 months ago
• irontec / itsbc
  Irontec Tiny SBC. OpenSIPS & RTPEngine based micro SBC with Web Administration
  ☆36Updated 7 years ago
• drachtio / drachtio-srf
  drachtio signaling resource framework
  ☆188Updated last month
• drachtio / drachtio-fsmrf
  Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf
  ☆54Updated last week
• gmaruzz / saraphone
  SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…
  ☆172Updated last year
• daimoc / Kurento-SIP-GW
  ☆35Updated 3 years ago
• asterisk / cyber_mega_phone_2k
  ☆62Updated 3 years ago
• xadhoom / mod_bcg729
  FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications
  ☆173Updated 3 years ago
• ricardojlrufino / webphone-sip
  WebRTC SIP based VoIP client software (+chrome extension)
  ☆107Updated 7 months ago
• asterisk / node-ari-client
  Node.js client for ARI. This library is best effort with limited support.
  ☆271Updated last year
• BetterVoice / freeswitch-container
  This project can be used to deploy a FreeSWITCH server inside a Docker container. The container currently uses the latest stable release …
  ☆244Updated 8 years ago
• voiceip / oreka
  OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder
  ☆170Updated 5 months ago
• openvoips / OV500
  OpenSource Freeswitch & Kamailio Billing, rating and Routing Software
  ☆109Updated last year
• versatica / tryit-jssip
  New tryit-jssip application
  ☆88Updated last year
• sippy / b2bua
  Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
  ☆188Updated last month
• davehorton / drachtio
  DEPRECATED!! please use https://github.com/davehorton/drachtio-srf
  ☆69Updated 6 years ago
• asterisk / asterisk-external-media
  ☆83Updated 4 months ago
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆125Updated 3 years ago
• nadirhamid / asterisk-audiofork
  Stream Asterisk audio over Websockets
  ☆181Updated 10 months ago
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆119Updated last week
• drachtio / drachtio-rtpengine-webrtcproxy
  Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)
  ☆43Updated last year
• collecttix / ctxSip
  ctxSip is a Javascript based SIP client that uses WebRTC and WebSockets to connect to your SIP server. The UI is designed to be launched …
  ☆325Updated last year
• dOpensource / dsiprouter
  UI Interface for implementing Kamailio to provide PBX Hosting and SIP Trunking Services
  ☆211Updated last week
• 2600hz / kazoo-configs-kamailio
  ☆35Updated last month
• unispeech / asterisk-unimrcp
  UniMRCP modules for Asterisk
  ☆49Updated last year