Alternatives and similar repositories for asterisk-cpp

Users that are interested in asterisk-cpp are comparing it to the libraries listed below

• BelledonneCommunications / bcg729
  Linphone.org mirror for bcg729 (git://git.linphone.org/bcg729.git)
  ☆130Updated last year
• sems-server / sems
  Sip Express Media Server
  ☆180Updated last month
• sippy / b2bua
  Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
  ☆193Updated last month
• asipto / siremis-php
  Web Admin Interface for Kamailio
  ☆103Updated 10 months ago
• OpenSIPS / opensips-cp
  A Web Control Panel Application for the OpenSIPS, which is intended for both system and user provisioning. It features more than 18 tools…
  ☆127Updated last month
• opentelecoms-org / ipp_audio_codecs
  Intel IPP audio codecs including G.729 and G.723.1 adapted for Asterisk
  ☆63Updated 13 years ago
• unispeech / asterisk-unimrcp
  UniMRCP modules for Asterisk
  ☆51Updated last year
• sipwise / sems
  SIP Express Media Server, very fast and flexible SIP media server
  ☆70Updated last week
• tomek-o / tSIP
  SIP softphone
  ☆180Updated last week
• voiceip / tinyphone
  Minimalist Windows / OSx / Linux SIP Softphone with API Control
  ☆152Updated 9 months ago
• openvoips / OV500
  OpenSource Freeswitch & Kamailio Billing, rating and Routing Software
  ☆114Updated last year
• caruizdiaz / kamailio-ws
  Kamailio scripts to call from websocket UA to classic UA, and vice versa.
  ☆44Updated 11 years ago
• daniele77 / aricpp
  Asterisk ARI interface bindings for modern C++
  ☆34Updated 10 months ago
• voiceip / oreka
  OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder
  ☆180Updated this week
• freeswitch / spandsp
  SpanDSP is a low-level signal processing library that modulates and demodulates signals commonly used in telephony, such as the "noise" g…
  ☆184Updated 5 months ago
• BelledonneCommunications / belle-sip
  Linphone.org mirror for belle-sip (git://git.linphone.org/belle-sip.git)
  ☆73Updated last week
• sippy / rtpproxy
  The RTPproxy is a high-performance software proxy for RTP streams that can work together with Sippy B2BUA, Kamailio, OpenSIPS and SER.
  ☆460Updated last month
• freeswitch / sofia-sip
  Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.
  ☆316Updated last month
• chakrit / pjsip
  Unofficial mirror/fork of http://svn.pjsip.org/repos/pjproject/trunk/ — check the Wiki for more information.
  ☆54Updated 11 years ago
• xadhoom / mod_bcg729
  FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications
  ☆177Updated 4 months ago
• versatica / tryit-jssip
  tryit-jssip application
  ☆88Updated 4 months ago
• sipcapture / sipgrep
  SIPGREP: Display and Troubleshoot SIP signaling over IP networks in console
  ☆171Updated 3 months ago
• BelledonneCommunications / flexisip
  Linphone.org mirror for flexisip (git://git.linphone.org/flexisip.git)
  ☆174Updated this week
• DoubangoTelecom / telepresence
  the open source SIP TelePresence system
  ☆147Updated 6 years ago
• havfo / WEBRTC-to-SIP
  Setup for a WEBRTC client and Kamailio server to call SIP clients
  ☆308Updated 2 years ago
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆95Updated last year
• meetecho / asterisk-opus
  Opus (transcoding) and VP8 (passthrough) support for Asterisk, needed for a better WebRTC integration
  ☆81Updated 10 years ago
• BarGabriel / SipServer
  A simple SIP server (proxy) for handling VoIP calls based on SIP using C++ on Windows & Linux platforms.
  ☆94Updated last year
• asterisk / cyber_mega_phone_2k
  ☆62Updated 3 years ago
• dandugula / mod_ipp_g729
  Hacked FreeSWITCH-G729 speech codec using Intel® Integrated Performance Primitive.
  ☆22Updated 12 years ago