Alternatives and similar repositories for react-verto-communicator

Users that are interested in react-verto-communicator are comparing it to the libraries listed below

• sems-server / sems
  Sip Express Media Server
  ☆178Updated last week
• drachtio / drachtio-siprec-recording-server
  SIPREC recording server based on drachtio and rtpengine
  ☆95Updated last year
• daimoc / Kurento-SIP-GW
  ☆35Updated 3 years ago
• havfo / WEBRTC-to-SIP
  Setup for a WEBRTC client and Kamailio server to call SIP clients
  ☆307Updated 2 years ago
• freeswitch / mod_janus
  ☆21Updated 5 months ago
• sippy / b2bua
  Sippy B2BUA is a RFC3261-compliant Session Initiation Protocol (SIP) stack and Back-to-back user agent (B2BUA) server software.
  ☆192Updated this week
• BelledonneCommunications / bcg729
  Linphone.org mirror for bcg729 (git://git.linphone.org/bcg729.git)
  ☆130Updated last year
• xadhoom / mod_bcg729
  FreeSWITCH G.729 module using the opensource bcg729 implementation by Belledonne Communications
  ☆177Updated 3 months ago
• drachtio / drachtio-rtpengine-webrtcproxy
  Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)
  ☆44Updated 2 years ago
• versatica / tryit-jssip
  tryit-jssip application
  ☆89Updated 2 months ago
• nimbleape / dana-the-stream-gatekeeper
  React based front-end demo for Asterisk's SFU
  ☆30Updated 2 years ago
• sippy / rtp_cluster
  RTP Cluster is a front-end for multiple RTPproxies
  ☆41Updated last year
• drachtio / drachtio-fsmrf
  Drachtio freeswitch-based media resource function -- http://davehorton.github.io/drachtio-fsmrf
  ☆55Updated 2 months ago
• altanai / opensipsexamples
  sample configurations and settings for opensips for various use-cases
  ☆63Updated 6 years ago
• giavac / janus_freeswitch_integration
  Integrating Janus and FreeSWITCH for an heterogeneous conferencing system
  ☆22Updated 6 years ago
• voiceip / oreka
  OpenSource G711, G722, G729, Opus & Other Format VoIP SIP Recorder
  ☆177Updated 3 months ago
• kamailio / kamailio-wiki
  Kamailio Wiki with content in markdown format
  ☆42Updated last month
• asterisk / cyber_mega_phone_2k
  ☆63Updated 3 years ago
• altanai / kamailioexamples
  VoIP architectures and use cases involving Kamailio SIP Server and its modules includes RTPEngine
  ☆130Updated 3 years ago
• freeswitch / sofia-sip
  Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.
  ☆311Updated 4 months ago
• navrotskyj / vertoPhone
  ☆23Updated 6 months ago
• asipto / siremis-php
  Web Admin Interface for Kamailio
  ☆103Updated 8 months ago
• jchavanton / voip_patrol
  VoIP signaling and media test automation
  ☆122Updated 3 weeks ago
• saghul / sipp-scenarios
  SIPp scenarios I use for testing SIP stuff
  ☆300Updated last year
• AGProjects / mediaproxy
  Media relay for RTP/RTCP and UDP streams
  ☆47Updated 6 months ago
• surendratiwari3 / KamInboundSIP
  This Project will provide the inbound sip using that we can route did call to customer ip or customer Phone number
  ☆24Updated 8 years ago
• kamailio / kamcli
  Kamailio Command Line Interface Control Tool
  ☆56Updated last month
• pbertera / SIPp-by-example
  SIPp examples
  ☆144Updated 2 years ago
• gmaruzz / saraphone
  SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. Sar…
  ☆182Updated last year
• havfo / multiparty-meeting-sipgw
  Multiparty-meeting (mediasoup) SIP gateway using Kurento
  ☆28Updated 6 years ago