voip

There are 36 repositories under voip topic.

• pion / webrtc

  Pure Go implementation of the WebRTC API

  audiogogolanglive-videoortcp2ppionpion-webrtcrtpsrtpstreamingvideovoipwasmwebassemblywebcamwebrtcwebrtc-apiwebrtc-librarieswebrtc-server
  Language:Go 9509

• wildfirechat / im-server

  即时通讯(IM)系统

  chatchat-applicationchatappchatserverimimserverinstant-messagingmessagermessagingmqtttalkvoipwebsocketwechatweixin
  Language:Java 6697
• ejabberd

  processone / ejabberd

  Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)

  erlangxmppjabberchatmessagingsipvoipiotpubsubbrokergroupchatstunturnejabberdjabber-servermqttbrokersserversip-server
  Language:Erlang 5238
• fonoster

  fonoster / fonoster

  🚀 The open-source alternative to Twilio

  cloud-communications-platformtwilioucaaskubernetesvoipprogrammable-voicewebrtctelephonycloudcpaascustomer-engagementjavascriptnodejstypescriptcustomer-experience
  Language:JavaScript 5042

• mumble-voip / mumble

  Mumble is an open-source, low-latency, high quality voice chat software.

  voicechatvoipaudioclientservergamingopen-sourcelinuxmacosquality-voice-chatwindowscross-platformcmakehacktoberfest
  Language:C++ 4855

• qTox / qTox

  qTox is a chat, voice, video, and file transfer IM client using the encrypted peer-to-peer Tox protocol.

  qtoxtoxc-plus-plusqtp2pencryptiondistributedvoipaudiovideomessagingprivacycommunication
  Language:C++ 4120

• starrtc / starrtc-android-demo

  🚀starRTC，即时通讯(IM)系统，免费IM系统（含单聊，群聊，聊天室，文件传输），免费一对一视频聊天，VOIP，语音对讲（回音消除），直播连麦，视频直播，RTSP拉流，RTMP推流，webRTC服务端，在线教育，白板，小班课，在线会议，视频会议，视频监控，局域网直连（无需服务器），兼容webRTC, 支持webRTC加速，P2P高清传输，安卓、iOS、web互通，支持门禁对讲，可视对讲，电视盒子，树莓派，海思，全志，任天堂switch，云游戏，OTT设备，物联网平台，C语言自研方案，支持二次开发成类微信，类映客等APP，✨万水千山总是情，来个star行不行✨，更多示例请访问：

  freeimvoiplivevideovideo-chatwebrtcjanuskurentojitsilicodecoturnraspberry-pigroupchatchatroom
  3190

• flutter-webrtc / flutter-webrtc

  WebRTC plugin for Flutter Mobile/Desktop/Web

  androiddesktopflutteriossipvoipwebwebrtcwebrtc-plugin
  Language:Java 3112

• starrtc / starrtc-server

  免费IM系统，IM即时通信消息系统(含一对一文字聊天，群聊，聊天室)，免费一对一voip实时通话，录屏，webrtc服务端，免费直播连麦，互动直播，视频直播，RTSP拉流，RTMP推流，语音对讲，免费在线会议，视频会议等服务端程序，支持物联网平台，✨万水千山总是情，来个star行不行✨

  freeinstant-messaginglivevoipmeeting
  Language:C 2458

• wildfirechat / android-chat

  开源即时通讯(野火IM)系统Android端

  chatimvoip
  Language:Java 2210

• danog / MadelineProto

  Async PHP client/server API for the telegram MTProto protocol

  mtprotobotphpmadelineprototelegrameasytelegram-apiluasecret-chatsproxyvoipamphpstickersinline-botscallsasyncmtproto-apihacktoberfest
  Language:PHP 2005

• webrtc-rs / webrtc

  A pure Rust implementation of WebRTC

  rtcwebrtcvoiprust
  Language:Rust 1995

• signalwire / freeswitch

  FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.

  freeswitchiotopen-sourcesignalwiresipsource-codetelephonyvoipwebrtc
  Language:C 1874
• kamailio

  kamailio / kamailio

  Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -

  kamailiosipvoipwebrtcvolteiottelephony
  Language:C 1652

• onsip / SIP.js

  A simple, intuitive, and powerful JavaScript signaling library

  javascriptnodejssipsipjstypescriptvoipwebrtc
  Language:TypeScript 1521

• pjsip / pjproject

  PJSIP project

  pjsipsiprtpnat-traversalvoipandroidiosandroid-ndk
  Language:C 1206

• sipcapture / homer

  HOMER - 100% Open-Source SIP / VoIP / RTC Packet Capture & Monitoring

  monitoringtroubleshootingvoiprtcpcapflowcallflowcdrcorrelationcapture-agentpacket-capturepacket-snifferanalyticswebrtcencapsulationsipstatisticsopensipskamailiohep
  Language:Shell 1161

• baresip / baresip

  Baresip is a modular SIP User-Agent with audio and video support

  libresiptelephonyvoipwebrtc
  Language:C 1002

• wildfirechat / ios-chat

  开源的即时通讯(野火IM)系统

  imchatchat-roomvoipmqttwechatweixin
  Language:Objective-C 988
• mirotalksfu

  miroslavpejic85 / mirotalksfu

  🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.

  sfuwebrtcmeetingself-hostedvoipcollaborationchatdockervideoaudiowebsocketreal-timejavascriptmediasoupvideo-communicationscalable-video-conferenceswebsitegoogle-meet-clonesocket-iogoogle-meet
  Language:JavaScript 958

• 64characters / Telephone

  SIP softphone for Mac

  sipvoipsoftphonemac-appphone-calls
  Language:Swift 952

• fonoster / routr

  Routr: Next-generation SIP Server

  sipwebrtcvoipinstant-messagingimsasteriskkamailiofreepbxfreeswitchfusionpbxhackoctoberfesttwilio
  Language:JavaScript 933

• BelledonneCommunications / linphone-android

  Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)

  linphonelinphone-androidvoipandroidvideosipcallconferencephoneliblinphone
  Language:Kotlin 896

• sipsorcery-org / sipsorcery

  A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.

  voipsipwebrtcrtpstunsdpcsharpdotneticevideo-callscommunicationsunityunity3d
  Language:C# 820
• sipvicious

  EnableSecurity / sipvicious

  SIPVicious OSS has been around since 2007 and is actively updated to help security teams, QA and developers test SIP-based VoIP systems and applications.

  sipvoippassword-crackersvwarsvcracksvmapwar-dialsvcrashsecuritysecurity-toolsaudit-sip
  Language:Python 662

• react-native-webrtc / react-native-callkeep

  iOS CallKit framework and Android ConnectionService for React Native

  androidcall-kitcallkitconnection-serviceconnectionserviceiosreact-nativevoipwebrtc
  Language:Java 652

• BelledonneCommunications / linphone-iphone

  Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of linphone-iphone (git://git.linphone.org/linphone-iphone.git)

  linphoneiphonevoipphonesipvideocallconferencechatmessage
  Language:Objective-C 503

• creytiv / re

  Generic library for real-time communications with async IO support

  sipvoipturnicewebrtc
  Language:C 480

• Tinywan / WebRTC-tutorial

  :books: WebRTC 中文教程

  webrtcsrtpwebrtc-udprtcpeerconnectionv8webrtc-videoh246v9rtcsignalp2psipvoip
  448

• tiredofit / docker-freepbx

  Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP

  dockerfreepbxasteriskmysqlcdrvoice-over-ipvoipsangomadigiums6overlayphonesippjsipiax
  Language:Dockerfile 448

• pion / mediadevices

  Go implementation of the MediaDevices API.

  webrtcdrivercodecvideo-callaudio-callmachine-learningface-recognitionlivestreamrtpvoipp2pgogolangmediadevicesmediadevices-apistreaming
  Language:Go 371
• tgcalls

  MarshalX / tgcalls

  Voice chats, private incoming and outgoing calls in Telegram for Developers

  pythontelegramwebrtcvoipcallsvideo-callvoice-chatgroup-chatpytgcallstgcallspyrogramliblibraryasyncasynchronousrealtimeffmpegvoicetelethonmtproto
  Language:Python 370

• sm0svx / svxlink

  Advanced repeater system software with EchoLink support for Linux including a GUI, Qtel - the Qt EchoLink client

  c-plus-plusham-radiocommunicationvoip
  Language:C++ 341

• starrtc / starrtc-web-demo

  一对一voip视频聊天，直播连麦，多人视频会议，在线会议web演示：

  livemeetingvoipwebrtcwebrtc-demoswebrtc-javascript-library
  Language:JavaScript 332

• meliht / Mr.SIP

  SIP-Based Audit and Attack Tool

  sipvoipauditpentestdosip-spoofingattackddos
  Language:Python 324

• discordjs / voice

  Implementation of the Discord Voice API for discord.js and other JS/TS libraries

  voicediscordvoipdiscordjsudptcpaudiodiscord-bothacktoberfest
  Language:TypeScript 307