Ekiga - Software Phone and Video Cnferencing Application

•    C++

Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet. It provides Audio and Video free calls through the internet. It supports standard telephony features like Call Hold, Call Transfer, Call Forwarding, Call Histroy and Call Monitoring.

Asterisk - IP telephony commuincation product suitable for call center

•    C

Asterisk, converts an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services, including IP PBXs, VoIP gateways, call center ACDs and IVR systems. It is maintained by Debian VoIP Team.

phonertc - WebRTC for Cordova apps - No longer active

•    Objective-C

Important Note: Use GitHub issues only for bugs and feature requests! For any other questions, use the StackOverflow forum. Want to learn more? See the wiki.

Telephone - SIP softphone for Mac

•    Swift

Telephone is a VoIP program which allows you to make phone calls over the internet. It can be used to call regular phones via any appropriate SIP provider. If your office or home phone works via SIP, you can use that phone number on your Mac anywhere you have decent internet connection. Opus codec is optional.

Pion WebRTC - A pure Go implementation of the WebRTC API

•    Go

Pion is the Modern Stack for Web Real-Time Communication. It implements the WebRTC API. Spend more time building and less time learning a new API. Its feature include PeerConnection API support for DataChannels, Send/Receive audio and video, Renegotiation. It provides API with direct RTP/RTCP access, API also allows developer to pass their own packetizer, Opus, PCM, H264, VP8 and VP9 packetizer, Sender/Receiver Reports, Easy integration with x264, libvpx, GStreamer and ffmpeg and lot more.

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：

🚀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：

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✨

•    C

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✨

Riot - Messaging and Colloboration made easy

•    Javascript

Riot is built on top of Matrix. Matrix is an open network for secure, decentralized communication delivering a community of users,bridged networks, integrated bots and applications plus full end-to-end encryption. Riot allows teams to communicate across a wide range of collaboration apps. If some team members use Riot while others use IRC, Slack or Gitter, Riot will allow these team members to seamlessly work together. Riot offers the richest network of communication bridges.

SFLphone - Softphone for Linux

•    C++

SFLphone is a softphone for Linux. It is SIP compatible and supports Multi-way conferencing, Peer to peer calls, Video calls, Instant Messaging, Multi-line, Call transfer, Call hold, Call recording and lot more. It encrypts voice using SRTP protocol.

Mumble - High Quality Voice Chat Software

•    C++

Mumble is a cross-platform voice over IP application. It is a client - server application where multiple users could use the same server and all their communication is encrypted. It could be used while gaming. It does audio video recording.

Linphone - Video SIP phone for Desktop and Mobile

•    C

Linphone is an internet phone or Voice Over IP phone (VoIP), it helps to communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone is available for Linux, Windows, MacOSX, and for mobile phones: Android, iPhone, Blackberry. It is using SIP protocol and it is compatible with any voip operator using SIP for its network.

Kamailio - The Open Source SIP Server

•    C

Kamailio (successor of former OpenSER and SER) is an open source implementation of a SIP Signaling Server. SIP is an open standard protocol specified by the IETF. The core specification document is RFC3261.

holla - An abstraction over P2P video/voice/data connections using WebRTC

•    Javascript

Pipes a WebRTC video stream to a video element. el can be a string (id), jquery element, or dom node.

linphone-android - Linphone

•    C

Linphone is a free VoIP and video softphone based on the SIP protocol. Download the Android sdk (API 28.0.0 at max) with platform-tools and tools updated to latest revision, then add both 'tools' and 'platform-tools' folders in your path and the android-sdk folder to ANDROID_HOME environment variable.

Jitsi - Video Calls and Chat

•    Java

Jitsi provides Secure Video calls, Conferencing, Chat, Desktop sharing, Instant Messaging, File transfer support for your favorite OS and IM network. Jitsi lets you connect to Facebook, GoogleTalk, XMPP, Windows Live, Yahoo!, AIM, and ICQ so that you can chat to all your friends in the simplest possible way. It supports IPv6, Secure calls with zRTP, Call history, Call hold etc.

Matrix - Decentralized group communication

•    Python

Matrix is an open standard for decentralised communication, providing simple HTTP APIs and open source reference implementations for securely distributing and persisting JSON over an open federation of servers. It can be used for Decentralised Group Chat, WebRTC signalling, Internet of Things, VOIP etc.

MadelineProto - PHP client/server for the telegram MTProto protocol

•    PHP

This library can be used to easily interact with Telegram without the bot API, just like the official apps. It can login with a phone number (MTProto API), or with a bot token (MTProto API, no bot API involved!).

Fonos - An open-source alternative to Twilio

•    Javascript

Project Fonos is open-source telecommunications for the cloud. It helps VoIP integrators quickly deploy new networks and benefit from value-added services such as Programmable Voice, Messaging, and Video.

coturn - Free open source implementation of TURN and STUN Server

•    C

The TURN Server is a VoIP media traffic NAT traversal server and gateway. It can be used as a general-purpose network traffic TURN server and gateway, too. On-line management interface (over telnet or over HTTPS) for the TURN server is available.

re - Generic library for real-time communications with async IO support

•    C

libre is using GNU makefiles. Make and OpenSSL development headers must be installed before building. The libre project is using the BSD license.