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Ekiga - Software Phone and Video Cnferencing Application
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.
voip skype-alternative internet-phone sip chatAsterisk - IP telephony commuincation product suitable for call center
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.
communication framework ip-telephony pbx voip call-centerphonertc - WebRTC for Cordova apps - No longer active
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.
cordova webrtc mobile video video-conferece voice voice-over-ip voip ecosystem:cordova cordova-android cordova-ios cordova-browserRiot - Messaging and Colloboration made easy
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.
instant-messaging chat secure-chat mobile-apps group-chat colloboration voip video-conferenceSFLphone - Softphone for Linux
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.
voip skype-alternative internet-phone chat sip softphoneMumble - High Quality Voice Chat Software
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.
voip skype-alternative internet-phone chat voip-server chat-server softphoneLinphone - Video SIP phone for Desktop and Mobile
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.
voip skype-alternative internet-phone sip xmpp chatKamailio - The Open Source SIP Server
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.
kamailio sip voip webrtc volte iot telephony sip-serverlinphone-android - Linphone
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.
linphone linphone-android voip android video sip call conference phone liblinphoneJitsi - Video Calls and Chat
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.
video-conference voip skype-alternative internet-phone sip xmpp chatMatrix - Decentralized group communication
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.
voip webrtc skype-alternative video-conference slack-alternative instant-messaging chatMadelineProto - PHP client/server for the telegram MTProto protocol
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!).
mtproto bot madelineproto telegram easy telegram-api secret-chats audio proxy voipQutecom - PC to PC Video and Voice calls
QuteCom is a SIP compliant VOIP client. It allows users to speak to other users of SIP compliant VoIP software. It helps to make calls to landlines, cellphones and allows to send SMS and make video calls.
voip skype-alternative internet-phone chat sip softphoneBlink - easy to use SIP client
Blink is a GUI for Mac, Windows and Linux built on top of SIP SIMPLE client SDK. It uses SIP (Session Initiated Protocol) for VOIP and Instant Messaging. It supports Multi-party conferencing, Missed Calls, Dialing out SIP addresses and telephone numbers. Currently File Transfer and Desktop sharing is supported only in Mac.
voip skype-alternative internet-phone sip chat softphonehomer - HOMER - 100% Open-Source SIP / VoIP Packet Capture & Monitoring
HOMER is a robust, carrier-grade, scalable SIP Capture system and VoiP Monitoring Application offering HEP/EEP, IP Proto4 (IPIP) encapsulation & port mirroring/monitoring support right out of the box, ready to process & store insane amounts of signaling, logs and statistics with instant search, end-to-end analysis and drill-down capabilities for ITSPs, VoIP Providers and Trunk Suppliers using SIP signaling protocol. Powered at the core by SIPCAPTURE Module for industry-standard Kamailio or OpenSIPS, HOMER provides virtually unlimited scope for granular capture configuration either stand-alone or using our companion Capture Agent Project.
monitoring troubleshooting voip rtc pcap flow callflow cdr correlation capture-agent packet-capture packet-sniffer analytics webrtc encapsulation sip statistics opensips kamailio hepSIPSorcery
At the heart of this project is an open source C# SIP stack. In addition there are related projects including a variety of SIP Servers such as a Proxy, Registrar and more. The SIP stack was launched in 2006 and is used in a number of live services particularly www.sipsorcery.com.
0991 1st my sip sip-server voip
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