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. 

 Asterisk supports a wide range of protocols for the handling and transmission of voice over traditional telephony interfaces including H.323, Session Initiation Protocol, Media Gateway Control Protocol, and Skinny Client Control Protocol. Asterisk merges voice and data traffic seamlessly across disparate networks. 

 It has drivers for VoIP protocols and PSTN interface cards and devices. It routes both inbound and outbound calls. Call details and duration are logged for billing. Calls could be recorded and monitored. It recognizes speech and converts to text. LDAP integration is provided for corporate security. It stores all information in a relational database. 

 Asterisk has support of IVR, Conference bridge, Automated attendant, Call Queuing, Call Recording, Call Retrieval, Caller ID Blocking, SMS Messaging, VoIP Gateways and lot more. 
	
 Hardware requirement: Asterisk requires no additional hardware. It turns an ordinary computer into a communications server. It runs on GNU/Linux, OpenBSD, FreeBSD, and Mac OS X. 
 
 API Support: External call management is possible in any programming or scripting language through Asterisk Gateway Interface.

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.

Asterisk - IP telephony commuincation product suitable for call center

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. 
 It supports HD sound quality audio and DVD quality video. It is interoperable with PBX like Asterisk. <img src="/AppImages/Article/ekiga_img1.jpg" alt="" class="float-center">

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. 

Ekiga - Software Phone and Video Cnferencing Application

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.

Pion WebRTC - A pure Go implementation of the WebRTC API

Fonoster is the Open Source Twilio Alternative. It helps tp engage with your customer with voice or messaging with a single easy-to-use platform. 
 
Its features include: 
<ul><li>Cloud initialization with Cloud-Init</li><li>Multitenancy</li><li>Easy deployment of PBXs functionalities</li><li>Programmable Voice Applications</li><li>NodeJS SDK</li><li>Web SDK</li><li>Support for Amazon Simple Storage Service (S3)</li><li>Secure API endpoints with Let's Encrypt</li><li>Authentication with OAuth2</li><li>Authentication with JWT</li><li>Role-Based Access Control (RBAC)</li><li>Plugins-based Command-line Tool</li><li>Support for Google Speech API</li><li>Experimental support for Cloud Functions</li><li>Experimental support for Secret management</li></ul>

Fonoster is the Open Source Twilio Alternative. It helps tp engage with your customer with voice or messaging with a single easy-to-use platform. 

Fonoster - The open-source alternative to Twilio

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. It provides Liblinphone, a powerful SIP VoIP video SDK that anyone can use to add audio or video call capabilities to an application. It supports IPv6, Webcam, Addressbook and lot more. 
 <img src="/AppImages/Article/linphone-img1.jpg" alt="" class="float-center">

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.

Linphone - Video SIP phone for Desktop and Mobile

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. <img src="/AppImages/Article/mumble-img1.jpg" alt="" class="float-center">

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.

Mumble - High Quality Voice Chat Software 

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. <img src="/AppImages/Article/sfl-img1.png" alt="" class="float-center">

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. 

SFLphone - Softphone for Linux

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. It is well integrated with Microsoft Outlook, Apple Address Book, Google Contact, LDAP directories. <img src="/AppImages/Article/jitsi-img1.jpg" alt="" class="float-center">

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.

Jitsi - Video Calls and Chat

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.

Riot - Messaging and Colloboration made easy 

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.
 
The Kamailio SIP server is designed for scalability, targeting large deployments (e.g. for IP telephony operators or carriers, which have a large subscriber base or route a big volume of calls), but can be also used in enterprises or for personal needs to provide VoIP, Instant Messaging and Presence. Kamailio is well known for its flexibility, robustness, strong security and the extensive number of features.
 
Among the powerful features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached; Json and XMLRPC control interface, SNMP monitoring.

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 - The Open Source SIP Server

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.

Matrix - Decentralized group communication

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.



coturn - Free open source implementation of TURN and STUN Server


Element is a messaging app that lets you chat securely with your family and friends. The communication is default end-to-end encrypted, cross-signed device verification and decentralisation to choose who you trust with your data. Element is unique. You own your data and decide where it lives. Element lets you talk securely to anyone, regardless of their app. It supports unlimited voice, video and messaging. It is built on Matrix.

Element - Secure and independent communication, connected via Matrix

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.
 <img src="/AppImages/Article/blink-img1.jpg" alt="" class="float-center">

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.

Blink - easy to use SIP client

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. 

Qutecom - PC to PC Video and Voice calls

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