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

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

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

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


Empathy is a messaging program which supports text, voice, and video chat and file transfers. It supports private and group chat and logs the complete conversation. It currently lacks privacy and encryption support. <img src="/AppImages/Article/empathy-img1.png" alt="" class="float-center">

Empathy is a messaging program which supports text, voice, and video chat and file transfers. It supports private and group chat and logs the complete conversation. It currently lacks privacy and encryption support. 

Empathy - Instant Messaging for GNOME

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Asterisk - IP telephony commuincation product suitable for call center

Linphone - Video SIP phone for Desktop and Mobile

Kamailio - The Open Source SIP Server

coturn - Free open source implementation of TURN and STUN Server

Empathy - Instant Messaging for GNOME

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Asterisk - IP telephony commuincation product suitable for call center

Linphone - Video SIP phone for Desktop and Mobile

Kamailio - The Open Source SIP Server

coturn - Free open source implementation of TURN and STUN Server

Empathy - Instant Messaging for GNOME

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