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...This project is a WebRTC softphone, and communication is achieved via SIP over a socket. Leading PBX systems like Asterisk or Freeswitch support socket connections. If your provider does not offer this feature, consider using a SIP proxy such as Kamailio, Opensip or Routr. The softphone is internally configured using a JSON definition. The configuration file can be loaded from either a server or a local file. 008 reads the file only once.

ICTAgent is a advance webphone offered as chrome extesion and when installed it will convert all available contacts as click to dial. This extension will embed WebRTC based VoIP Phone in browser which will remain connected to the main server, and will be responsible for inbound and outbound calls, transfer calls, sending DTMF as well sending Fax. It can harvest contacts from the random pages. Agent can access contents, contacts from a single application. Browser can automatically fetch...

MagnusBilling is a VoIP server system that brings together the best IP Billing Telephony software in the market, creating a comprehensive, flexible and superior tool. Enterprises that need the freedom to build their own VoIP server. You will be able to chose your termination, giving your business larger flexibility and autonomy. Many VoIP distributors use the best of each VoIP server, termination to be able to be more competitive and to have better price and quality. Magnus Sever is...

OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list

VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).

The G.O.N.E. is a softphone (or soft phone) running over the web, fully multi-plataform, it implements the SIP protocol, and is built to work on any SIP server, like Asterisk, and others. GONE will work on a complex sistem, but this will be showed a bit