Search Results for "sofia-sip"

...This agent utilizes embedded artificial intelligence models operating directly on the softphone, ensuring efficiency and reduced operational costs. 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.

SipLine is a modern, lightweight SIP softphone for Windows 10/11 built with .NET 9 WPF. It provides enterprise-grade VoIP communication with a focus on performance, security, and extensibility. Features Ultra-Fast: Startup in <0.5s, minimal CPU usage HD Audio: Opus and G.711 codecs with adaptive jitter buffer Security: TLS transport, SDES-SRTP encryption, Windows DPAPI credential protection Multi-Account: Up to 5 simultaneous SIP accounts Plugin SDK: C# SDK for custom integrations (CRM, call recording, automation) Plugin Marketplace: Community and premium extensions Headset Support: Jabra, Poly, Sennheiser, Logitech with HID controls Call Quality: Real-time MOS Score, Jitter, Packet Loss and RTT monitoring Enterprise: Silent MSI installer, GPO support, JSON centralized config Compatibility: 3CX, Asterisk, FreePBX, OVH, Twilio, RingCentral, and more Language packages : https://sipline.feelautom.fr/languages

jfPhone is a VoIP/SIP SoftPhone for Desktops.

VitalPBX is a free telephone and communications PBX system for companies. It is a complete platform that can be installed on the physical hardware on the site or as a hosted application. VitalPBX acts as the upper layer interface for the Linux base and then Asterisk (one of the most popular communication toolkits in the world). For this reason, VitalPBX is the graphic user interface between you and the complex world of modern communications. VitalPBX will help you implement a secure...

jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange.

Vodia PBX has included a complete suite of business telephony features for on-premise and cloud-based telephony deployments. Sign up at https://vodia.com/en/register and receive your free Vodia license.

Cross-platform SIP softphone

wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .

Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).

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

IMS-Communicator is an IMS softphone based on the old version of the sip-communicator java project, implemented on top of the JAIN-SIP stack and the Java Media Framework API. It can be used with Open IMS Core (http://www.openimscore.org).

PartiSIPation is a sip user agent which has an easy replaceable gui. So it can be used as a softphone as well as it can be integrated in any kind of application, e.g. online games.

This project is a SIP based softphone developed as a Java applet that can be embedded into a website. The applet has a simple user interface and is intended to be used by journalists to make and deposit audio recordings via a SIP proxy.

Rabboni VOIP is an Open Source Cross Platform Softphone developed with trolltech QT LIBRARY it uses PJSIP - Open Source SIP Stack

A command line SIP/H323 softphone capable of sending and receiving audio files as well as sending out of band DTMF digits. Supports a BNF format configuration language for scripting call scenarios. Useful for example system testing.

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).

Among the functionality: Include a large variety of codecs (G711, GSM, and SPEEX) - Protocol SIP - Other technical functionalities the support of DTMF (tonalities) although support ENUM (to employ numbers of SIP instead of the addresses of SIP).

HATEFNA هاتفنا .....http://www.hatefna.org ..... (PC&Mobile)IP SoftPhone over SIP and ASTERISK using JAVA for PC and Symbian C++ for Symbian Mobiles Phones ..... برمجيات للمهاتفة ببروتوكول الإنترنت

Softphone SIP com voz, vídeo e Mensagem Instantânea. DTMF, Nat-Traversal, Suporte a Skin(Macromedia Flash), SMS, Historico de Chamadas, Presença, Hold, Auto-Answer e etc. Sip Softphone with Voice, Video and IM.

SIP SoftPhone desenvolvido em Java baseado na JAIN-SIP, JMF e Sip-Communicator 1.0. Possui também o envio de SMS, Agenda, NAT Traversal, DTMF Send/Receive, CallTo e etc. A Java SIP SoftPhone based in JAIN-SIP, JMF and Sip-Comminicator 1.0.

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

Softphone web open-source construído com PHP + Swoole, focado em sinalização SIP, mídia RTP, ponte de mídia em tempo real, integração com libspech e arquitetura sem WebRTC.