Search Results for "ip"

Open Phone Abstraction Library (OPAL) is a C++ multi-platform, multi-protocol library for Fax, Video & Voice over IP and other networks. Also included is the Portable Tool Library (PTLib) which is a C++ multi-platform abstraction library and collection o

Open Source - GPLv3 inc images. *** PLEASE NOTE: There are now 2 seperate versions here. *** One is Pre Firefox 57. The other is Post Firefox 57. *** For those providing mirrors, please enable your users to realize this. Vidalia Based, Tor as a Service Solution. MicroSip: enables FREE PC to PC video calling with no account sign-up and no middleman server. WASTE: enables FREE Conference VoIP, chat, file transfer and support. *** AI Powered *** Tor/i2p: enables safer...

New version 1.2. This is the code to STUNTMAN - an open source STUN server and client code by john selbie. Compliant with the latest RFCs including 5389, 5769, and 5780. Also includes backwards compatibility for RFC 3489. ICE and WebRTC ready. Version 1.2 compiles on Linux, MacOS, BSD, and Solaris. Supports the STUN protocol on both UDP and TCP for both IPv4 and IPv6. Windows binaries are also provided. Additional features are in development. This is a mirror of the code on...

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.

Comprehensive business management system The full-cycle automation ERP system - [ Full Freeware ] IP PBX - Complete synchronization with CRM system - Call accounting for PBX (PBX billing) - Phone call control and monitoring - Phone recording systems Accounting, Financial management - Trade management, warehouse - Management and tax accounting Corporate information system - Client relations management system (CRM) - Project, tasks, orders management - On-line conferences (forums) - Instant messenger, chat, sending SMS Staff management system [HRM] - Personnel records, documents circulation - Time and Attendance tracking system - Computer usage monitoring (time tracking) - Work time accounting (five levels) - Accrual and deduction system - Fines system (two levels) - Salary calculation Access control systems Management of access rights to the premises Video surveillance system (IP,Analog) Photofixation, Videofixation Technological process automation

Activa brings the Asterisk IP PBX to the call center. Built on top of Asterisk, Activa components enable successful call center implementations adding value in areas such as computer telephony, screenpop&click2dial, agent control, automatic dialing...

Vem ai o Novo Disc-OS Aguardem.. Uma Empresa Orkestral Estamos Voltando! Modificações: Asterisk 21 Nova Interface UI Centos 8 Nodejs Disc-OS é uma distribuição de um PABX IP baseado em software livre. Desenvolvido para o mercado brasileiro com interfaces em português, de fácil instalação e configuração, contendo Linux customizado, software Asterisk 1.4 e configurador Disc.

The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. Thanks to its simplicity, SIP messages are often used in creative ways for which these were not originally designed (e.g. using periodical OPTIONS packets as NAT keep-alive instead of using STUN or TURN) and thus SIP traces of the captured traffic often contain "useless" traffic from the actual signalization point of view. ...

The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. A good way to design optimization techniques for SIP deployment would be to analyze SIP traffic from existing networks. However, publicly available analyses of SIP traffic are rare and thus not a lot of knowledge exists about typical behavior of a SIP server (as opposed to, for example, HTTP servers). Our aim is to promote analysis of real SIP-server behavior. ...

...It uses following protocol stack: MAP (moForwardSM and mtForwardSM) TCAP (Begin with Invoke + End with ReturnResultLast) SUA (SCCP User Adaptation): ASP UP/ASP ACTIVE + ACK's and CLDT messages only SCTP (Linux Kernel - SCTP implementation) IP Current implementation is tested in SAMK NGN [http://www.samk.fi/ngn] laboratory's 3G mobile network (real Nokia MSC/VLR network hardware with UTRAN RLC and node-B). Thesis URL: http://urn.fi/URN:NBN:fi:amk-2013052210011

What is t38modem? From your application view point it's a fax/voice modem pool. From IP network view point it's a H.323/SIP endpoint with T.38 fax support. From your view point it's a gateway between an application and IP network. Works with HylaFAX.

FreeSentral is a full IP PBX consisting of a Linux Distribution, an IP PBX and a Web Graphical User Interface for easy configuration.

Simu server project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications.

This is a voice over ip system.

VVOIP is voice and video over IP is a P2P application without any protocol like SIP (no server is required) etc.. application. It use decent codecs for video and voice encrypted streams over IP. Moreover it is also comptiable with VoIP modems, you'd con

R.V.A is a Voice over IP radius accounting system. It receives radius accounting packets from Cisco routers and places them in a database for call detailed analysis.

MyPhone is Voice&Video IP Telephony Chat client. It is H323 protocol compatible, based on OpenH323 project (www.openh323.org). Client is written in VC++, using MFC/PWLib libraries, compatible with any reasonable M$Windows version. Freeware&OpenSource

InterPhone is meant to be an open source project for setting up a server that will provide services such as phone-to-phone or pc-to-phone using VOIP (Voice over IP)

FreePhone is an easy to use IP phone application. FreePhone supports major platforms and protocols. Features include Conference calling, Voice mail and Chat rooms. A SDK is available for you to make additional features.

Open source Call Center software, based on H.323/SIP protocols. Contains Nauphone(Software multi-channel IP Phone, with conferences and ability to integrate with other systems(CRM)) and NauLib(VoIP Library based on OpenH323 and Vovida SIP)

PNX system develops the middle-ware source code to glue Asterisk with a number of powerful telephony products such as: 1) OpenH323 H.323 stack 2) Vovida SIP stack 3) Bayonne Voice Automation Platform The advantages? An advanced IP PBX supporting the wide

An H.323 compliant Linux server that accepts calls from intenet "phone" software and routes them out onto the public telephone network via a voice modem.

Siphon will ultimately be a Software Voice-Over-IP phone using the SIP protocol. It will support quicknet cards as well as traditionnal soundcards. It is developped for Linux (gtk) and the win32 platform.