Search Results for "sip"
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Showing 154 open source projects for "sip"
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Cloud tools for web scraping and data extractionAutomate web data collection with cloud tools that handle anti-bot measures, browser rendering, and data transformation out of the box. Extract content from any website, push to vector databases for RAG workflows, or pipe directly into your apps via API. Schedule runs, set up webhooks, and connect to your existing stack. Free tier available, then scale as you need to.
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Ovmeet
Video conferencing and collaboration platform
...Supports WebRTC, RTMP, RTSP, SIP protocols, enabling interoperability with streaming devices, hardware endpoints, and different media sources. -
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008
Open-source event-driven AI powered Softphone
...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. -
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. ... -
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jfPhone
VoIP/SIP SoftPhone (formerly known as jphonelite)
jfPhone is a VoIP/SIP SoftPhone for Desktops. -
Nonprofit Budgeting SoftwareMartus' collaborative and easy-to-use budgeting and reporting platform will save you hundreds of hours each year. It's designed to make the entire budgeting process easier and create unlimited financial transparency.
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IssabelPBX
Issabel PBX - Unified Communications
Open Source and Unified Communications partners created a new platform based on an Elastix® fork (currently purchased by 3CX) to provide the community with continuity, peace of mind and support needed to continue with their PBX and operation developments. Contribute to the funding of Issabel on https://www.patreon.com/issabel -
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ictcore
ICTCore: Unified Communications Framework for web developers
ICTCore is an open-source Communications Platform as a Service (CPaaS) designed to empower developers and system integrators to build, deploy, and manage communication-enabled applications with ease. With support for voice, SMS, email, and fax channels, ICTCore provides a programmable communication layer that enables rapid development of ICT-based solutions using standard development skills.Following are few projects developed over ICTCore communications framework ICTFax open source fax... -
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ictdialer
SMS, Fax, Voice Broadcasting and auto dialer Software
ICTDialer is an open-source, multi-user auto dialer software designed for voice broadcasting and fax broadcasting. It supports both inbound and outbound communications using advanced telephony protocols such as T.38, G.711 pass-through, and SIP-based VoIP communication. ICTDialer is built on top of renowned open-source technologies, including FreeSWITCH, ICTCore communications framework, and a PHP-based Angular framework. ICTDialer can be used in following faxing scenarios Voice Broadcasting Email to fax / web to fax / fax to email ATA support supporting both sending and recieving Fax over Fax machines using ATA. ... -
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A C++ framework utilizing Design Patterns for creating Linux and Windows communications applications that contain Dialogic® products. Includes media and network classes (analog, digital, SIP, H323), multithreaded event handling, distributed app support.
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Elastix
Unified Communications Server
Elastix is a software-based PBX powered by 3CX and based on Debian. An open-standards solution, Elastix is an easy to install and manage UC system compatible with popular IP phones, gateways and SIP trunks. Elastix is complete with unified communications features such as integrated WebRTC video conferencing, chat, presence and softphones and smartphone clients for Windows, Mac, iOS and Android. -
Free and Open Source HR SoftwareGive your HR team the tools they need to streamline administrative tasks, support employees, and make informed decisions with the OrangeHRM free and open source HR software.
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VitalPBX
Unified Communications System - PBX System
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... -
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Cisco TMS PhoneBook - PHP Version
Emulates Cisco TMS PhoneBook for CMM or Endpoints with PHP
...Anyway, this tool emulates the TMS PhoneBook and allows you to search and add new participants to a existing conference. The search is regular expression and the filter compare the user query with the endpoint name or H323/SIP dial string from the database. All you need is a Web server (tested with apache) with PHP and "RewriteEngine on" to make it work Optionally you can use a SQL server to have a real database, but I used a simple array as db. -
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Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (softphones & hardphones) to work behind an IP masquerading firewall or router.
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Jitsi Desktop
Jitsi is an audio/video and chat communicator that supports protocols
Jitsi Desktop is a free open-source audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, IRC and many other useful features. Jitsi Desktop is the heritage of Jitsi Meet. While some components are still used in e.g. Jigasi, the project is not actively developed anymore. Improvements, bugfixes and builds are entirely based on community contributions. -
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jphonelite
Java VoIP Softphone (SIP) (replaced by jfPhone)
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. -
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jfBroadcast
VoIP/SIP Autodialer
VoIP/SIP AutoDialer. Broadcasts a message with the option to transfer person to another number. Now includes new survey options. -
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pcapsipdump is libpcap-based SIP sniffer with per-call sorting capabilities. It writes SIP/RTP sessions to disk in a same format, as "tcpdump -w", but one file per SIP session (even if there is thousands of concurrent SIP sessions). Getting started: http://pcapsipdump.sf.net/
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Protocol plugin for Office 365/Lync/OCS
for Adium, Pidgin, Miranda and Telepathy IM Framework
Please make sure to read this page: http://sourceforge.net/p/sipe/wiki/Frequently%20Asked%20Questions/ A third-party Pidgin/Adium/Miranda/Telepathy plugin for the extended version of SIP/SIMPLE used by various products: * Skype for Business * Microsoft Office 365 * Microsoft Business Productivity Online Suite (BPOS) * Microsoft Lync Server * Microsoft Office Communications Server (OCS 2007/2007 R2) * Microsoft Live Communications Server (LCS 2003/2005) -
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OpenSIPS (former OpenSER) is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. IMPORTANT: this is no longer the main hosting for the project. This was moved on GITHUB - https://github.com/OpenSIPS/opensips
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Vodia PBX
A Better PBX For Your Business
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. -
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Accent Tool Suite
Accent interfaces and controls systems through goals and policies
Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances... -
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call Centers and Contact Centers for Quality monitoring (QM) purposes. -
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SIPSL (SIP Service Layer) is a Programmable Session Border Controller (SBC) and SIP B2BUA, as defined in RFC3261. Written in c++ and multithreaded. The Application logic can be implemented by extending the application class and implementing the call backs.
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Taki
SIP softphone
Cross-platform SIP softphone -
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PLEASE NOTE that all current AstLinux development is on GitHub: https://github.com/astlinux-project/astlinux Thanks to SourceForge for providing free hosting to the AstLinux Project for more than 10 years! Old source archive remains on SourceForge. AstLinux is a custom Linux distro centered around Asterisk, the Open Source PBX. AstLinux has many features that make it ideal for embedded and commercial Asterisk based solutions.
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
...https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .
