Open Source C VoIP Software

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. VoIPmonitor can also decode audio.

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.

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 browsing. Tor/i2p Profile: Browse over Tor/i2p on Firefox (Vanilla, ESR), Waterfox (Classic, New), Palemoon , LibreWolf and legacy CyberFox. As with all versions of Tor - do not rely on this for strong anonymity. A usability enhanced Privacy Pack. Use alongside Tor Browser Bundle. An installer, for : Vista 32/64, Win7 32/64, Win8 32/64, Win10 32/64, Win11, Linux (Wine)

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

Replacement for the SCCP channel driver in Asterisk. Extended features include Shared Lines, Presence / BLF, customizable Feature Buttons, and Custom Device State. Visit our discussion mailing list for help and join us as a developer if you like. The project moved to https://github.com/chan-sccp/chan-sccp

Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

The Open Source IMS Core Project

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/

This project aims to build a software reference implementation of the EBU standard for the transmission of high quality, low latency, audio streams over IP networks. (EBU-tech 3326)

Application to remote control Polycom VVX IP phones via their web interface (using HTTPS). This application is initially intended to perform certain actions on phones running in Microsoft Lync mode. These actions include: * get device information * get status * sign in using PIN authentication * sign out * reboot * factory reset It also supports performing actions in batch reading data from a CSV file.

Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip

TurnServer is a implementation of Traversal Using Relay around NAT (TURN) protocol. This protocol allows a client to obtain IP addresses and ports from such a relay.

KAMAILIO (OpenSER) - robust, secure and scalable Open Source (GPL) SIP (RFC3261) server implementation with large features set (over 90 extension modules). As of May 2009, source code is hosted by GIT repository at http://sip-router.org

oSKY (OpenSKY) is a full communication framework to implement PSoTN (Public Social Telephony Network). In contrast with PSTN (Public Switched Telephony Network), oSKY intends to implement an open paradigm in the internet domain, gaining more than 100 years of experience from traditional telephone network. oSKY goals: -enhanced security (SSL, SCTP) -embedded NAT tunneling -Global title discovering on Internet style -Peer2Peer mode and Gateway mode -more to come ... oSKY project drivers: - drastically reduce bandwidth usage by performing smart lookup of enum numbering space. avoiding PSTN outbound trunking when not necessary and linking IPPABXs in the internet domain with failover capabilities.

The project is a preconfigured VoIP PBX VM Image based on Asterisk. But provides more advance features, such as LDAP integration, VQ monitoring via RTCP XR reports, LDAP user login, integrated billing system and telephone directory PDF generator.

A $10 Analog Telephone Adapter (ATA) that allows low cost routers such as the WRT54G to connect to an analog telephone. The application is low cost VOIP, in particular for the developing world.

8ix Zenith CE spells Asterisk derived IP Telephony solution.When you want robust communication, without the huge cost and recurring bill that comes with the way you do business, 8ix Zenith CE addresses your every communication need.

A set of scripts to allow sending and receiving of fax documents on HylaFAX using res_fax and res_fax_digium of Asterisk

Video Applications for Asterisk (libh43m, app_h324m, app_mp4, app_rtsp, app_transcode, app_swf) H324m stack, mp4 record and playblack, rtsp support, video transcoding and swf file playback.

This project is aim to develop module named app_avi.so for asterisk to add the ability of play avi file.

این پروژه تغییر نام یافته است و با نام آواسینتا ادامه می یابد. برای اطلاعات بیشتر به صفحه اصلی آواسینتا در شبکه پلاس مراجعه نمایید. https://shabakehplus.com/projects/avasinta/

آواسینتا ویرایشی اختصاصی شده از نرم افزار ویپ آستریسک می باشد که از زبان فارسی نیز پشتیبانی می نماید. این برنامه رایگان بوده و روی سیستم عامل اوبونتو ارائه می شود. برای دریافت اطلاعات بیشتر د ر مورد سیستم تلفن آواسینتا می توانید به صفحه اصلی آواسینتا در شبکه پلاس مراجعه نمایید https://samiyacode.ir/avasinta/

Castadiva is a test bed based on low-cost off-the-shelf devices, which is used to test protocols developed for MANETs. Castadiva is completely compatible with the file format used by the ns-2 simulator.

Libre Software to manage audio/video connection from a PC to a libre hardware real time stethoscope hardware, developed to allow doctors make a remote diagnose in real time in isolated rural areas of developing countries or for medicine teaching

Farfon, was the world's first designed from the ground up IAX2 phone created to work with Asterisk PBX.