Search Results for "sofia-sip"

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. ...

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...

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/

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

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

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

LuaSofia is a Lua binding of Sofia-Sip library. Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. We decided to use git, source code can be get at: git://github.com/ppizarro/luasofia.git

sipsak is a command line tool which can send simple requests to a SIP server. It can run additional tests on a SIP server which are usefull for admins and developers of SIP enviroments. Note: the project is maintained on GitHub https://github.com/nils-ohlmeier/sipsak

Very fast and very flexible SIP server

The Milkfish is an Embedded Communications Software Project. Main objective is to provide a basic NAT traversal solution for cheap hardware enabling a practical setup of multiple SIP phones in a LAN sharing a WAN connection with one public IP address.

OSP Module enables SER (SIP Express Router) to use the Open Settlement Protocol for secure Authorization, Routing and Accounting. OSP is an ETSI protocol, Technical Specification 101 321

This projects intend to port to Windows / Interix the famous and performant SIP Express Router from IPTEL (SER)

Asterisk based SIP b2bua realization. It's patched asterisk and some AGI script. Features: full vovida's b2bua radius emulation, extended radius attributes, radius failover, LCR, Call failover, Codec based routing and other useful things.

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.

dtmfbox is a small softswitch application (SIP/CAPI), that can be used to control different tasks over telephone keyboard via DTMF. Mostly, it was made to run on the AVM FRITZ!Box 7170 (mipsel) but works under Unix/Linux and Windows, too.

This project is for people who want to access one or more HFC-S PCI A based ISDN BRI cards over the DAHDI interface and maybe want to use other DAHDI hardware in parallel, too. DAHDI driven hardware can be used by Asterisk, FreeSWITCH, Yate and other telephony software. The HFC-S PCI A based cards are very cheap and thus ideal for private BRI users. The OSLEC is being developed since a long time ago and said to be very good. The superiority of "hardware echo cancellers" is a myth. This...

SIP over IPv4 and SIP over IPv6 are different worlds, making it impossible to register IPv4-only phones on IPv6-only services, and so on. SIPproxy64 is a proxy that translates between the two worlds. It is small enough to run inside a router.

This is a open source media gateway controller. The protocol used is MEGACO/H.248. In a later version also SIP will be supported.

Live-cd com Servidor SIP baseado em opensips. O Servidor de Telefone Livre transformará um simples PC numa central SIP multidominio. Estamos construindo a documentacao do projeto na wiki do sistema trac.

This project provides a meter of QoS on VoIP using the protocol SIP Media Loopback

Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).

A SIP-based Distributed Hash Table

This is a simple software, comprised of a server and a client, that can be used together with any IP-phone using the SIP protocol (TELE2 and Bredbandsbolget in sweden). It also retrives information, sush as name, adress, etc about the number from eniro.s

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