RSS 81 projects tagged "Telephony"

Download Website Updated 24 Mar 2014 OpenSIPS

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Pop 243.00
Vit 34.45

OpenSIPS is a mature implementation of a SIP server/proxy. It is more than a SIP proxy/router, as it includes application-level functionalities. OpenSIPS, as a SIP server, can server as the core component of any SIP-based VoIP solution.

Download Website Updated 06 Mar 2014 Kamailio

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Pop 411.05
Vit 55.18

Kamailio (formerly OpenSER) is a high-performance SIP (RFC3261) server with a flexible architecture and many extensions. The server implements proxy, registrar, redirect, and location SIP/VoIP services. It has support for UDP, TCP, TLS, and SCTP transport layers, DNSsec, ENUM, AAA via database, RADIUS, DIAMETER, gateways to SMS and XMPP, least cost routing, load balancing, NAT traversal, and call processing language. Kamailio implements SIMPLE presence and instant messaging extensions, and includes an embedded XCAP server and MSRP relay. It can be also used as a routing SIP sever for WebRTC via WebSocket.

Download No website Updated 14 Jan 2014 SFLphone

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Pop 543.23
Vit 31.73

SFLphone is an SIP/IAX2 compatible softphone. The goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed for intensive corporate use.

Download Website Updated 28 Apr 2013 listen2fritz

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Pop 25.14
Vit 4.48

listen2fritz provides real-time notification of incoming calls on an AVM FRITZ!Box Fon. The notifications are sent by way of a Dream-Multimedia-TV Dreambox or an instant messenger program.

Download Website Updated 07 Apr 2013 amrwb

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Pop 28.53
Vit 1.00

amrwb is a shared library wrapper for the 3GPP AMR Adaptive Multi-Rate - Wideband (AMR-WB) Speech Codec (3GPP TS 26.204).

Download Website Updated 07 Apr 2013 amrnb

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Pop 30.63
Vit 1.00

amrnb is a shared library wrapper for the 3GPP Adaptive Multi-Rate Floating-point (AMR) Speech Codec (3GPP TS 26.104).

Download Website Updated 21 Feb 2013 Ekiga

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Pop 422.80
Vit 22.03

Ekiga (formely known as GnomeMeeting) is a soft phone, video conferencing, and instant messenger application for use over the Internet. It supports HD sound quality and video up to DVD size and quality. It is interoperable with many other standards compliant software, hardware, and service providers as it uses both of the major telephony standards, SIP and H.323.

Download Website Updated 14 Jan 2013 HOMER SIP Capture

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Pop 23.75
Vit 21.41

HOMER is a robust, carrier-grade, scalable SIP capturing system and monitoring application with hEP, IP Proto4 (IPIP) encapsulation, and port mirroring/monitoring support right out of the box, ready to process and store large amounts of signaling with instant searches, end-to-end analysis, and drill-down capabilities for ITSPs, VoIP providers, and trunk suppliers using SIP signaling.

No download Website Updated 14 Apr 2012 chan_nms

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Pop 33.09
Vit 1.00

CHAN_NMS is an Asterisk PBX channel driver that supports the NMS Communications (acquired by Dialogic) Open Access line of products, which includes E1/T1 boards for the PCI(-X), PCIe, and cPCI(e) buses.

Download Website Updated 26 Dec 2011 mediastreamer

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Pop 127.16
Vit 4.75

Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.

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VTE

Determine if data received from a pseudo-terminal is a control sequence.

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GNUnet

A framework for secure peer-to-peer networking.