59 projects tagged "Telephony"
Updated 14 May 2013
GNU Gatekeeper
| Pop | 557.34 |
|---|---|
| Vit | 206.45 |
The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.
Updated 14 Apr 2013
Ip phone Scanning Made Easy (ISME)
| Pop | 395.14 |
|---|---|
| Vit | 12.59 |
Ip phone Scanning Made Easy (ISME) scans a VOIP environment, adapts to enterprise VOIP, and exploits the possibilities of being connected directly to an IP Phone VLAN. It seeks to get the phone's configuration file directly from a TFTP server, enable SIP/SIPS (TCP/UDP), communicate with an embedded Web server and Web server banner, identify the editor by MAC address, and identify potential default login/password combinations which should be changed.
Updated 07 Apr 2013
amrwb
| Pop | 60.96 |
|---|---|
| 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).
Updated 07 Apr 2013
amrnb
| Pop | 59.57 |
|---|---|
| Vit | 1.00 |
amrnb is a shared library wrapper for the 3GPP Adaptive Multi-Rate Floating-point (AMR) Speech Codec (3GPP TS 26.104).
Updated 06 Feb 2013
Skype
| Pop | 533.22 |
|---|---|
| Vit | 20.98 |
Skype uses P2P (peer-to-peer) technology to provide voice- and video-based communication with other Internet users. The technology is extremely advanced, but easy to use. It features excellent sound quality, end-to-end encryption, and automatic negotiation of firewalls or routers. Among major features are SkypeOut and SkypeIn, adding the possibility to make low-cost calls to land line phones and having a fixed number to be able to receive calls from land line phones.
Updated 18 Jan 2013
MyConnection Server
| Pop | 198.14 |
|---|---|
| Vit | 21.93 |
MyConnection Server is broadband testing software which measures connections for bandwidth speeds and connection quality for time critical applications such as VoIP, Video conferencing, and IPTV. It helps organizations assess networks for deployment of new/additional services and identify and resolve last mile customer connectivity problems with little need for the customer to assist in the resolution process. A network route testing component details the path of the connections and where packet loss and latency occur, including discovery of multiple routes to a destination. Remote Test Agents enable technical staff to customize and interactively manage the bandwidth testing process and perform extended quality testing over hours or days to address and resolve intermittent problems as required. Satellite Servers establish additional connection testing points at the application edge to accurately test actual application network paths.
Updated 07 Jan 2013
Yet Another Telephony Engine
| Pop | 425.65 |
|---|---|
| Vit | 27.47 |
Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.
Updated 22 Sep 2012
OTRS
| Pop | 652.91 |
|---|---|
| Vit | 13.74 |
OTRS is a platform independent Web-based help desk system that supports service organization of any kind (e.g. IT service, customer and technical product service, complaint management, public services, etc.) to increase their efficiency. It increases transparency as well as service quality and lowers your total cost of ownership. It has been certified ITIL V3 compatible by PinkVERIFY for incident, problem, change, service asset and configuration, request fulfillment, and knowledge management. Other ITIL processes like service catalog and service level management are supported as well.
Updated 26 Dec 2011
mediastreamer
| Pop | 157.75 |
|---|---|
| Vit | 5.72 |
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.
Updated 26 Dec 2011
linphone
| Pop | 494.35 |
|---|---|
| Vit | 21.79 |
Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, G722, ilbc, amr, Theora, H263-1998, MPEG4, H264, VP8, and snow.
