48 projects tagged "voip"
Updated 18 May 2013
baresip
| Pop | 209.70 |
|---|---|
| Vit | 97.11 |
baresip is a bare-bones SIP user agent. It supports SIP, SDP, RTP/RTCP, and STUN/TURN/ICE, and IPv4 and IPv6, and is RFC-compliant and has portable C89 and C99 source code. A modular plugin architecture provides stdio, cons, and evdev user interfaces, celt, g711, g722, gsm, ilbc, l16, and speex audio codecs, alsa, coreaudio, gst, portaudio, oss, winwav, and mda audio drivers, speex_pp, speex_aec, speex_resamp, and sndfile audio filters, the avcodec video codec, avformat, quicktime, qtcapture, v4l, and v4l2 video sources, sdl, opengl, and x11 video display drivers, and srtp media encoding.
Updated 06 May 2013
libre
| Pop | 152.26 |
|---|---|
| Vit | 21.37 |
libre is a generic library for real-time communications with asynchronous I/O support. It is written in portable POSIX source code that conforms to the ANSI C89 and ISO C99 standards. It is robust and fast, with a low memory footprint. It also features RFC compliance and support for IPv4 and IPv6. Protocol implementations include SIP, SDP, RTP/RTCP, BFCP, DNS, and STUN/TURN/ICE.
Updated 21 Feb 2013
Ekiga
| Pop | 846.75 |
|---|---|
| Vit | 46.55 |
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.
Updated 23 Jan 2013
Discretio
| Pop | 85.94 |
|---|---|
| Vit | 2.22 |
Discretio is an application (and service) for secure VoIP on smartphones. It allows users to establish calls over the Internet while using state of the art encryption technologies to avoid eavesdropping.
Updated 14 Jan 2013
HOMER SIP Capture
| Pop | 34.26 |
|---|---|
| Vit | 11.24 |
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.
Updated 07 Jan 2013
Yet Another Telephony Engine
| Pop | 426.83 |
|---|---|
| Vit | 27.57 |
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 21 Dec 2012
fonosip-siphon
| Pop | 30.82 |
|---|---|
| Vit | 2.06 |
The fonosip-siphon ultility installs preferences, settings, and icons for using FonoSIP with the Siphon softphone.
Updated 14 Nov 2012
OZEKI VoIP Service Checker
| Pop | 18.44 |
|---|---|
| Vit | 13.66 |
The Ozeki VoIP Service Checker is a tool you can use for checking the availability of VoIP services. It can be used to verify whether a given service exists, see whether it can be reached from a particular network, and check whether the user details (such as the username and password) to access the service are correct. It was developed because many mobile Internet providers block VoIP traffic.
Updated 22 Jul 2012
Text translation for Asterisk using MS Tra...
| Pop | 36.00 |
|---|---|
| Vit | 1.57 |
Text translation for Asterisk using MS Translator uses the Microsoft Translator API to translate text strings or detect their language and return them as Asterisk channel variables.
Updated 22 Jul 2012
Speech synthesis for Asterisk using MS Tra...
| Pop | 40.66 |
|---|---|
| Vit | 2.06 |
Speech synthesis for Asterisk using MS Translator text-to-speech service to synthesize speech and play it back to the user. It supports a variety of languages, local caching of voice data, and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.
