15 projects tagged "SIP"
Updated 21 Feb 2013
Ekiga
| Pop | 749.73 |
|---|---|
| Vit | 45.78 |
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 08 Jan 2012
SIPFwd
| Pop | 61.89 |
|---|---|
| Vit | 4.75 |
The SIP forwarding daemon (implemented as a stateless SIP proxy) allows you to seamlessly forward SIP requests to other SIP servers. Its main purpose is to enable users to use their own domain name in SIP URIs without the hassle of having to run a full-blown SIP server (by forwarding SIP requests to third-party SIP servers). Configuration information is stored in an SQLite database, and low resource consumption is a main priority for the project.
Updated 26 Nov 2009
Jingle Nodes
| Pop | 26.04 |
|---|---|
| Vit | 35.68 |
Jingle Nodes is a relay auto-discovery service that provides Jingle Relay Type Candidates that can be used in ICE-UDP and also on RAW-UDP Jingle sessions. Relay candidates can provide NAT traversal for users who don't have STUN/TURN Support, but also for users with STUN/TURN support for whom negotiation failed. It is designed to allow you to communicate freely with your friends without being attached to closed service providers like Skype or telecommunications carriers.
Updated 07 Feb 2010
Open Unified Recording
| Pop | 44.05 |
|---|---|
| Vit | 34.66 |
Open Unified Recording (OUR) is a full featured Linux-based VoIP/SIP call recording engine, indexing, and retrieval system. The system resides on the network and passively captures SIP sessions.
Updated 21 Sep 2011
SIP Inspector
| Pop | 71.75 |
|---|---|
| Vit | 2.35 |
SIP Inspector is a tool to simulate different SIP messages and scenarios. You can create your own SIP signaling scenarios, customize SIP messages, and monitor incoming and outgoing messages. The tool can play RTP streams from a pcap file.
Updated 21 Jun 2010
WikiPBX
| Pop | 37.31 |
|---|---|
| Vit | 1.00 |
WikiPBX is a PBX Web interface for FreeSWITCH. Multiple "accounts" are supported per server instance: each account is effectively a completely independent PBX. Configuration is layered so that XML files go on top of what is stored in the database. This allows you to use a database, but stays out of your way if you choose to use flat files. Extensions, SIP endpoints, and gateways can be configured via a Web interface. Live calls can be viewed, hanged up, and transferred. Call history (CDR records) can be viewed over the Web interface. There is a Web interface for managing IVRs. "Sound clips" can be easily recorded for use in dialplan or IVRs. Audio or text-to-speech can be injected into live calls.
Updated 27 Aug 2010
Amatix Office
| Pop | 48.66 |
|---|---|
| Vit | 1.02 |
Amatix Office is a complete communication platform for small and medium-sized businesses. It provides email, calendar, contacts, conventional telephony like ISDN, new generation VoIP telephony like SIP, instant messaging, presence, and other features needed to power a business. Amatix Office is very easy to install and use and does not require any special IT skills. It is a live software appliance which boots a computer directly from a CD or USB flash drive without any software installation.
Updated 13 Dec 2010
SEMS
| Pop | 48.44 |
|---|---|
| Vit | 1.46 |
SEMS is a media and application server for SIP based VoIP services. It shows good performance doing basic services like announcements and conference for combination with external application servers. Thanks to its easy-to-use and flexible application development framework and back-to-back user agent support, application logic and media serving can be combined in the same process. Basic applications like announcement, pre-call announcement, RBT, conference, voicemail, mailbox, and lots of example applications are available. Scripting can be done in Python and a simple state machine description language. Support All commonly used free codecs (including g711, gsm, iLBC, speex, adpcm, and l16) are supported. Other features include wideband, ZRTP encryption, a SIP registrar client, an XMLRPC server/client, and a DIAMETER client.
Updated 06 May 2013
libre
| Pop | 155.76 |
|---|---|
| Vit | 19.38 |
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 18 May 2013
baresip
| Pop | 243.02 |
|---|---|
| Vit | 43.55 |
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.
