Search Results for "sofia-sip" - Page 3
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Showing 323 open source projects for "sofia-sip"
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Taki
SIP softphone
Cross-platform SIP softphone -
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CRUSH
Data reduction and imaging for select astronomical cameras
CRUSH is a pioneering reduction and imaging suite for many astronomical cameras operating at far-infrared, submillimeter and millimeter wavelengths, such as SHARC-2, SCUBA-2, LABOCA, GISMO, or SOFIA/HAWC+ (to name a few). -
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PLEASE NOTE that all current AstLinux development is on GitHub: https://github.com/astlinux-project/astlinux Thanks to SourceForge for providing free hosting to the AstLinux Project for more than 10 years! Old source archive remains on SourceForge. AstLinux is a custom Linux distro centered around Asterisk, the Open Source PBX. AstLinux has many features that make it ideal for embedded and commercial Asterisk based solutions.
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Scilab Image Processing Toolbox
Advanced image processing toolbox for Scilab on Unix/Linux/Mac OS
SIP is the image processing and computer vision package for SciLab, a free Matlab-like programming environment. SIP reads/writes images in formats like JPEG, PNG, and BMP. It does filtering, segmentation, edge detection, morphology, and shape analysis. Download from Git http://siptoolbox.sourceforge.net/devel -
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SIP_SERVER
SIP_SERVER_METI
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Zen Load Balancer becomes Zevenet
becomes ZEVENET
High performance and easy to use open source load balancer with enterprise solutions. -
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MCU Media Server
SIP Video Multiconference Media Server with WebRTC support.
REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. . -
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Notepad++ SIP Language
Notepad++ Language for highlighting SIP traces
SIP traces are great. They provide very detailed information, but can be very dense. This is not only true for newbies. A block of text is tough on the eyes, particularly when parsing through multiple calls, even for the most experienced technicians. Having a mechanism to color code the traces can make it easier both to look through them while troubleshooting, and to convey findings to someone else. -
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Callflow Sequence Diagram Generator
Callflow Sequence Diagram Generator
The callflow sequence diagram generator is a collection of awk and shell scripts that will take a packet capture file that can be read by wireshark and produce a time sequence diagram. This is useful to view & debug SIP callflows or other network traffic -
Red Hat Enterprise Linux on Microsoft AzureRed Hat Enterprise Linux (RHEL) on Microsoft Azure provides a secure, reliable, and flexible foundation for your cloud infrastructure. Red Hat Enterprise Linux on Microsoft Azure is ideal for enterprises seeking to enhance their cloud environment with seamless integration, consistent performance, and comprehensive support.
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Animal is AN IMAging Library written in C. Its simple API supports over 80 image formats, and is intended to make massive use of other image processing libraries. Animal aims at image analysis and recognition. It is mainly the C basis of the SIP toolbox.
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nagios_check_asterisk_ami
For Asterisk 1.8 through 13. Includes PJSIP, DAHDI, SIP, and SCCP
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wxCommunicator is a cross platform SIP softphone written in C++ utilizing customized sipXtapi user agent library and wxWidgets 2.8.9 GUI library. For a list of supported features see http://wxcommunicator.sourceforge.net/features.html .
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SIP Data Filter (SiDaFir)
Simple and efficient tool for SIP trace filtering
The Session Initiation Protocol (SIP) is a signaling communications protocol widely used nowadays for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks. Thanks to its simplicity, SIP messages are often used in creative ways for which these were not originally designed (e.g. using periodical OPTIONS packets as NAT keep-alive instead of using STUN or TURN) and thus SIP traces of the captured traffic often contain "useless" traffic... -
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jpbxlite
Java VoIP/SIP PBX system (replaced by jfPBX)
jPBXLite is a VoIP/SIP PBX. Supports SIP extensions, voicemail, trunks, conferences, queues (ACD) and an IVR system. Support video conferencing with jPhoneLite/1.4.0. NOTE:THIS PROJECT WAS RENAMED AND IS NOW jfPBX. Please go to jfpbx.sourceforge.net -
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astipupdate script, detects public ip and sip peer ip change , blocks unwanted sip traffic. script start parameter will permit sip traffic only for trusted hosts via iptables, can also update them with updatepeer parameter if ip changes. updatepublic parameter reloads asterisk on public ip change (in case you have too long register intervals). copy this script to /etc/init.d and activate it with update-rc.d astipupdate defaults enable in cron like: */5 * * * * root /etc/init.d/astipupdate...
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OpenMotif-Mac
Making old wine run on new wineskins, or something.
A modified distribution of IST OpenMotif 2.1.32 to run from /Applications and interface correctly with SIP, compatible with 10.4 through 10.11, PowerPC and i386. Includes all binaries, libraries and development headers. To install, open the disk image and drag the folder to /Applications. Do not change any names or relative paths. For 10.4, use the 10.4 download. For 10.5 through 10.11, use the 10.5 download. Binaries are unsigned. You use this tool at your own risk, under the Open... -
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SipCo Lite
SipCo Lite is a SIP based VoIP sofphone which supports VPN tunnel.
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Mobicents is the leading Open Source VoIP Platform. It is the First and Only Open Source Certified implementation of JSLEE 1.1 (JSR 240), and SIP Servlets 1.1 (JSR 289). Mobicents also includes a powerful and extensible Media Server.
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Asterweb is an Asterisk Realtime Configuration utility written in PHP. It configures the realtime settings for voicemail, extensions and sip buddies. Users can login with their voicemail user and pin and check their voicemail.
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A TAPI driver for SIP. SIPTAPI gives you a click2dial feature with any TAPI enabled application (e.g. MS Outlook) and SIP PBX/proxy. This is the "free" version of SIPTAPI. There is also an enhanced commercial version available at www.ipcom.at.
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Steve Murray Phone
simple SIP video phone source/lib/dll/API with h264, ulaw and g729
I coded up this simple SIP engine and video phone a few years ago. I realized how easy it is to speak SIP without a lot of classes or abstraction. It is just plain-english SIP. The SIP is copy-pasted from wireshark, and then just does replace of whatever characters need to be changed to make or answer a specific call. I don't program much VOIP any more, and pretty soon I might have thrown away all my hard drives with this code on it. So I figured I would put my work somewhere in case... -
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Git repo: https://github.com/asipto/siremis Web management interface for Kamailio (OpenSER) - handle subscriber profiles, access control lists, accounting, least cost routing and load balancing, monitoring charts, xmlrpc communication with SIP server
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Sippy B2BUA and RFC3261 SIP Stack
This project has been relocated to http://github.com/sippy/rtpproxy h
This project has been relocated to http://github.com/sippy/rtpproxy http://github.com/sippy/b2bua -
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Peers is a very simple softphone. It's a SIP User-Agent, written in java, it works on windows, linux and mac. It can be used with SIP servers like opensips or asterisk IPBX. It supports G711 codec (PCMU and PCMA) and telephone-events (DTMF).