voip sdk
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PlayVoIP 0.1
PlayVoIP is a webbased VoIP service enabler featured with user management. more>>
PlayVoIP is a webbased VoIP service enabler featured with user management, statistical report and registration application system based on LAMP and Asterisk.
At the beginning this project was specially designed for simple and free Indonesian VoIP network, VoIP Rakyat.
<<lessAt the beginning this project was specially designed for simple and free Indonesian VoIP network, VoIP Rakyat.
Download (0.027MB)
Added: 2005-12-02 License: GPL (GNU General Public License) Price:
1422 downloads
JCom SDK 1.2
JCom SDK provides a network package for the Java language. more>>
JCom SDK provides a network package for the Java language.
The JCom SDK is a network package for the Java language. It allows the programmer to create client/server application with little knowledge about networks and no knowledge about TCP/IP. It is basically a set of wrappers built around Javas internal TCP/IP socket classes.
The JCom SDK is a network package for the Java language. It allows the programmer to create client/server application with little knowledge about networks and no knowledge about TCP/IP. The JCom SDK is basically a set of wrappers built around Javas internal TCP/IP socket classes. To create a client connection, the programmer needs only to create the client class with the appropriate IP address. As for the server, the programmer needs only to specify on which port the server will receive the connection and when to start listening. All aspect of the connection is handle internaly. The programmer needs only to consult its event queue to receive messages and add messages to remote queues to send messages.
The initial version of the JCom SDK was design as a network module for a school project. However, I saw that with a little modification, I could make the module generic for any application. The idea came to me when a few friends asked me if they could borrow my network modules for their project. After a little coding and a lot of comments, the first public version of the SDK was released. Thought I am not available to answer questions or offer support, suggestions and comments are always welcomed.
Enhancements:
- JCom is now thread safe (big thanks to Marc)
<<lessThe JCom SDK is a network package for the Java language. It allows the programmer to create client/server application with little knowledge about networks and no knowledge about TCP/IP. It is basically a set of wrappers built around Javas internal TCP/IP socket classes.
The JCom SDK is a network package for the Java language. It allows the programmer to create client/server application with little knowledge about networks and no knowledge about TCP/IP. The JCom SDK is basically a set of wrappers built around Javas internal TCP/IP socket classes. To create a client connection, the programmer needs only to create the client class with the appropriate IP address. As for the server, the programmer needs only to specify on which port the server will receive the connection and when to start listening. All aspect of the connection is handle internaly. The programmer needs only to consult its event queue to receive messages and add messages to remote queues to send messages.
The initial version of the JCom SDK was design as a network module for a school project. However, I saw that with a little modification, I could make the module generic for any application. The idea came to me when a few friends asked me if they could borrow my network modules for their project. After a little coding and a lot of comments, the first public version of the SDK was released. Thought I am not available to answer questions or offer support, suggestions and comments are always welcomed.
Enhancements:
- JCom is now thread safe (big thanks to Marc)
Download (0.040MB)
Added: 2007-04-05 License: LGPL (GNU Lesser General Public License) Price:
932 downloads
DPML SDK 2.0.2
DPML SDK is a management platform for Java-based component development, resource management, and deployment. more>>
DPML SDK is a management platform for Java-based component development, resource management, and deployment.
DPML SDK is composed of the Transit resource management layer, the Depot multi-project build system, the Metro component management runtime engine, and the Station application management console.
Enhancements:
- New features and enhancements added since version 2.0.1 include the addition of support for component composition via reference to nested parts within the internal structure of a component using part URIs as the construction mechanisms.
- This version also includes numerous code and documentation quality enhancements and updates to the checkstyle task involving migration from Checkstyle 4.1 to 4.3.
<<lessDPML SDK is composed of the Transit resource management layer, the Depot multi-project build system, the Metro component management runtime engine, and the Station application management console.
Enhancements:
- New features and enhancements added since version 2.0.1 include the addition of support for component composition via reference to nested parts within the internal structure of a component using part URIs as the construction mechanisms.
- This version also includes numerous code and documentation quality enhancements and updates to the checkstyle task involving migration from Checkstyle 4.1 to 4.3.
Download (5.7MB)
Added: 2007-04-26 License: The Apache License 2.0 Price:
911 downloads
Other version of DPML SDK
License:The Apache License 2.0
JVOIPLIB 1.4.0
JVOIPLIB is an object-oriented Voice over IP (VoIP) library written in C++. more>>
JVOIPLIB is an object-oriented Voice over IP (VoIP) library written in C++.
It is based upon work done for my thesis at the School for Knowledge Technology (or School voor Kennistechnologie in Dutch), a cooperation between the Hasselt University and the Maastricht University.
A part of this library was developed at the Expertise Centre for Digital Media (EDM) in Diepenbeek, Belgium. The EDM is a research institute of the Hasselt University.
Main features:
- Easy VoIP session creation and destruction.
- Highly configurable sessions: sampling rate, sample interval, compression type, ... can all be selected by the user. These features can also be changed during a session.
- Openness and extensibility: the object-oriented nature of the library makes it very easy to add features; new components can easily be tested by registering them as User Defined modules.
- Support for 3D effects: for my thesis I also did some research and development about VoIP in networked virtual environments, which included adding 3D effects to sound. For this reason, Ive added this feature to the library.
<<lessIt is based upon work done for my thesis at the School for Knowledge Technology (or School voor Kennistechnologie in Dutch), a cooperation between the Hasselt University and the Maastricht University.
A part of this library was developed at the Expertise Centre for Digital Media (EDM) in Diepenbeek, Belgium. The EDM is a research institute of the Hasselt University.
Main features:
- Easy VoIP session creation and destruction.
- Highly configurable sessions: sampling rate, sample interval, compression type, ... can all be selected by the user. These features can also be changed during a session.
- Openness and extensibility: the object-oriented nature of the library makes it very easy to add features; new components can easily be tested by registering them as User Defined modules.
- Support for 3D effects: for my thesis I also did some research and development about VoIP in networked virtual environments, which included adding 3D effects to sound. For this reason, Ive added this feature to the library.
Download (0.56MB)
Added: 2005-10-18 License: LGPL (GNU Lesser General Public License) Price:
1466 downloads
Minisip VoIP Provider
The routing in incoming calls(answering machine, transmission, for example) is configurable in the webinterface. more>> <<less
Download (4.19MB)
Added: 2009-04-26 License: Freeware Price: $0
198 downloads
Tapioca VoIP 0.3.9
Tapioca is a framework for Voice over IP (VoIP) and Instant Messaging (IM). more>>
Tapioca is a framework for Voice over IP (VoIP) and Instant Messaging (IM). Its main goal is to provide an easy way for developing and using VoIP and IM services in any kind of application.
Tapioca VoIP project was designed to be cross-platform, lightweight, thread-safe, having mobile devices and applications in mind.
Main features:
- Create a solution that integrates all components used by VoIP and IM applications in a single, reliable and easy to use framework, which is able to work on different platforms.
- Spare resources, providing central services for multiple applications. Eg.: The control of all incoming and outgoing SIP requests are managed by the SIP service, avoiding the creation of one SIP stack and allocation of a network port for each SIP-based application.
- Reduce the overhead of control layers and library dependencies.
<<lessTapioca VoIP project was designed to be cross-platform, lightweight, thread-safe, having mobile devices and applications in mind.
Main features:
- Create a solution that integrates all components used by VoIP and IM applications in a single, reliable and easy to use framework, which is able to work on different platforms.
- Spare resources, providing central services for multiple applications. Eg.: The control of all incoming and outgoing SIP requests are managed by the SIP service, avoiding the creation of one SIP stack and allocation of a network port for each SIP-based application.
- Reduce the overhead of control layers and library dependencies.
Download (0.34MB)
Added: 2006-06-09 License: LGPL (GNU Lesser General Public License) Price:
1235 downloads
VoiceOne 0.5.0
VoiceOne project is an easy to use web based GUI for the Asterisk PBX. more>>
VoiceOne project is an easy to use web based GUI for the Asterisk PBX. It gives full control over PBX settings from the admin configuration panel.
You can manage extensions, VoIP trunks, users queues and rules sets, and dynamically create a IVR.
Main features:
- Client/Server architecture based on Web services
- Relies on Asterisk Realtime Architecture (ARA)
- SIP extensions management (support for Zap/IAX/mISDN soon added)
- Remote offices via IAX with RSA public key encryption
- Supports VoIP and traditional Telco providers
- Powerful IVR creation system
- Queues system management
- Customizable users profile
- Powerful configuration of mISDN ports/interfaces (thanks to guys at beroNet for their support)
- Policies system for users/groups management
- SIP, IAX and mISDN general/default options configuration
- Static-like text editor for conf files
- Easy setup wizard
- lots more...
Enhancements:
- The IVR implementation is now completely dynamic and database based.
<<lessYou can manage extensions, VoIP trunks, users queues and rules sets, and dynamically create a IVR.
Main features:
- Client/Server architecture based on Web services
- Relies on Asterisk Realtime Architecture (ARA)
- SIP extensions management (support for Zap/IAX/mISDN soon added)
- Remote offices via IAX with RSA public key encryption
- Supports VoIP and traditional Telco providers
- Powerful IVR creation system
- Queues system management
- Customizable users profile
- Powerful configuration of mISDN ports/interfaces (thanks to guys at beroNet for their support)
- Policies system for users/groups management
- SIP, IAX and mISDN general/default options configuration
- Static-like text editor for conf files
- Easy setup wizard
- lots more...
Enhancements:
- The IVR implementation is now completely dynamic and database based.
Download (2.2MB)
Added: 2007-07-09 License: LGPL (GNU Lesser General Public License) Price:
843 downloads
OLPC Live CD SDK Build 385
OLPC Live CD SDK is a Linux distribution created by Red Hat and based on Fedora Core for the One Laptop Per Child. more>> <<less
Download (314MB)
Added: 2007-04-17 License: GPL (GNU General Public License) Price:
925 downloads
Quake 4 1.3 SDK
Quake 4 SDK contains the full public game source code, and map, mod and art samples. more>>
Quake 4 SDK contains the full public game source code, and map, mod and art samples.
About Quake 4:
Earth is under siege by the Strogg, a barbaric alien race moving through the universe consuming, recycling and annihilating any civilization in their path. In a desperate attempt to survive, an armada of Earths finest warriors is sent to take the battle to the Strogg home planet.
You are Matthew Kane, an elite member of Rhino Squad and Earths valiant invasion force. Fight alone, with your squad, or in hover tanks and mechanized walkers as you engage in a heroic mission to the heart of the Strogg war machine.
Battle through the beginning of the game as a combat marine, then after your capture, as a marine-turned-Strogg with enhanced abilities and the power to turn the tide of the war.
Built on id Softwares revolutionary DOOM 3 technology, QUAKE 4 also features fast-paced multiplayer competition modeled after the speed, feel, and style of QUAKE III Arena.
<<lessAbout Quake 4:
Earth is under siege by the Strogg, a barbaric alien race moving through the universe consuming, recycling and annihilating any civilization in their path. In a desperate attempt to survive, an armada of Earths finest warriors is sent to take the battle to the Strogg home planet.
You are Matthew Kane, an elite member of Rhino Squad and Earths valiant invasion force. Fight alone, with your squad, or in hover tanks and mechanized walkers as you engage in a heroic mission to the heart of the Strogg war machine.
Battle through the beginning of the game as a combat marine, then after your capture, as a marine-turned-Strogg with enhanced abilities and the power to turn the tide of the war.
Built on id Softwares revolutionary DOOM 3 technology, QUAKE 4 also features fast-paced multiplayer competition modeled after the speed, feel, and style of QUAKE III Arena.
Download (61.5MB)
Added: 2006-08-01 License: Freeware Price:
1191 downloads
VoiceBuntu 1.02
VoiceBuntu (Ubunterisk) is an Ubuntu based live Linux distribution that uses Asterisk and VoiceOne to provide VoIP service. more>>
VoiceBuntu (Ubunterisk) is an Ubuntu based live Linux distribution that uses Asterisk and VoiceOne to provide VoIP service.
VoiceBuntus focus is to run asterisk with no installation needed. VoiceBuntu - ubuntu itself - has built in boot prompt cheap code one can use called persistent.
This feature allows the user to use VoiceBuntus persistent feature in order to keep settings on a flash disk or memory stick.
To start, just burn the ISO file on a CD-ROM then start the live cd distribution and wait until everything is ready.
Please note, that apache will start at the very last moment of the boot cycle. So you may a to wait 1 or 2 minutes to let everything load.
Once everything is prepared, start your browser and point to the IP address of your running VoiceBuntu (i.e. http://192.168.0.10).
If you have no DHCP Server available you can press F6 at boot screen and add additionally add the command parameter
static=xxx.xxx.xxx.xxx
netmask=xxx.xxx.xxx.xxx
bcast=xxx.xxx.xxx.xxx
gateway=xxx.xxx.xxx.xxx
dns1=xxx.xxx.xxx.xxx
dns2=xxx.xxx.xxx.xxx
where xxx.xxx.xxx.xxx is the IP of your choice.
By that VoiceBuntu boots with static IP address on eth0 network device.
Enhancements:
- This live CD release is based on Ubuntu 7.04 and VoiceOne 0.5.0_2 due to a minor bug in VoiceOne version 0.5.0.
<<lessVoiceBuntus focus is to run asterisk with no installation needed. VoiceBuntu - ubuntu itself - has built in boot prompt cheap code one can use called persistent.
This feature allows the user to use VoiceBuntus persistent feature in order to keep settings on a flash disk or memory stick.
To start, just burn the ISO file on a CD-ROM then start the live cd distribution and wait until everything is ready.
Please note, that apache will start at the very last moment of the boot cycle. So you may a to wait 1 or 2 minutes to let everything load.
Once everything is prepared, start your browser and point to the IP address of your running VoiceBuntu (i.e. http://192.168.0.10).
If you have no DHCP Server available you can press F6 at boot screen and add additionally add the command parameter
static=xxx.xxx.xxx.xxx
netmask=xxx.xxx.xxx.xxx
bcast=xxx.xxx.xxx.xxx
gateway=xxx.xxx.xxx.xxx
dns1=xxx.xxx.xxx.xxx
dns2=xxx.xxx.xxx.xxx
where xxx.xxx.xxx.xxx is the IP of your choice.
By that VoiceBuntu boots with static IP address on eth0 network device.
Enhancements:
- This live CD release is based on Ubuntu 7.04 and VoiceOne 0.5.0_2 due to a minor bug in VoiceOne version 0.5.0.
Download (417.1MB)
Added: 2007-07-14 License: GPL (GNU General Public License) Price:
832 downloads
Sofia-SIP 1.12.6
Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification. more>>
Sofia-SIP is an open-source SIP User-Agent library, compliant with the IETF RFC3261 specification.
Sofia-SIP project can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services.
The primary target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL.
Main features:
SIP features
- Sofia-SIP implementation follows RFC3261 and related key RFCs. INFO, UPDATE and REFER methods are supported. Also supported is SIMPLE presence and instant messaging, with the MESSAGE, SUBSCRIBE/NOTIFY and PUBLISH methods. Features such as early sessions, provisional responses, early media, caller preferences and session timers are included. Full set of transports, including both TCP and UDP over either IPv4 or IPv6, are supported.
SIP Offer-Answer module
- Sofia-SIP provides an implementation of the SDP offer-answer negotiation as specified in RFC3264. This is an essential component in using SIP to establish media sessions such as VoIP and video conferencing.
NAT traversal support
- Support for STUN as specified in RFC3489. STUN functionality is available via a separate module, so it can also be used independently from the base SIP stack. SIP extensions such as symmetric response routing (RFC3581/rport) are supported as well.
SIP security support
- Signaling can be secured by use of SSL/TLS. Also HTTP basic and digest authentication methods are supported.
<<lessSofia-SIP project can be used as a building block for SIP client software for uses such as VoIP, IM, and many other real-time and person-to-person communication services.
The primary target platform for Sofia-SIP is GNU/Linux. Sofia-SIP is based on a SIP stack developed at the Nokia Research Center. Sofia-SIP is licensed under the LGPL.
Main features:
SIP features
- Sofia-SIP implementation follows RFC3261 and related key RFCs. INFO, UPDATE and REFER methods are supported. Also supported is SIMPLE presence and instant messaging, with the MESSAGE, SUBSCRIBE/NOTIFY and PUBLISH methods. Features such as early sessions, provisional responses, early media, caller preferences and session timers are included. Full set of transports, including both TCP and UDP over either IPv4 or IPv6, are supported.
SIP Offer-Answer module
- Sofia-SIP provides an implementation of the SDP offer-answer negotiation as specified in RFC3264. This is an essential component in using SIP to establish media sessions such as VoIP and video conferencing.
NAT traversal support
- Support for STUN as specified in RFC3489. STUN functionality is available via a separate module, so it can also be used independently from the base SIP stack. SIP extensions such as symmetric response routing (RFC3581/rport) are supported as well.
SIP security support
- Signaling can be secured by use of SSL/TLS. Also HTTP basic and digest authentication methods are supported.
Download (2.5MB)
Added: 2007-04-26 License: LGPL (GNU Lesser General Public License) Price:
920 downloads
LuxMapGenerator SDK 1.0
Create your own random map generator for the RISK style strategy game Lux. This SDK comes with API documentation and examples. The language used is Ja... more>> <<less
Download (36KB)
Added: 2009-04-03 License: Freeware Price: Free
203 downloads
OSP Toolkit 3.4.0
OSP Toolkit project is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321). more>>
OSP Toolkit project is a client side implementation of the ETSI OSP VoIP Peering protocol (ETSI TS 101 321).
The OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.
<<lessThe OSP Toolkit project was begun in 1998 and the code has been incorporated into many commercial and open source VoIP products.
Download (0.41MB)
Added: 2007-05-08 License: BSD License Price:
901 downloads
Linux LiveCD VoIP Server 2.0.23
Linux LiveCD VoIP Server can be used to provide a Vonage type service. more>>
Linux LiveCD VoIP Server can be used to provide a Vonage type service, or to create a voip pbx for a campus or business with up to thousands of SIP phones.
It is based on the Open Standard SIP Express Router (SER) and Asterisk. It can serve as a SIP Proxy, VoIP PBX, VoIP gateway or Class 5 Softswitch.
Main features:
- Easy Web user administration and real-time accounting.
- All in one solution to VoIP and SIP enable your business.
- Allows you to make your own SIP numbering plan. Centrex service.
- Can be connected to multiple A-Z wholesale termination providers and to your own PSTN termination gateway/router.
- Can do Least Cost Routing
- Includes nat traversal, stun server, media server for conference call bridge, voicemail to email, incomming virtual numbers (DIDs), follow me forwarding.
- Commercial pre-paid, post-paid and flat rate account support. No calling card (no b2bua in base system).
- Requires no software installation - it is a liveCD.
- Supports any SIP soft or hardware phones, such as popular XTen, Cisco ATA 186, Grandstream, Sipura, Bugetone, Linksys PAP2 and more.
- Supports SIP for Video Conferencing (Xten / CounterPath EyeBeam)
- Supports Ecrypted SIP with Xten / Counterpath Secure Xten Pro
- Requires a PC with fixed ip connection to the internet. 256 MBytes of RAM, CDRom reader, ide or sata hard disk to store call and user database and web site.
- Remote ssh configuration and administration help
Enhancements:
- Kernel 2.4.34.1 and minor bugfixes.
<<lessIt is based on the Open Standard SIP Express Router (SER) and Asterisk. It can serve as a SIP Proxy, VoIP PBX, VoIP gateway or Class 5 Softswitch.
Main features:
- Easy Web user administration and real-time accounting.
- All in one solution to VoIP and SIP enable your business.
- Allows you to make your own SIP numbering plan. Centrex service.
- Can be connected to multiple A-Z wholesale termination providers and to your own PSTN termination gateway/router.
- Can do Least Cost Routing
- Includes nat traversal, stun server, media server for conference call bridge, voicemail to email, incomming virtual numbers (DIDs), follow me forwarding.
- Commercial pre-paid, post-paid and flat rate account support. No calling card (no b2bua in base system).
- Requires no software installation - it is a liveCD.
- Supports any SIP soft or hardware phones, such as popular XTen, Cisco ATA 186, Grandstream, Sipura, Bugetone, Linksys PAP2 and more.
- Supports SIP for Video Conferencing (Xten / CounterPath EyeBeam)
- Supports Ecrypted SIP with Xten / Counterpath Secure Xten Pro
- Requires a PC with fixed ip connection to the internet. 256 MBytes of RAM, CDRom reader, ide or sata hard disk to store call and user database and web site.
- Remote ssh configuration and administration help
Enhancements:
- Kernel 2.4.34.1 and minor bugfixes.
Download (MB)
Added: 2007-03-25 License: Free To Use But Restricted Price: $399
636 downloads
Verona 0.9.3
Verona is a portable VOIP toolkit based on eXosip, oSIP, oRTP, curl, and ffmpeg. more>>
Verona is a portable VOIP toolkit based on oRTP, oSIP, eXosip, curl, and ffmpeg.
The toolkit includes a phApi module, which is a high level VOIP API that allows a developer to create voice/video over IP applications in a dozen lines of code.
It includes an example of a minimalistic SIP user agent that supports audio and video transmission. The toolkit is basically the VOIP engine used by the openwengo project.
Enhancements:
- This release optimizes and simplifies the build system.
- It adds support to cross-compile for ARM Linux.
- It adds support for transmission of DTMF in the SIP INFO packet.
- It is compatible with the latest version of Intels IPP.
<<lessThe toolkit includes a phApi module, which is a high level VOIP API that allows a developer to create voice/video over IP applications in a dozen lines of code.
It includes an example of a minimalistic SIP user agent that supports audio and video transmission. The toolkit is basically the VOIP engine used by the openwengo project.
Enhancements:
- This release optimizes and simplifies the build system.
- It adds support to cross-compile for ARM Linux.
- It adds support for transmission of DTMF in the SIP INFO packet.
- It is compatible with the latest version of Intels IPP.
Download (13.6MB)
Added: 2006-10-25 License: GPL (GNU General Public License) Price:
1095 downloads
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