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Experimental open source project, which aims to test possibility of VoIP calls "from browser to browser". Main part is Java application implementing UDP hole punching, which is sent to both clients using JWS. See it online in action at project homepage
VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
Put idle assets to work with competitive interest rates, borrow without selling, and trade with precision. All in one platform.
Geographic restrictions, eligibility, and terms apply.
AsterFax provides an Email to Fax gateway for Asterisk. AsterFax lets you send an email by Fax. Enter the phone no. in the 'To' address, compose your email message and click send. You can also fax a MS-Word document or other attachment.
miTester for SIP is an automated SIP testing tool designed and developed to take care of the complex pre-deployment testing of SIP applications easily. This SIP testing tool can be used to simulate SIP call-flows & automate functional, regression tests.
For developers who are seeking to integrate SMS functionality into their software or applications, this project offers a range of simple solutions in the form of our SOAP, HTTP and SMTP API's that will allow seamless integration with our systems.
The Genalyze log analyzer, a simple yet highly extensible framework for parsing log files, was designed for parsing T-Server log files generated by Genesys contact center software platform, yet is generic and easy enough to adapt for other log formats.
Prototype testbed implementation of the IETF Media Server Control (MEDIACTRL) SIP Control Framework, comprehensive of both control and processing functionality (as in IMS MRF, Media Resource Function).
Java Media Library for wiring together Sinks and Sources. Can have rtp Sinks and Sources, FileRecorders/FilePlayers, Microphone, etc. The framework has default sink/sources already. It does not provide Text-to-Speech Source or Speech Recognizer Sink.
A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
J8051 is an API for communication between a 8051 microcontroller and a cell phone using Java ME, Bluetooth and "C" language callback functions, providing a easy way for embed systems developers build a communication between.
Olyo is an open source project which aims at implementing a prototype for Peer-to-Peer Session Initiation Protocol (P2PSIP) and providing real-time multimedia services over Internet based on this prototype.
Olyo is promoted by MINE lab, BUPT.
Java version of Dialogic's SwitchKit API. This API is actually a more "Java Centric" API, with listeners, filters, and a very extensible and modular framework that finally uses JNI to access the native libraries provided by Dialogic.
RTP text/t140 Library is a reference implementation for RTP Payload Type for Text Conversation (RFC 4103). The library has source code for encoding and decoding RFC 4103 data, and may be used either as a plug-in to JMF or in a separate RTP sender/receive
Xen-Agi is an open source java library that allow you to build java applications that interact with the Asterisk Server for VOIP and PBX functionality. This project supports the FastAGI protocol exclusively and is inspired by the Asterisk-Java project.
The Jive API is a lightweight Java framework for writing IVR applications.
The premise for the Jive API is to allow powerful IVR applications to be written without the user having to worry (or know) about the underlying IVR platforms.
SIP to Skype Gateway/Bridge/Converter/Adapter. Make and receive SIP to Skype Calls and Skype to SIP Calls. Call Skype users using speed dial or use SkypeOut. Make SIP calls from Skype using a SIP provider or SIP PBX. Use as a Skype Trunk with a PBX.
VoIP Toolkit / Call Control with Integrated Media. High-level Java API for creating SIP enabled VoIP applications. Suitable for either the desktop (softphone, phone applet, incoming call gatekeeper) or server-side (auto attendant, ACD, voicemail).
Emergency Context Resolution with Internet Technologies
The project is based on a draft from files draft-ietf-ecrit-*
Developed by IETF ECRIT Working Group
TRAUMA helps you to analyze your telephone bills of the Deutsche Telekom (main German telephony provider). | TRAUMA wertet Telefonrechnungen der Deutschen Telekom aus, wie man sie unter RechnungOnline als SYLK-Tabellendatei herunterladen kann.