Open Source Java VoIP Software

Ever wanted to cut out background noise when talking with others on Skype? Now it's possible! NoiseGator is a light-weight noise gate application that routes audio through an audio input to an audio output. In real-time the audio level is analysed and if the average level is higher than the threshold the audio bypasses as normal. However, if the average level goes below the threshold, the gate closes and the audio is cut. When used with a virtual audio cable it can act as a noise gate for a either a sound input(microphone) or sound output(speakers). Can also be used to gate noise from your own mic or play your microphone through your speakers. REQUIREMENTS: - Java 7 or higher for Windows. - Java 6 or higher for Mac. Java 7 recommended. - A virtual audio cable is required for use with VOIPs: For Windows users I recommend the VB-Cable driver (http://vb-audio.pagesperso-orange.fr/Cable/index.htm). Mac users can use SoundFlower.

JDA strives to provide a clean and full wrapping of the Discord REST api and its Websocket-Events for Java. This library is a helpful tool that provides the functionality to create a discord bot in java. Discord is currently prohibiting the creation and usage of automated client accounts (AccountType.CLIENT). We have officially dropped support for client login as of version 4.2.0! Note that JDA is not a good tool to build a custom discord client as it loads all servers/guilds on startup, unlike a client which does this via lazy loading instead. If you need a bot, use a bot account from the Application Dashboard. Creating the JDA Object is done via the JDABuilder class. After setting the token and other options via setters, the JDA Object is then created by calling the build() method. When build() returns, JDA might not have finished starting up. However, you can use await ready() on the JDA object to ensure that the entire cache is loaded before proceeding.

jphonelite is a Java SIP VoIP SoftPhone for Desktops (Windows, Linux, Mac) and Android. Features 6 lines with transfer, hold, conference (up to all 6 lines), g711 u/a, g722, g729a, and video (video support in Linux or Windows only and includes H263/H264/VP8). Applet includes full JavaScript support. STUN/TURN/ICE supported. Encrypt media with SRTP. DTLS Key Exchange. Video conferencing with jPBXLite/0.12+ NOTE : THIS PROJECT HAS BEEN SUPERSEDED BY jfPhone. Please go to jfphone.sourceforge.net to get latest version.

Asterisk PBX Integration Zimlet is an Extension for Zimbra Collaboration Suite. The Zimlet does Interface with the Asterisk Manager Interface to integrate with Asterisk PBX. The main focus is dial-on-click for Phone numbers inside Contacts and Emails.

A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated: https://sourceforge.net/donate/?user_id=2370023

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.

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. .

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

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.

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.

A complete contact center solution for the Asterisk PBX, capable of handling inbound and outbound campaigns. Provides an agent toolbar for CTI. For managers an admin panel controls every aspect of the call center. Stunning historical and realtime stats.

Accent (Advanced Component Control Enhancing Network Technologies) is a comprehensive tool suite that interfaces a variety of communications systems and allows these systems to be controlled through goals (high-level user aims) and policies (lower-level system rules). Accent has been applied to the domains of: o Call Control: for telecommunications, particularly call control in Internet telephony o Home Care: for home automation and telecare, particularly for domestic appliances and devices to support home care o Sensor Networks: sensor networks, particularly managing wind farms However, the approach is generic and extensible for managing other kinds of systems. Accent is essentially software. As most of Accent in in Java, it needs at least a Java Runtime Environment but more likely a Java Software Development Kit. To interact with the real world, Accent needs relevant hardware such as a communications server (softswitch) or home automation equipment.

Cross-platform Java Teamspeak client

JAVA Asterisk Predictive Dialer using AMI (Asterisk Manager Interface)

Use this when attachments aren't offered with your email service. (i.e. Proxy Mail). Simply Supply a link to download the 11KB Audio Player and Copy/Paste your Message into the Email. See Website For Demo. http://audiotext.sourceforge.net

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

Session Initialtion Protocol (SIP) is widely used for telephone services over the Internet. CafeSip provides a suite of open-source tools and applications for creating customized SIP services and applications using the Java.

Asterisk based callcenter system. Administration interface to manage queues, agents, accounts, skills - track calls, create statistics. Realtime queue monitor. Agent web interface with build in IAX softphone.

Castadiva is a test bed based on low-cost off-the-shelf devices, which is used to test protocols developed for MANETs. Castadiva is completely compatible with the file format used by the ns-2 simulator.

client-server full-java chat, provided with audio&video (JMF), custom rooms, custom forums, instant- messaging and file sharing

A secure VoIP client for Google Android based on SipDroid (http://sipdroid.org). CounterTap uses SRTP to secure the media stream. Please read the FAQs in the help forum before installing. NB: I will restart dev work later in 2010, pending resources.

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

ECharts is a state machine-based programming language for event-driven systems derived from the standardized UML Statecharts language. ECharts has a proven track-record in a large-scale commercial deployment. Take a look at what ECharts has to offer!

Visual Framework Tool to scan/sniff address space, enumerate users, crack credentials, pattern based dial spoofing and security reporting for Voip protocols. This software is intended to give a general framework to build and plug Voip protocol analizers in order to fix security issues and enhance VoiP platforms confidence. It IS NOT intended to be a cracking tool for malicious system breakers, but a really software MUST for security people to assure Voip platform security.

Google Contacts to Grandstream xml Phonebook Format Converter