voip
iVoIP 0.8.0
An open source VoIP client for the iPhone more>> An open source VoIP client for the iPhone
This project is an open source VoIP client for the iPhone
Current status:
- ported SIP client to iPhone -- can hear audio (playout works)
- Athough iVoIPs audio capture works the playout stops.
- trying to figure out how to make audio-capture and playout work simultaneously
System requirements:
- iPhone
Express Talk VoIP Softphone 3.21
Voip application to let you make calls through your computer. more>> Voip application to let you make calls through your computer.
Express Talk softphone is a software that works like a telephone to let you make calls through your computer. You can call anyone on the internet who has installed it (or any other SIP softphone).
Calls computer to computer are always free. You can also call ordinary real telephone numbers anywhere in the world if you sign with a VoIP gateway service company. See here for some recommended telephone gateway companies.
Business users might like to upgrade to Express TalkBE Business Edition. The Express Talk Business Edition offers extra features designed specifically with business users in mind.
These include call conferencing, the ability to transfer calls, call recording, push to talk and more advanced configuration options.
Express Talk also works perfectly as an extension with free VoIP Virtual PBx Systems. If you are looking for a complete telephone system to run on your office server read here.
Main features:
- Make voice and video calls free between computers.
- Supports computer to phone via a VoIP SIP gateway provider.
- Configure up to 6 lines on the one phone with the ability to put calls on hold.
- Supports caller ID display and logging.
- Includes a phone book with quick dial configuration.
- Integrates with Microsoft Address Book.
- Supports call transfer (Business Edition).
- Record phone calls to wav (Business Edition).
- Allows up to 6 people to join one call using the Call Conferencing feature (Business Edition).
- Provides easier communication using the Push to talk intercom (Business Edition).
- Includes Do Not Disturb button (Business Edition).
- Is available for PocketPC so you can take your VoIP numbers with you wherever you go.
- One user Business Edition license can be used for both computer and Pocket PC versions.
- Supports emergency numbers calling (such as 911, 999, 000).
- Works with a headset or in speakerphone mode with just a standard microphone and set of speakers.
- Webcams or usb video phones can be configured for video calls.
- Works seamlessly with recommended IP phones and usb phones.
- Can be configured to work behind NATs and Firewalls.
- Includes data compression (GSM, uLaw, ALaw, PCM and G726), echo cancellation, noise reduction, comfort noise and more.
- Supports video (H263, H261) for webcams or IP Phones (Windows version only).
- Uses the standard SIP protocol so it can use a number of telephone gateways, SIP systems or other internet phone software. Click here for a list of SIP service providers.
- Works with our VoIP Virtual PBx to create a LAN based PBx for offices or call centers.
- Used with the VRS Call Recorder, this softphone can record and save phone calls to MP3, wav and more.
- Plays on-hold music to callers on hold. Can also link to the IMS On-Hold Messages Player Software to create professional mixes of music and messages on the fly.
System requirements:
- Broadband connection.
- Optional headset.
Enhancements
- Various bug fixes including intercom, muting, address book.
Voix Phone 1.0.2
Voix Phone is a multiplatform IAX soft phone more>> Voix Phone is a multiplatform IAX soft phone
Voix Phone is a multiplatform IAX soft phone with an engine derived from Voix Manager, the powerful Asterisk call manager interface, from wich it inherits its stability and robustness.
Voix Phone has been thought with simplicity in mind, all feature needed by the user, fast and easy usable, with the minimum configurations, just fill the phone login information and play.
IAX is one of the least VoIP signaling standard that eliminates the problems imposed upon the competing SIP standard by NAT firewalls. IAX is supported primarily by Asterisk.
NOTE: Freeware for non commercial use.
Main features:
- IAX/IAX2 protocols
- Call transfer
- Calls Incoming status
- Redial
- Access voice mail message with one button
- Missed call Log
- Dial Missed call
- Hold function
- Hold status (number of users waiting)
- Quick dial
- Mute
- Logs
- Support for multiple audio devices
- Available codecs GSM, ulaw, alaw, speex, ilbc
- DTMF tones sending
- Echo cancellation
- Adaptive Jitter Buffer
- Address book
- Automatic user registration
- Account password encryption
System requirements:
- Processor: minimum PowerPC G4
- Memory: minimum 256 MB
- Internet connection: wired or wireless
Enhancements
- Fixed some bugs
- Improved the interface
- Added compatibility to leopard
- Added DND and call Forwarding features.
Voix Phone Mac 1.0.2
Voix Phone Is a multiplatform IAX soft phone, its engine derives from Voix Manager, the powerful Asterisk call manager interface, from wich it inherits stability and robustness. Voix Phone has been thought with simplicity in mind, all feature needed by the user, fast and easy usable, with the minimum configurations, just fill the phone login information and play. more>>
Voix Phone Mac - Voix Phone Is a multiplatform IAX soft phone, its engine derives from Voix Manager, the powerful Asterisk call manager interface, from wich it inherits stability and robustness.
Voix Phone has been thought with simplicity in mind, all feature needed by the user, fast and easy usable, with the minimum configurations, just fill the phone login information and play.
We hope that this our contribution could be useful to who requires of a simple but advanced soft phone, Voix Phone is distributed freeware for non commercial use.
Why IAX ?
IAX is one of the least VoIP signaling standard that eliminates the problems imposed upon the competing SIP standard by NAT firewalls. IAX is supported primarily by Asterisk.
Enhancements:
Version 1.0.2
Fixed some bugs, Added call Forwarding and DND features, Added full compatibility to Leopard
System Requirements:<<less
OpenSIPS 1.4.4
An open source SIP server implementation more>> An open source SIP server implementation
OpenSIPS is an GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and quality) to be used in professional SIP server platforms. OpenSIPS started as a fork of Fokus Fraunhofer SIP Express Router (SER) project.
OpenSIPS wants to be a more open project, not only from license point of view, but more open as project management, especially for external contributions.
OpenSIPSs goal is to overcome the development latency of current SER project and to ensure a shorter path into a release for new added features.
Main features:
- robust and performant SIP (RFC3261) Registrar server, Location server, Proxy server and Redirect server
- small footprint - the binary file is small size, functionality can be stripped/added via modules
- plug&play module interface - ability to add new extensions, without touching the core, therefore assuring a great stability of core components
- stateless and transactional statefull SIP Proxy processing
- support for UDP/TCP/TLS/SCTP transport layers
- IPv4 and IPv6
- support for SRV and NAPTR DNS
- SRV DNS failover
- IP Blacklists
- multi-homed (mhomed) and multi-domain support
- flexible and powerful scripting language for routing logic
- variables support in script - script variables, pseudo-variables (access to the SIP messages), AVPs (values persistent per SIP transactions)
- management interface via FIFO file and unix sockets
- authentication, authorization and accounting (AAA) via database (MySQL, Postgress, text files), RADIUS and DIAMETER
- digest and IP authentication
- Presence Agent support (many additional integration features)
- XCAP support for Presence Agent
- CPL - Call Processing Language (RFC3880)
- SNMP - interface to Simple Network Management Protocol
- management interface (for external integration) via FIFO file, XMLRPC or Datagram (UDP or unixsockets)
- NAT traversal support for SIP and RTP traffic
- ENUM support
- PERL Programming Interface - embed your extensions written in Perl
- Java SIP Servlet Application Interface - write Java SIP Servlets to extent your VoIP services and integrate with web services
- load balancing with failover
- least cost routing
- support for replication - REGISTER offer new functions for replicating client information (real source and received socket).
- logging capabilities - can log custom messages including any header or pseudo-variable and parts of SIP message structure.
- modular architecture - plug-and-play module interface to extend the servers functionality
- gateway to sms (AT based)
- multiple database backends - MySQL, PostgreSQL, Oracle, Berkeley, flat files and other database types which have unixodbc drivers
- straightforward interconnection with PSTN gateways
- dialog support (call monitoring, call termination from proxy side, call profiling)
- XMPP gateway-ing ( transparent server-to-server translation)
- impressive extension repository - over 70 modules are included in OpenSIPS repository
Scalability:
- OpenSIPS can run on embedded systems, with limited resources - the performances can be up to hundreds of call setups per second
- used a load balancer in stateless mode, OpenSIPS can handle over 5000 call setups per second
- on systems with 4GB memory, OpenSIPS can serve a population over 300 000 online subscribers
- system can easily scale by adding more OpenSIPS servers
- OpenSIPS can be used in geographic distributed VoIP platforms
- straightforward failover and redundancy
Enhancements
- After another month from 1.4.3 release, OpenSIPS improves itself with a new minor release, 1.4.4. Thanks to hard testing and fixing of a several people, new issues (critical and minor) were fixed on the OpenSIPS 1.4 branch.
- It is highly recommended to upgrade to this release, as it provides important stability improvements - OpenSIPS 1.4.4 is now available for download on project web site and SF download system.
Ventrilo 2.3.2pt15
Ventrilo - VoIP group communications software more>>
By offering surround sound positioning and special sound effects on a per user, per channel, per server or global configuration level the program provides each user the option to fully customize exactly how they wish to hear sounds from other users or events.
Ventrilo is best known for its superior sound quality and minimal use of CPU resources so as not to interfere with day to day operations of the computer or during online game competitions.
It is also preferred for the simple user interface that any first time computer user can very quickly learn because the most commonly used features are immediately visible and can be activated with a single click of the mouse.
Ventrilo can be used for home / personal applications like talking to friends and family, or playing organized online games where Voice Comm can make them more exciting and productive.
CommuniGate Pro 5.2.18
CommuniGate Pro is appreciated to be a beneficial Internet messaging server application implementing which provides an integrated platform for Store-and-Forward (E-mail, Calendaring) and Real-Time (VoIP, Video, Instant Messaging, White Boards) communications over IPv4 and IPv6 networks. more>> <<less

Vonage Companion 1.0.2
Vonage Companion makes it possible for you to make and receive phone calls over your Vonage Pro account on your Mac. more>>
Vonage Companion 1.0.2 makes it possible for you to make and receive phone calls over your Vonage Pro account on your Mac. When using Vonage Companion, incoming calls can ring both your telephone connected to the Vonage router and your Mac running Vonage Companion.
Now you can get unlimited long distance in the USA and cheap long distance around the world using VoIP technology.
- Mac OS X 10.4.11 or later, Vonage Pro account.
Virtual VideoPhone 1.1
Virtual VideoPhone - Telephone with real-time video more>>
Our yakForFree service requires that both the caller and recipient download and join the yakCommunity. To make calls outside of the yakCommunity, check out yaks WorldCity VoIP products and services: yakToAnyone, yakBasic, and yakUnlimited. Various paid plans add more features, such as 911 service, local number portability, and calls to regular phones.
Main features:
- FREE yak Member-to-Member Worldwide Calling
- FREE Virtual VideoPhone Gadget
- Clear High Quality Voice and Video Calls
- FREE member-to-member calling
- 2-way VideoPhone Calling (Requires PC camera)
- Access into the yakCommunity
- PC Compatible Intel Pentium 4 or AMD Athlon, 1700 MHz (or better)*
- Mac OS X version 10.2 or later, PowerPC G3 or better.
Ventrilo 3.0.4
VoIP group communications software. more>> VoIP group communications software.
Ventrilo is the next evolutionary step of Voice over IP (VoIP) group communications software. Ventrilo is also the industry standard by which all others measure them selves as they attempt to imitate its features.
By offering surround sound positioning and special sound effects on a per user, per channel, per server or global configuration level the program provides each user the option to fully customize exactly how they wish to hear sounds from other users or events.
Ventrilo is best known for its superior sound quality and minimal use of CPU resources so as not to interfere with day to day operations of the computer or during online game competitions.
It is also preferred for the simple user interface that any first time computer user can very quickly learn because the most commonly used features are immediately visible and can be activated with a single click of the mouse.
If voice communication isnt your cup of tea then you can always use Ventrilo as a more reliable form of IRC without the hassles or frustration of trying to teach someone how to setup a real IRC client.
Ventrilo can be used for home / personal applications like talking to friends and family, or playing organized online games where Voice Comm can make them more exciting and productive.
Main features:
- Voice communication with multiple people.
- Cross channel communications.
- Phantom users for listening in on other channels.
- User-to-user private conversations.
- Individual channels that can be created dynamically.
- Sub channels.
- Muted channels.
- Queued channels.
- Channel transmit time limits.
- Channel client limits.
- Channel feature filtering.
- Password protected server login and channels.
- Advanced channel control options and filters.
- Individual admin passwords for channels.
- Text-to-speech (TTS) voice generation.
- Key binding to execute special program functions, play wave files, send TTS messages.
- Separate phonetic spelling of user and channel names for proper TTS.
- Built in chat similar to IRC.
- Users can enter dynamic comments for all other users to see.
- User assigned names for server connections. No need to remember IP numbers.
- Wave file or TTS event notifications for program events.
- Remote administration of server features and current users.
- Remote administration of persistent server properties.
- Mute Microphone and Mute Sound with key binds for both.
- User adjustable sound effects.
- Persistent mute states for specific users on a given server.
- Persistent channel admin passwords.
- Persistent special effects including surround sound for users, channels, servers, global.
- Built in server browser.
- Ability to record and playback voice streams. Great for watching demos with associated voice comms.
- Voice activation or Press-to-talk transmission modes.
- Binds to change channels or cycle through all available channels.
- Multiple platform support for servers.
- Multiple platform support for clients.
- Servers are now ready and supported on 64bit platforms.
- The clients and servers do NOT contain any Ad Ware or Spy Ware features.
- Ability to restrict server admin features. Useful for hosting services.
- Server specified codecs to control voice quality and bandwidth usage.
- User customization of display modes.
- User customization of user list icons and toolbar icons.
- Clients will soon support *NIX platforms.
- Portable cross platform server.
System requirements:
- Processor: 1.0 GHz or higher.
- Sound Input and Output device.
- Microphone.
Enhancements
- Fixed crash when using Russian language desktop, and possibly languages other than English.
- Modified Setup-Voice and Bindings editor so that Mouse-1 cannot be selected as a hotkey.
- Modified bindings editor so that 16 users / channels can be defined per hotkey (instead of the old 4 limit).
- Added support for Special Effects (SFX). But only for Volume control at this time. Hope to have more advanced SFX in 3.0.5 client.
- Fixed Complaint List window so that month value of event is displayed correctly. November was showing up as 10 instead of 11.
- Fixed bug with phantoms not being added or deleted correctly.
Asterisk Voicemail for iPhone 0.13
Allows you to check your voicemail messages on your Asterisk phone system from your iPhone more>> Allows you to check your voicemail messages on your Asterisk phone system from your iPhone
Asterisk Voicemail for iPhone allows you to check your voicemail messages on your Asterisk phone system from your iPhone (or iPod Touch). It has been suggested that I rename the software to something more generic, since the iPod Touch is also fully supported.
Asterisk Voicemail for iPhone is quite similar to the "Asterisk Voicemail for Apple Touch".. Thoughts?
It works similar to the native iPhone Visual Voicemail, allowing you to list messages, listen to messages, display caller-id information, delete messages, move messages, return calls and change voicemail settings all from your iPhone.
The technology behind it is Asterisk (The Open-Source VoIP PBX), PHP for the backend, Smarty and iUI for the frontend.
NoteThis software is a PHP web-application which resides on your Asterisk server. Dont be confused into thinking that this is a native iPhone application. If you dont run or manage an Asterisk server, then this is not for you.
System requirements:
- iPhone or iPod touch.
Enhancements
- Changed listen link to listen.php, removing the need for Apache Alias.
- Added g_debug_path. Debug path was hard coded which caused crash if not exist.
openSpeak 0.1 RC2
VoIP application made for gamers more>> VoIP application made for gamers
openSpeak is a free and open source VoIP solution that is aimed at clan and casual gamers who like to chat while playing a game.
Main features:
wxWidgets Client:
- Voice communication
- Text messaging
- Favorites management
- Adaptive ring-buffer size1)
- Adaptive per-user jitter-buffer size2)
- Anti-saturation mechanism
Standalone (CLI) Server:
- Servername/-MOTD support
- Password protection
- MaxUsers
- Querying the Server for informations
Enhancements
- Mac support (in addition of Linux and Windows)
- Improved netcode stability
- Better sound quality (thanks to a jitter buffer implementation)
- Better looking chat windows
- Lower latency (about -50% on LAN) (no yet in the SVN trunk)
