Free sip software for mac

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.

Sip Audio Feeding Agent is an easy to use tool to capture the microphone input stream from sound card, sends audio stream (uLaw) to clients using RTP/SIP.
You must have a properly configured sound card in your OS to use this software. If you want to capture live radio, use an audio connector to connect radio output to the sound cards input.

System requirements:
- Java

OpalVOIP is a C++ multi-platform, multi-protocol library for Fax, Video and Voice over IP and other networks.
Opal runs on Mac OS X, Linux, Windows, Solaris, xBSD and Windows Mobile.
NOTE: OpalVOIP is licensed and distributed under the terms of the Mozilla Public License 1.0 (MPL).

Main features:
- Low latency RTP stack designed specifically for real-time multimedia
- Full featured H.323, SIP and IAX2 protocol stacks
- Audio codecs including G.711, GSM06.10, Speex and iLBC.
- Video codecs including H.261 and H.263
- Run-time loadable codec interface for proprietary or codecs such as G.729, H.263, H.264 and MPEG4
- Completely Open Source using the commercially friendly MPL 1.1 license

System requirements:
-
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- PTlib

Enhancements
- Fixed race condition in silence insertion algorithm where silence buffer is zero length before the first packet is received. Some downstream channels cant handle this. Now initialise silence buffer to be at a minimum 10 milliseconds.
- Fixed compile without video
- Backport of 22121 from trunk Reintroduce implicit check for NULL buffer accidentally removed in revision 22089
- Fixed empty Alert-Info field being sent in SIP INVITE.
- Fixed display name member of OpalConnection containing ONLY the display name for SIP.
- Fixed call routing issue in simpleopal
- IF we use string options to override the username (Calling-Party-Number) then we override it, no ifs buts or maybes!
- Fixed incorrect detection of remote SIP client putting local side on hold, thanks hongxing.
- Fixed incorrect trace log messages.
- Fixed fax NSE tone detect of CNG/CED being passed up to connection and application, for some reason commented out! Fixed OpenPhone detection that fax mode is already in use when get a CNG/CED indication.
- Fixed switching media format for existing channel, e.g. T.38 mode, was broken by someone.
- Slight change to routing algorithm so if the B-Party is explicitly determined, even when there is no matching "source" route in the table.

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.

Peers is a minimalistic and free softphone, developed using Java. Peers will allow a user to call from one PC to another on a local area network, using SIP/RTP/SDP with Ulaw encoding for voice.

System requirements:
-

Enhancements
New features:
- register management (initial register, register refresh, unregister)
- authentication using message digest (RFC2617)
Improved features:
- media capture/sending using pipes and three threads
- using TestNG for tests
- no singleton is used anymore
- xxxRequestManagers and xxxMethodHandlers are instanciated only once for uas and uac
- provisional responses can create or update dialog info (remote target, etc.)
Bugs fixed:
- 1994625 provisional responses with to-tag