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| - | = The SIPfoundry sipXezPhone =
| + | #REDIRECT [[sipXezPhone]] |
| - | == Introduction ==
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| - | SIPfoundry has made available a comprehensive set of SIP stacks, libraries, tools and SDKs to build SIP client solutions. Examples include the sipXtapi user agent SDK, which consists of sipX libraries such as [http://www.sipfoundry.org/sipXcallLib/ sipXcallib], [http://www.sipfoundry.org/sipXtackLib/ sipXtacklib], [http://www.sipfoundry.org/sipXmediaLib/ sipXmedialib], and [http://www.sipfoundry.org/sipXportLib/ sipXportlib].
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| - | Currently, the sipXezPhone [http://scm.sipfoundry.org/rep/sipX/main/sipXcallLib/examples/sipXezPhone source] is located in the examples directory under the sipXcallLib project.
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| - | sipXezPhone, for the first time, demonstrates the capability of this user agent SDK by implementing a simple softphone. sipXezPhone is intended to serve as an example implementation for developers to learn how to use sipXtapi. In a very simple way embeded SIP user agent applications can be built or different skins can be applied to a softphone implementation such as sipXezPhone.
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| - | sipXezphone is very different as compared to sipXphone. While sipXphone is a fully featured implementation of a SIP softphone that is derived from the original Pingtel xpressa JAVA softphone, sipXezPhone is much leaner. Constant improvements are being made to sipXezPhone as both the phone application as well as sipXtapi evolve to include new features.
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| - | sipXezPhone, is written using C++, wxWidgets, and the [http://scm.sipfoundry.org/rep/sipXcallLib/main/doc/sipXtapi/html/index.html sipXtapi API]. sipXtapi and sipXezPhone are built to be platform and operating system agnostic. Currently both Windows and Linux platforms are supported with MAC OS X a clear posibility.
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| - | == Screenshot ==
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| - | [[Image:ezPhone3.png]]
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| - | == List of Features ==
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| - | sipXezPhone is an easy to use SIP softphone with a simple graphical user interface and a small footprint. As such it is a good starting point for experimenting with SIP technology or to build your own softphone or embedded SIP client.
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| - | sipXezPhone currently supports the following list of features:
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| - | * Call Transfer
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| - | * Automatic Echo Cancellation
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| - | * Caller ID
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| - | * SIP URL calling
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| - | * Most recently used numbers list
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| - | * Volume and record level adjustments
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| - | * STUN support
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| - | The list for the next release includes: Custom ring tones, conferencing, integrated address book and a detailed call history.
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| - | == License ==
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| - | sipXezPhone as well as the sipXtapi user agent SDK are licensed under the L-GPL open source license like all the other sipX components on SIPfoundry.
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| - | This license specifically allows commercial usage of the library without the viral infection associated with GPL. Essentially, you may use the code as-is and build on top of it without exception, however, if you modify the existing code (e.g. bug fix), you must make changes available under LGPL. For more information on open source licensing or SIPfoundry, please see http://www.opensource.org or http://www.sipfoundry.org.
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| - | == Library Components ==
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| - | === sipXtapi User Agent SDK ===
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| - | The [http://scm.sipfoundry.org/rep/sipXcallLib/main/doc/sipXtapi/html/index.html sipXtapi SDK] is a C application programming interface for voice communications over IP. Specifically, sipXtapi provides a generalized telephony interface on top of the Session Initiation Protocol (SIP), RFC 3261, and the real-time Transport Protocol (RTP), RFC 1889. While the SIP and RTP protocols provide signaling and media transport infrastructure, sipXtapi also includes many other protocol and standards implementations needed for voice communications.
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| - | '''sipXtapi Features:'''
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| - | * TAPI-like API
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| - | * Multiple simultaneous calls
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| - | * Hold/Unhold, Mute/Unmute
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| - | * Volume and Gain control
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| - | * Selectable audio input device, speaker device, and ringer device
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| - | * Logical separation between ringer and speaker
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| - | * Silent Call Forwarding and Rejection
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| - | * Tone Generation (DTMF, busy, ring back, etc.)
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| - | * Play audio from file
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| - | * Bind to specific network interfaces
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| - | * RFC3261 compliant SIP stack
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| - | * RFC2833 Out of band DTMF tones
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| - | * RTP suppression on mute
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| - | * Configurable Ports (UDP, TCP, and RTP)
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| - | * Multiple SIP identities (line appearances)
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| - | * Line authentication
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| - | * DNS SRV timeout control
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| - | * SIP Proxy control
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| - | * Rport
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| - | * G711 Codec
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| - | * Fixed non-adaptive jitter buffer
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| - | == More Information ==
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| - | For more information on the sipXezPhone and sipXtapi projects, please refer to the [http://www.sipfoundry.org/sipXcallLib/ sipXcalllib project].
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| - | '''Configuration:''' The [http://sipx-wiki.calivia.com/index.php/SipXezPhone_-_Configuration_Help Configuration help] page explains some of the configuratin settings for sipXezPhone.
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| - | '''Platform Support:'''
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| - | *'''Windows:''' [[HowTo compile sipXezPhone|Build instructions]] are available for the Windows platform.
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| - | *'''Linux:''' An experimental [[SIPfoundry.org sipXezPhone Ebuild for Gentoo Linux|ebuild for Gentoo Linux]] is also available.
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| - | For more information, or if you'd like to contribute to the project, please join the [https://list.sipfoundry.org/mailman/listinfo/sipx-dev sipX-dev mailing list] or directly email Michael Cohen (mcohen@pingtel.com), one of the core developers.
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