sipphone

SipUnit provides a test environment geared toward unit testing SIP applications. SipUnit project extends the JUnit test framework to incorporate SIP-specific assertions, and it provides a high-level API for performing the SIP operations needed to interact with or invoke a test target.

A test program using the SipUnit API is written in Java and acts as a network element that sends/receives SIP requests and responses. The SipUnit API includes SIP User Agent Client (UAC), User Agent Server (UAS), and basic UAC/UAS Core functionality - the set of processing functions that resides above the SIP transaction and transport layers - for the purpose of interacting with the test target.

SipUnit uses the JAIN-SIP reference implementation as its underlying SIP stack/engine. The primary goal of SipUnit is to abstract the details of SIP messaging/call handling and facilitate free-flowing, sequential test code so that a test target can be exercised quickly and painlessly.

A test program using SipUnit API:

1. Extends SipTestCase
2. Creates SipUnit API objects - SipStack, SipPhone, SipCall, etc.
3. Calls methods on the object(s) to set up and initiate action toward a SIP test target. For example: SipPhone.makeCall("sip:roger@nist.gov", SipResponse.OK, ....) makes a vanilla call to sip:roger@nist.gov and blocks until an OK is received or a timeout occurs. The test target could be any node up to and including the final destination of the INVITE request message.
4. Verifies the results of the action involving the test target using both the SIP-specific assert methods provided by SipUnit and the standard JUnit assert methods. For example: assertHeaderContains(sipCall.getLastReceivedResponse(), "From", "sip:amit@nist.gov"), assertEquals("Unexpected response received", SipResponse.OK, sipCall.getReturnCode()).

Here are some key features of "SipUnit":

· A basic set of SIP-specific assert methods - assertHeaderPresent(), assertHeaderContains(), assertBodyPresent(), etc.
· High level API for interacting with a test target.
· Low-level SIP messaging access for interacting with a test target.
· Registration/unregistration and call processing with or without authentication (DIGEST).
· Support for testcase-specified timeouts.
· Support for different routing configurations.

Whats New in This Release:

· Support was added to the SipPhone and SipSession classes for running SipUnit tests from behind a NAT and communicating with a SIP server on the Internet.
· A STUN example test was included.
· An enhancement that allows more flexible multiple SIP stack creation was incorporated.

vEmotion for VoIP description
vEmotion for VoIP - Support all voice enabled IMs and VoIP clients vEmotion for VoIP supports voice enabled IMs and VoIP clients.

vEmotion is a perfect VoIP audio assistant. It flawlessly record VoIP calls into MP3 or WAV files with data and infos encrypted and password protected. Also add life to your VoIP voice convsations. Enjoyable things include background music, voice emotion (lively audio clips), text to speech that enable you to talk without openning your mouth.

vEmotion is the perfect tool for podcasting, online journalism, conducting business, and much more!

The program is compatible with nearly all VOIP clients, among which Skype, ICQ, Msn/WLM, Gtalk, AIM/AIM Triton/AIM Pro, Yahoo Messenger and QQ were fully tested and explicitly supported. Other VoIP clients, such as voipcheap, teltel, sipphone, are supported via simple user customization. You can tune up all these IM and VOIP clients with single installation of the software.

Here are some key features of "vEmotion for VoIP":

· All pc based (pc to pc, pc to phone, phone to pc) VoIP calls can be recorded.
· Record calls into mp3 with rate up to 48000 Hz. Thats better than CD quality!
· Auto beginning and auto end recording. You wont miss any call or oppositely record a blank.
· Single channel, double channels (save your voice and your callers in separate channels), and single channel double rate, three recording modes available.
· Automatically select record file name according to preset format strings.
· Epoch recorded. Retrieve the exact time you said some words even after 50 years!
· Save memo together with call records. And real-time memoire let you write down any idea before you forget.
· Save multiple records for a single call, manually or automatically.
· Have data and information encrypted for safety. Also take passwords to protect your records against unauthorized access.
· Manage (sort, find, star, replay, share) call records very easily.
· Two methods to share records with friends: send records via email or play records directly within vEmotion for callers.
· Ability to run as a background service (stealth mode) without showing any user interface.
· Plus an general pc recorder make it a perfect tool for podcasting, online journalism, conducting business, and much more!
· Background music make VoIP conversations more romantic.
· Send voice emotions (lively audio clips) to your friends to enjoy more vivid talking.
· Play a greeting message at beginning of a call. Greeting messages can be recorded from within vEmotion using the microphone recorder.
· TTS (text-to-speech) ability. You can talk without openning your mouth!
· Skype, Icq, Msn / Wlm, Gtalk, Aim / Triton / Pro , Yim, and QQ are explicitly supported and fully tested.
· By user customization, supporting list expands to include nearly all voice enabled IM and VoIP clients. To list a few, Sipphone, VoIPCheap, TelTel, Sipdiscount , Harbibi, VoIPStunt, Express talk, and so on.
· Automatically upload call records to a server computer running Call Record Center. You can manage a large number of call records collectively.
· Anyone can use windows will feel at home with it.
· Feel free to talk as you talked. It at no time intervenes with your talking or messes up your system.
· Totally free for download, and you can experience all its features without paying.

Zfone is my new secure VoIP phone software, which lets you make secure phone calls over the Internet. It encrypts your call so that only the other person can hear you speak. Zfone lets you whisper in someones ear, even if their ear is a thousand miles away.

In the future, the ZRTP protocol used by Zfone will be integrated into standalone secure VoIP clients, but today we have a software product that lets you turn your existing VoIP client into a secure phone. The current Zfone software runs in the Internet Protocol stack on any Windows XP, Mac OS X, or Linux PC, and intercepts and filters all the VoIP packets as they go in and out of the machine, and secures the call on the fly.

You can use a variety of different software VoIP clients to make a VoIP call. The Zfone software detects when the call starts, and initiates a cryptographic key agreement between the two parties, and then proceeds to encrypt and decrypt the voice packets on the fly.

It has its own little separate GUI, telling the user if the call is secure. Its as if Zfone were a "bump on the cord", sitting between the VoIP client and the Internet. Think of it as a software bump-on-the-cord. Maybe a bump in the protocol stack.

Zfone has been tested with these VoIP clients: X-Lite, Gizmo, and SJphone. And these VoIP service providers: Free World Dialup, iptel.org, and SIPphone.

A good way to test your VoIP client is to make a call to a server that simply echoes your voice back to you, just so you can see if it works at all. A good example of an echo server is the one offered by Free World Dialup. To call that, enter this SIP URI in the "Call" text box: 613@fwd.pulver.com, then press the call button.

To make your VoIP client work with Zfone, you will need to open your VoIP clients preference panel and set the SIP port number to 5060.