h323

Vidophone H323 Softphone description
Vidophone H323 is a leading H323 Softphone dialer in industry that works automatically under NAT with Stun protocol Vidophone H323 is a leading H323 Softphone dialer in industry that works automatically under NAT with Stun protocol. Best voice quality with G.723.1 and G.729/A codec, other codec supported iLBC, GSM, G711.

With Hold, mute functions, mic/spk volume control, audio device selection, auto login to web billing , call history, address book, Fully skinable, available for private label and logo.

Vidophone H323 dialer is perfect for Middle East market where some country may block sip port.

Here are some key features of "H323 Softphone":

· Using G.723.1, G.729/A for best quality, other codec supported iLBC, GSM, G711
· Fully Comply with ITU-T H323 latest recommendations
· PC to PC call, PC to Phone call, and Phone to PC call.
· Volume and Microphone gain control
· Full duplex audio .
· Selection of the audio device used by the softPhone (audio cards must be full-duplex).
· Support Fast Start Procedure, H245 In Setup, and H245 Tunneling.
· Adaptive Jitter buffer .
· Real time audio streaming.
· Silence Supression.
· DTMF sending in Q931 and RFC2833
· Works Automatically behind NAT with STUN
· Interoperability: Interoperable with most softphones,IP-Phones,Gateways,Gatekeepers, IP-PBX and other VoIP entities.
· Address Book and last call.
· Hold and Mute.
· Fully attractive skinable interface.
· Private Branding and load your logo.
· Auto Login to your billing website page.
· Low binary size approximately 1,6 MB in easy installable installer.
· Not using any .NET or Java Runtime Library


Requirements: NO .NET, VB, Java Runtime Library

Limitations: 30-days trial

Requirements:

· NO .NET, VB, Java Runtime Library


Limitations:

· 30 day trial

VOIP H323 Phone SDK description
VOIP H323 DLL Soft Phone SDK brings H.323 protocol support for ActiveX VOIP H323 DLL Soft Phone SDK brings H.323 protocol support for ActiveX. With this SDK one can create a simple program to connect and start speaking with anyone with a direct IP Address or use the Gateway or the Gatekeeper to connect to /or from PSTN lines.

Now VOIP open source projects have made it easier to implement a customized VOIP Solution eliminating costly PSTN lines. Though this implementation might turn out fairly complex our VOIP Implementation Team at Research-Lab will guide you remotely for the same.

Here are some key features of "VOIP H323 Phone SDK":

· IP-Address based direct local connectivity within LAN,
· Microsoft NetMeeting Compatible,
· Play and Record Wave File,
· Gateway Specification with/without Login,
· Gatekeeper Specification with/without Login, Proxy/Firewall support for Version 2.0 [i.e. for Registered Users],
· Voice/Ring Volume Adjustment [i.e. for Registered Users],
· Save a conversation to a wave file [i.e. for Registered Users],
· Easily Create a Answering Machine,
· Easily Create a Interactive Voice Response,
· Easily Create a Professional Call Center


Whats New in This Release:

· Added H323 Wave Library for wave file playback and recording

Voice&Video Internet/Lan Chat client based on H323 protocol.
Main features:
Direct PC-2-PC communication without any intermediary.
H323 Gatekeepers support.
Video communication with 176x144 (QCIF) and 352x288 (CIF) resolution.
Audio communication with a number of selectable codecs and quality.
Compatible with NetMeeting, GnomeMeeting and many other H323-based software and hardware.
Can transmit voice even through low bandwidth modem lines.
Easy to setup and configure.
Multilingual (English, Ðóññêèé and more...).
Its absolutely FREE and OpenSource! [Yeee!]

MyPhone is Voice&Video IP Telephony Chat client.
It is H323 protocol compatible, based on OpenH323 project.
Client is written in VC++, using MFC/PWLib libraries, compatible with any reasonable M$Windows version. Freeware and Open Source.

Embedded Windows CE VOIP Developers Kit is your embedded telephony solution for developing software/hardware to enable device to device phone calls using SIP/H323 phone right at the electronics level. The design is based on Hybrid Combination of RFC SIP Protocol and Windows RTC and is today successfully ported to Windows CE/Pocket PC/Smart Phone.

Requirements:

· Pocket PC/2002/Windows Mobile 2003

Konference is (or better: will be) a video-conferencing application for KDE. Since the rewrite (2005/01/25) it supports SIP as the signalling protocol. No longer H323 folks.

Whats New in This Release:

· Added support for transmitting an image instead of video, added support for Arts, fixed various segfaults and reworked configure-dialogs a bit

Orasi IVR description
Orasi IVR is a software based IVR utilizing Host Server Processors and resources Orasi IP IVR is a software application based IVR. It utilizing Host Server Processors and resources.

It doesnt need separate DSP as found in Telephony boards. Orasi IP IVR Employing mature H323 as voip protocol will give much compatibility between other hardwares and hardware independent.

Orasi IVR is very suitable as softswitch add on like GnuGK Gatekeeper.


Here are some key features of "Orasi IVR":

· Use H323 protocol with codec G.723.1, GSM, G711u/a
· Easy scriptable language for custom IVR flow
· Voicemail (Wav to MP3 conversion software will be provided)
· Database suppport using ODBC
· Gatekeeper registration.
· Transfer call
· Low host server resources

Support USB phone
PC to PC, PC to Phone, Phone to PC
H323 compliant
Touch-tones [DTMF]
Dial/ Redial/Hangup
Dynamic CODEC Selection
Caller ID
Call Timer
Backspace/Clear/Delete
Microphone & Speakers Levels
Last Caller-IDs
Recent Calls Dialed
Recent Calls Received
Sound Device Selection
New Familiar-looking Menu
Phonebook
Speed Dial
Benefits:
Easy to install
Intuitive user interface & Menu.
Easy to navigate and configure.
G.723/729/GSM codecs included
NAT/Firewall support
Multiple gatekeeper selection
Customised skin can be provided by Author