Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet.

It supports HD sound quality and video up to DVD size and quality.

It is interoperable with many other standard compliant softwares, hardwares and service providers as it uses both the major telephony standards (SIP and H.323).

Features

• Ease of use with a modern Graphical User Interface.
• Audio and Video free calls through the internet.
• Free Instant Messaging through the internet with Presence support.
• Audio (and video) calls to landlines and cell phones with support to the cheapest service providers.
• High Definition Sound (wideband) and Video Quality up to DVD quality (high framerate, state of the art quality codec and frame size).
• Free of choice of the service provider.
• SMS to cell phones if the service provider supports it (like the default provider).
• Standard Telephony features support like Call Hold, Call Transfer, Call Forwarding, DTMF.
• Remote and Local Address Book support: Remote Address Book support with authentification using the standard LDAP technology, Local Address support in Gnome (Evolution).
• Multi platform: Windows and GNU/Linux
• Wide interoperability: Ekiga use the main deployed stantards for telephony protocols (SIP and H.323) and has been tested with a wide range of softphones, hardphones, PBX and service providers.

Call

• Call Hold: This effectively pauses Video & Audio transmission.
• Call Transfer: You can transfer the remote user to another user.
• Call Forwarding on No Answer, on Busy, on Always: This allows you to con?gure Ekiga to forward incoming calls to a speci?ed user.
• Message Waiting Indications Support
• Dialpad: This allow you to dial numbers using a graphical alphanumeric keypad.
• DTMFs Support: This feature is necessary when using services asking you to dial numbers.
• Calls History: This is a convenient way about all outgoing and incoming calls.
• Call Monitoring: Statistics about the network traffic caused by Ekiga are displayed in the status bar.
• Enum Support: Enum is a standard method to provide a unified numbering system between the public switched telephone network (PSTN) and various VoIP providers. It is officially deployed country wide.

Operating System

Linux, Windows

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Twinkle is a softphone for your voice over IP and instant messaging communcations using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages.

Features

In addition to making basic voice calls Twinkle provides you the following features:

* 2 call appearances (lines)
* Multiple active call identities
* Custom ring tones
* Call Waiting
* Call Hold
* 3-way conference calling
* Mute
* Call redirection on demand
* Call redirection unconditional
* Call redirection when busy
* Call redirection no answer
* Reject call redirection request
* Blind call transfer
* Call transfer with consultation (attended call transfer)
* Reject call transfer request
* Call reject
* Repeat last call
* Do not disturb
* Auto answer
* Message Waiting Inidication
* Voice mail speed dial
* User defineable scripts triggered on call events
E.g. to implement selective call reject or distinctive ringing
* RFC 2833 DTMF events
* Inband DTMF
* Out-of-band DTMF (SIP INFO)
* STUN support for NAT traversal
* Send NAT keep alive packets when using STUN
* NAT traversal through static provisioning
* Persistent TCP connections for NAT traversal
* Missed call indication
* History of call detail records for incoming, outgoing, successful and missed calls
* DNS SRV support
* Automatic failover to an alternate server if a server is unavailable
* Other programs can originate a SIP call via Twinkle, e.g. call from address book
* System tray icon
* System tray menu to quickly originate and answer calls while Twinkle stays hidden
* User defineable number conversion rules
* Simple address book
* Support for UDP and TCP as transport for SIP
* Presence
* Instant messaging
* Simple file transfer with instant message
* Instant message composition indication
* Command line interface (CLI)

VoIP security

* Secure voice communication by ZRTP/SRTP
* MD5 digest authentication support for all SIP requests
* AKAv1-MD5 digest authentication support for all SIP requests
* Identity hiding

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Zoiper Classic IAX & SIP multilanguage and multiplatform (Windows, Linux and Mac OS X) softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure.

Zoiper Classic BIZ provides secure high-quality voice calls and conference, fax sending and receiving functionality, and enhanced IP-calling features wrapped in a compact interface and small download size.

With Zoiper Classic FREE you can:

• make and receive voice calls with supreme quality
• send and receive faxes (watermarked)
• transfer calls
• hold calls
• carry Zoiper wherever you go on a USB stick, flash card etc. You can run Zoiper Classic FREE from the USB/flashcard without writing to the Windows registry.
• With Zoiper Classic FREE you can register one SIP and one IAX account and 2 lines.

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Features:
SIP industry standard support
Send DTMF as RFC2833
Proved interoperability with most ITSP, IP-PBXs, softphones and Gateways

G.729 **, G.723.1, iLBC, GSM full rate (06.10) and G.711 uLaw and G.711 ALaw codecs supported in software
Video codec supported are H.263 and H.261
Codec preference management

Acoustic Echo Canceller
VAD, CNG
Adaptive Jitter buffer

PC to PC call, PC to Phone call, and Phone to PC call
Multi Lines, Multi Profiles
Basic Telephony, Mute, Redial
Business Telephony, Forward, Transfer, Hold, Conference

Audio Video Wizard
Volume and Microphone gain control

USB handsets integration

Address Book, Call history, Call logs
Hot keys and Speed-dial

Fully attractive skinable interface
Private Branding and load your logo
Locked to a ITSP service profile/GUI
Localization

Call timer
Account Balance Display
Auto Login to your billing website page

DNS SRV support
NAT/Firewall traversal
HTTP Tunnel (coming soon)
SIP call signal and RTP media stream encryption, bypass VoIP blocking

ActiveX webdialer for web click-to-calls

PC Platforms:

MS Windows 98SE/ME/2K/XP/2003

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Donwload Site1

SJphone® is a VOIP softphone that allows you to speak with any other softphone running on a PC/ PDA, any stand-alone IP-phone, or using VOIP service provider with any traditional wired or mobile phone. It supports both SIP and H.323 standards and is fully inter-operable with most major VOIP vendors and Service Providers.

Having SJphone installed is not enough to talk except over your local network. You have to obtain an account with any ITSP like Vonage or Free World Dialup, or install your own VoIP server.

New features:
1. Audio Wizard enabling the user to fine tune Audio options
2. New Interface tab in Options enabling the user to select SJphone Interface language.
3. Improved interoperability for a number of SIP softswitch vendors
4. Protection against DoS attacks. The SJphone now consumes only limited resources, and in case of overload  SJphone answers with the “503 Service unavailable” message.

Requirement:
A wideband internet connection like DSL, 802.11a/b/g/n wireless, or an old good wired Ethernet or E1/T1.  Dial-up users are welcomed too, but a GSM or iLBC codec must be used.
-  You can use any today x86 PC running MS Windows ME/2K/XP, MAC OS, Linux.  Also you can use a PDA running PocketPC 2003; choose and download your version of SJphone from our website. Ask us about other technologies.
- Full-duplex sound is a must. To check it, run a recorder and player at the same time. If the both work properly, you have a full-duplex soundcard, that’s OK.

Services and profiles:
- You can make direct free calls to your party’s IP-address using “PC-to-PC” services.
- For internet calls and for calls to / from regular telephones, you may obtain a VOIP service provider. You can sign up to several different services and easily switch between them. Install a new service profile, initialize it with your login and password, and enjoy!
- Experienced users may build their own IP-telephony network; they can create new service profiles themselves.
- Our software works with with most ip-PBX, SIP-proxy and H.323 gatekeepers, and gateways just by adding a new service profile.
- How to use the services behind firewalls, see part 3.

How to get best quality:
- Most notebooks and PDAs have embedded microphone and speakers. To get better audio quality and to avoid echo you may use any USBphone (download an appropriate driver from our website), BlueTooth or a simple wired headset;
- We recommend you to install the latest updates of Operating System, sound and network drivers, and DirectX (check with dxdiag.exe, it should be 7.0 or higher). Get updates from the vendor website or just run MS Windows update.
– Please make sure that your microphone and speaker provide a good enough sound quality;
- You may have problems with old consumer PDAs that weren’t designed for VOIP. Most of latest devices have embedded wireless, good CPU and sound system now, and work perfectly;
- Check Internet connectivity: Try to ping your party’s IP address. A good response time is under 100 ms unless the party you are calling to is from down under.  If the ping time seems too large, you Internet connection is too busy now.  Try to stop any other Internet activity. If the problem persists, ask your Internet provider or system administrator to fix it.
- For the best voice quality use G.711 codec on broadbands, for dial-ups try GSM, iLBC, or other codes;
- Normally you should hear Voice over IP much better than using old plain or mobile telephony.  Sound delays, echo, voice outage, and distortions are network or hardware problems;

Download SJphone

SJphone for PC (stable):

SJphone for MS Windows XPsp2, v.1.65.377a, 20.03.07 (NEW!)
SJphone for MS Windows XPsp2/2000sp4, v.1.60.289a, 06.19.05
SJphone Linux, v.1.60.299, 09.24.05
SJphone MAC, v.1.60.299, 09.24.05

SJphone for WLAN-PDA with Windows CE(stable):
SJphone CE .cab for Pocket PC 2003 v.1.60.303c, 10.26.05
SJphone CE .exe for Pocket PC 2003, v.1.60.303c, 10.26.05
SJphone CE .cab for Pocket PC 2003SE, v.1.60.303c, 10.26.05
SJphone CE .exe for Pocket PC 2003SE, v.1.60.303c, 10.26.05

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