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

Download

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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

Download

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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.

Download

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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

Download:

Donwload Site1

CounterPath’s eyeBeam 1.5 a multimedia communicator designed to enhance the user’s communications experience in Voice over IP. Whether you are a consumer or Small Office/Home Office (SOHO) user, eyeBeam allows you to stay connected and manage your calls and availability with a comprehensive suite of carrier-grade Voice, Video, IM and Presence features.

Whether you are having a simple voice conversation or hosting a conference call with multiple collaborative applications, eyeBeam offers users the ability to personalize your softphone and manage your availability.

Features

eyeBeam 1.5 supports voice, video conferencing, instant messaging and includes several features that enhance the overall user experience. It also includes network management, platform support and security capabilities.

• Open Standards based next generation telephony client
• Session Initiated Protocol (SIP) based signaling for all interactive media sessions
• Enhanced Quality of Service (QoS) for voice & video calls
• Security offering signaling & media encryption via TLS and SRTP streams
• Performance Management of the SIP end-point
• High Compression codec support
• Comprehensive Personal Address Book, including detailed calls lists and history
• Zero-Touch Configuration of your audio or video devices
• Microsoft Outlook® 2003 integration allowing users to import their address book into their eyeBeam contact list and dial directly from the application
• IM & Presence Management
• Multi-party and ad hoc Voice and Video Conferencing [IP & PSTN]
• Voice & Video Call Recording
• Detachable & Sliding Drawers for quick Video and Contact information
• Toast pop-ups allowing the management of incoming calls

Standard Features

• Six lines
• Call display and Message Waiting Indicator
• Speakerphone
• Mute
• Redial
• Hold
• Do not disturb
• Call ignore
• Call history – list of received, missed, dialed
• Call forward
• Call transfer
• Voice/video call recording
• Voice/video conferencing

Price: $34

via CounterPath

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

For more info click here!

Incoming search terms for the article:

• voip softphone free

Make phone calls using your PC, Mac or Pocket PC

* Installs and ready to make calls in just minutes
* Use with Any VoIP SIP Provider for Free or low cost phone calls
* Many great features for the business or power VoIP user

Softphone Features

* Make voice and video calls free between computers
* Supports PC to phone via a VoIP SIP gateway provider
More information on VoIP
* Place callers on hold
* Basic on-hold music included
* Supports caller ID display and logging
* Includes a phone book with quick dial configuration
* Integrates with Microsoft Address Book.
* With the PocketPC version you can take your VoIP numbers wherever you go
* Data compression, echo cancellation, noise reduction and comfort noise
* Supports emergency numbers calling (e.g. 911, 999, 000)

Features Commercial Version Only

* Configure up to 6 phone lines
* Call transfer
* Phone call recording
* Do Not Disturb
* Push to talk intercom
* Call conferencing for up to 6 people
* Use one license both your computer and Pocket PC

Download Express Talk!

OzVoIP is a leader in the deployment of Internet business telephone systems in Australia. We pride ourselves in being an informed source of clear and competent advice to our customers and then being able to customise solutions according to their needs.

OzVoIP Business PBX Solutions

Some PBX systems are clearly made for low-end needs. Others are powerful business tools that will increase productivity, improve customer satisfaction, enhance the company’s image and keep long term costs under control.

If you are truly serious about your business, you should look for the following things in your PBX system:

Minimum Feature Set

• ACD queuing
• Dynamic agent presence
• Distributed call routing
• Follow-me calling
• Centralized voice mail
• Transfers between extensions
• Simplified web-based system management
• Auto-attendant with custom greetings
• Company directory accessible by first or last name
• Selectable music or custom information on hold
• Scalability
• Integrated conferencing
• Capacity
• Reliability

Features Important for Some Situations

• Skills-based call routing
• “Never-full” voicemail
• Voice and fax mail notifications
• Web and email delivery of voice messages
• Real-time system and queue monitoring
• Detailed system and queue logs
• Multiple number, multiple ID system entry
• DID – Direct Inward Dial
• Geographic call distribution

via OzVoIP

Call Union is the internet telephone calling service that lets you call from anywhere to anywhere at a fraction of what the telephone company charges. Using VoIP (Voice over IP) your telephone call is carried over broadband internet instead of the traditional telephone lines.

Free Internet Call

With Call Union you can call any other Call Union internet phone, where ever it may be for free, just dial their 10 digit Call Union number that begins with 510 to make a free internet phone call.

Features

Free PC-Phone

Download the PC phone and start to make calls immediately.

Free internet phone calls

Internet Calls are always free. Call domestically. Call internationally. It doesn’t matter. If the person you are calling has a Call Union internet phone you can talk as long as you like for free.

Caller ID

Internet Calls can include / not include your name and number, with Caller ID & Caller ID Block

Free number for incoming calls

Request your free UK 0845 number so that anyone can call your Call Union phone

Use your own phone number

If you purchase “Starter Pack 3″, which includes a Sipura 3000, incoming calls can be transferred to your Call Union number, so that callers can be forwarded to another number or to voice mail

Call Waiting

Call Waiting allows you to accept an incoming call while you’re already on a phone call.

Call Hold

Call Holding allows you to hold an incoming call while you’re already on a phone call.

Call Transfer

You can transfer a call to another Call Union user

Call Forwarding

Send incoming calls to any phone number you choose and then to Voice Mail.

Call Return

Call return and Call Back feature, enables you to call the last number that called you, or the last number that you called by dialling *69 or *66

Do not disturb

Need some peace, dial *78 to activate and *79 to deactivate DnD on your Sipura

Voice Mail

If you can’t be reached by Call Forwarding, the caller is prompted to leave a message, which can be emailed to you or you can listen to online as well as from your Call Union phone.

3-Way Calling

With 3-Way Calling, you can connect with two other parties at the same time

Web Control Panel

With Call Union Online Account Management, you’re in complete control of your account. You can access your account 24/7 to check call activity, view and change your billing information, manage voicemail, and edit your user information. And because it’s available real-time, you have the most accurate and up-to-date information every time you login.

Itemised Billing

Call Union Billing gives you complete control with instant, secure access to your billing information. You can access your account 24/7 to check your invoices, review your complete call history, and change your credit card information. And because it’s available real-time, you have the most accurate and up-to-date information at all times.

International Rates

AUS and NZ: UK 1.4p/min.
China: US 2.9¢/min
France / Germany: UK 1.4p/min.
Moscow / St. Petersburg: US 2.3¢/min
South Africa: UK 4.7p/min.
United Kingdom: UK 1.4p/min.
United States: US 2.3¢/min

via Call Union

iCallGlobe is one of the leading VoIP (Voice over Internet Protocol) Service Provider specialising in Internet Telephony Solutions for businesses, VoIP reseller programs and VoIP carrier services across varied areas of expertise. iCallGlobe provides a cluster of profitable VoIP solutions, which include world class quality VoIP Call Termination at cost-effective rates. Our vision is to emerge as a major VoIP Service Provider by integrating telephony system with the contemporary online world.

iCallGlobe Limited has been a prominent VoIP service provider since 2004. Ever since our inception, we have charted a significant growth in developing customer base and carrier-interconnections with our successful VoIP business solutions.

• Presently handling more than 75 million minutes and lateral partnerships with over 30 Tier-1/ILD operators.
• Achieved a booming 500% growth in terms of revenue in 2006.
• Equipped with state-of-the-art infrastructure at multiple strategic locations- U.S,Singapore and UK.
• Maximum ACD, ASR and minimum PDD ensures excellent voice quality to power tune your communication structure.
• Efficient, 24/7 technical support 365 days a year through online chat and email guarantees to deliver within 90 minutes turn around the fastest resolution to your queries

VOIP Services

• Carrier VoIP services: User can get benefit from the collaborative agreements with over 40 global network providers. We are a carrier’s carrier driving economic growth through a range of value-added services conforming to the highest quality standards. We offer A-Z call termination at industry leading costs.
• Reseller VoIP services: Business users wanting to enter the lucrative domain of VoIP, but not having technical expertise for the same, can depend on our reselling services. The investment requirements are minimal. With options for brand building and round-the-clock technical support, our reseller services are among the best in the industry.
• Contact center solutions: The customer relationship management services are just right to enhance the efficiency, productivity and profitability of call centers and contact centers.
• Corporate business solutions: Businesses, having a global presence, can manage their activities more efficiently using these solutions from our kitty. The hosted VoIP solutions are just perfect to connect the different corporate offices into a cohesive whole. With seamless VoIP call termination and the freedom to work from virtual offices, the corporate business solutions are proving their worth in more ways than one.
• Managed partitioning services: These on-demand technical solutions come with dedicated switches, using which our customers can offer VoIP enabled services to end-users. We help our customers to create their own brand identities and develop fulfilling relationships with clients.
• Solution for ISPs: ISPs can make the most of our customized solutions to sell voice services to end-users. With equipment specifications, technical support and round-the-clock monitoring, these solutions are proving to be quite valuable in different contexts.

VOIP PBX

• DID management: Get DID numbers and carry out your business transactions and internal communications all over the world, sitting in one place.
• Customised call reporting: Get detailed reports of all phone calls of your employees and clients, with date, time and call durations.
• Cost-effective calling: Call all over the world at cost-effective rates with our virtual PBX solution and save big on telephone bills. We also offer an array of additional features like call transferring and forwarding.
• Voice mail, transferable on your e-mail ID
• Conference features: Manage conferences, set conference timings and number of participants, send customised e-mail invites and check records of previous and future conferences. In short, your one-stop solution for organising efficient tele-conferences.
• Wireless connectivity: Do away with tangled wires and complicated controls. iCallGlobe brings you a much simpler and wireless means to communicate, both internally and externally.

via iCallGlobe