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Mobile Testing with Open Source VoIP

Open Source Softphones and Testing 

Open Source VoIP softphones are most needed when we want to start doing some test on our system, to get the real user experience. not all softphones are good for testing purposes but let's start examining those tools that are free. Please comment and suggest your own/preferred testing tool for our mobile/voip user experience test. Softphones are good for manual and automatic testing. Some even give an API for better testing and operation reliability. The one I selected are all top quality open source softphones, which give you the opportunity to set it and customize it according to your needs, please if doing so, dont forget to share knowledge and abilities. Join the programming force.

 

 

Ekiga (GnomeMeeting)

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).
Main Features of the Ekiga Softphone Version 3.2 in a nutshell

  • 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.

http://www.ekiga.org/   , License: GPL

 

 

Twinkle

 

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.
 

 

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 Indication, 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)

Twinkle is available for Linux only
http://www.xs4all.nl/~mfnboer/twinkle/, License: GPL

 

 

Kiax

Kiax started in early 2004 as a small program mainly aimed to provide a simple user interface for making VoIP calls with Asterisk PBX (an open source VoIP PBX). Its first versions showed the existing need for a user-friendly, free and open softphone. Currently Kiax has been downloaded by more than 60 000 users (stats from SourceForge.net) and is available for direct installation from the repositories of the major Linux distros (Ubuntu, SuSE). While it is functionally rich and considerably stable its development has reached a state where modification and customization became difficult. With the help of Forschung-Direkt and MIXvoip Kiax development has been restarted. Kiax ver.2 is a complete re-write of the softphone which aims to clean up the design issues and to provide a more flexible architecture for extention and customization.

Key features and characteristics of Kiax ver.2:

  • Decoupled Signaling, Storage and Visualization aspects
  • Modularized, lightweight core layer
  • GCC4 ready code
  • Single codebase for Linux, Windows and MacOS
  • SQLite as default storage backend
  • QT4.4 as GUI frontend
  • Webkit integration
  • Even simpler (than old Kiax) to use UI
  • Completely brandable
  • Remote configuration
  • Simplified integration with service providers (via JSON)
  • Support for multiple service providers
  • Support for simultaneous calls
  • Registry fail-over support
  • Live CDR and Contacts search
  • Codecs: G711, iLBC, GSM, Speex
  • Noise reduction filter
  • I18n support

http://kiax.sourceforge.net, License: LGPL-licensed core, GPL GUI

 

QuteCom

QuteCom is the new name for the open source softphone previously known as WengoPhone. QuteCom began life as OpenWengo developed by French VoIP provider Wengo as a free softphone for its telephony service.
QuteCom is cross-platform (Windows, Linux, Mac OS X) and integrates voice and video calls and instant messaging. The number of protocols supported is on the same level as other multi-protocol IM clients. The application is developed with the Qt cross-platform toolkit.

http://www.qutecom.org/ , License: GPL

SIPhone

Home of the World's first free SIP/VoIP application for iPhone and iPod Touch 1 and 2.
Siphon SIP/VoIP project is the first in his category that works on iPhone and iPod Touch 2 with headset for all SIP providers. It is a native application approved running on 2.X using internal micro/speaker and headset.
The Application supports the SIP standard, preserving compatibility with hundreds of SIP providers and offers a GUI which preserves the apple design of native iPhone applications.
Be careful, this version didn't test on iPod Touch 1. One thing is sure, Touchmod's micro doesn't work with iPhone 2.X OS. You need a microphone with a HW key inside “approved” by Apple. Such a microphone are: iVoice III from Macally.
Currently, Siphon is localized in 15 languages.

http://code.google.com/p/siphon/ , License: GPLv2

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SIP Communicator

SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular VoIP and instant messaging protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, Bonjour, IRC and a whole lot of other useful features.
SIP Communicator is completely Open Source / Free Software, and is freely available under the terms of the GNU Lesser General Public License.
These are SIP Communicator main features:

  • Audio calls
  • Video calls
  • Desktop streaming
  • Desktop sharing
  • Audio conference calls
  • Audio level display
  • Call recording
  • Attended transfer
  • Blind transfer
  • Call encryption with SRTP and ZRTP
  • Mute
  • Hold
  • Support for ICE
  • Wideband audio
  • Noise suppression
  • Echo cancellation

SIP Communicator is cross-platform and is developed in Java.
http://sip-communicator.org, License: GPL

 

 

SFLphone

SFLphone is a SIP and IAX2 (Asterisk) compatible softphone for Linux developed by Canadian Linux consulting company Savoir-Faire Linux.
The SFLphone project's goal is to create a “robust enterprise-class desktop phone” and is designed to cater for home users as well as the “hundred-calls-a-day receptionist”.
Its main features include support of unlimited number of calls, multi-accounts, call transfer and hold. Call recording is another useful feature.
SFLphone has clients for GNOME (integrated options), KDE and Python and it now supports the PulseAudio sound server, so users can experience additional functionality like sound mixing and per-application volume control.
The softphone is designed to connect to the Asterisk open source PABX.

URL: http://www.sflphone.org/ , License: GPL3

 

 

Empathy

Empathy is a messaging program which supports text, voice, and video chat and file transfers over many different protocols. You can tell it about your accounts on all those services and do all your chatting within one application.
Empathy uses Telepathy for protocol support and has a user interface based on Gossip. Empathy is the default chat client in current versions of GNOME, making it easier for other GNOME applications to integrate collaboration functionality using Telepathy.
Current features:

  • Multi-protocol: Google Talk (Jabber/XMPP), MSN, IRC, Salut, AIM, Facebook, Yahoo!, Gadu Gadu, Groupwise, ICQ and QQ. (Supported protocols depend on installed Telepathy Connection Manager components.) Supports all protocols supported by Pidgin.
  • File transfer for XMPP, and local networks.
  • Voice and video call using SIP, XMPP and Google Talk.
  • Some IRC support.
  • Conversation theming (see list of supported Adium themes).
  • Sharing and viewing location information.
  • Private and group chat (with smileys and spell checking).
  • Empathy Conversation window
  • Conversation logging.
  • Automatic away and extended away presence.
  • Automatic reconnection using Network Manager.
  • Python bindings for libempathy and libempathy-gtk
  • Support for collaborative applications (“tubes”).

URL: http://live.gnome.org/Empathy, License: GPL

 

 

miniSIP

Minisip is a SIP User Agent ("Internet telephone").
It can be used to make phone calls, instant message and videocalls to your buddies connected to the same SIP network.
Features:

  • SIP compliant (RFC 3261 and more)
  • Multiple lines (users) on the same phone
  • Multiple incoming/outgoing calls simultaneously
  • Runs on multiple Operating Systems (Linux PC, Linux familiar IPAQ PDA, Windows XP and soon Windows Mobile 2003 SE)
  • Focus on security: TLS, end-to-end security, SRTP, MIKEY (DH, PSK, PKE)
  • Instant Messaging
  • Video conferencing
  • Spatial audio
  • Push-to-Talk (P2T)
  • Full Mesh audio conferencing
  • STUN support
  • Call Logging

URL: http://www.minisip.org/ , License: GPL

 

 

Linphone

Linphone is an internet phone or Voice Over IP phone (VoIP).
Features:

  • With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging.
  • Linphone makes use of the SIP protocol, an open standart for internet telephony. You can use Linphone with your favorite SIP VoIP operator.
  • Linphone is free-software (or open-source), you can download and redistribute it freely.
  • Linphone is available for PCs (linux, windows), MacOSX and for mobile phones: Android, iPhone.

URL: http://www.linphone.org/ , License: GPL

 
 

 

 

 


 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

 

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