AGEphone Desktop Support

AGEphone is a softphone based on the international SIP standard. The AGEphone microSIP stack, an original SIP parser as well as an IP packet processing program make operation at super small size and with super light weight possible.

AGEphone is designed to support a large number of IP telephone services. The software offers great sound quality through a newly developed packet buffering which makes high quality audio phone calls come true.

Moreover, by supporting UPnP as well as STUN, AGEphone is able to provide NAT traversal for a large number of routers. AGEphone works very well together with other applications: address books from Outlook can be imported easily, and the Click to Call function makes it possible for you to start dialing a phone call by simply clicking on a callto: tab linked on the internet or in an Email.

AGEphone provides various skins, so you can adjust its appearance to your liking.

 

Based on international standards:

RFC3261: SIP: Session Initiation Protocol
RFC2327: SDP: Session Description Protocol
RFC1889: RTP: A Transport Protocol for Real-Time Applications
RFC2833: RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC3489: STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
RFC3581: An Extension to the SIP for Symmetric Response Routing
UPnP :Universal Plug and Play 

Key Functions

High quality sound, low delay buffering	The world's smallest and lightest VoIP stack
UPnP, STUN Support	Possibility to simultaneously use multiple services
Answering Machine	Click to Call Function
DND (Do not Disturb)	VAD (Voice Activity Detection)
Call History	Number Auto Complete
Address Book	Outlook/Outlook Express Support
Speed Dial Function	Inband/RFC2833 DTMF send
Call Melodies	Calling Rules
Multiple Codecs	Voice Recording
Symmetric Support (RFC3581)	Serverless Mode included

 

Please feel free to try AGEphone without registering for a serial number. AGEphone will then work in trial mode. Please keep in mind that while in trial mode, phone calls are limited to 60 seconds.

Version History

Ver.1.83 -> Ver.1.85 [10/02/2008]
* SIP-protocol bugfixes
* Backup server option added

Ver.1.82 -> Ver.1.83 [08/04/2008]
* No ringtone on call fix

Ver.1.75 -> Ver.1.80 [07/02/2008]
* microSIP stack update
* AIR-GUI: Supports transparency effects
* Paging function extension: Auto-answer over call-info header
 
Ver.1.74 -> Ver.1.75 [04/04/2008]
* Management of more than 10 audio devices possible.
* Basic QoS Options added
* extended handset support

Ver.1.65 -> Ver.1.66 [06/25/2007]
* Real VGA (WVGA) support (AGEphone WM5)
* Real Keyboard support (AGEphone WM5)
* Improved audio buffer.

Ver.1.64 -> Ver.1.65 [05/25/2007]
* Improved Address ook Filter
* Session Timer Handling improved
* fixed several smaller problems

Ver.1.63 -> Ver.1.64 [04/13/2007]
* popup on incoming call improved
* USB handset support improved
* fixed a problem with Session Timer (UAC)
* added standby mode (WM 5)
* added automatic dialup when disconnected (WM 5)

Ver.1.62 -> Ver.1.63 [03/28/2007]
* Fixed security vulnerabilities when receiving malformed SIP packets

Ver.1.61 -> Ver.1.62 [03/16/2007]
* Fixed security vulnerabilities when receiving malformed SIP packets
* Fixed a problem where calls could not be answered with the enter key
* Fixed a problem when adding contacts from the call history to the address book

Ver.1.60 -> Ver.1.61 [03/10/2007]
* Improved Address Book
* Different USB handsets can now be selected in the setup menu
* Fixed Direct Sound input issue on Vista
* IPv6 global addresses can now be selected in the interface menu

Ver.1.56 -> Ver.1.60 [02/28/2007]
* DirectSound Support
* IPv6 Support
* Improved Dial Plan Settings for International Calls
* Local Music on Hold
* New, Improved Address Book

Ver.1.55 -> Ver.1.56 [12/15/2006]
* Registration to servers improved
* Fixed a problem with INVITE cancel

Ver.1.54 -> Ver.1.55 [11/24/2006]
* Fixed a problem with multiple users on one computer
* Audio device control improved

Ver.1.53 -> Ver.1.54 [11/10/2006]
* Windows Vista support
* Audio device control improved

Ver.1.52 -> Ver.1.53 [11/05/2006]
* Improved Voice Quality
* Fixed a problem with call forwarding

Ver.1.51 -> Ver.1.52 [10/17/2006]
* Improved Voice Quality during recording
* Fixed a problem with the settins screen

Ver.1.50 -> Ver.1.51 [10/04/2006]
* Fixed a problem with the address book
* Fixed a problem with call forwarding

Ver.1.42 -> Ver.1.50 [09/29/2006]
* Added Call Forwarding and "Provider Crossover" Function
* Improved Jitter Buffer and Voice Quality
* Improved UPnP Connectivity

Ver.1.41 -> Ver.1.42 [08/18/2006]
* Added function to automatically execute programs on call connection / disconnection
* Improved SIP stack
* Fixed an issue that occured when entering the serial number in the PDA version

Ver.1.40 -> Ver.1.41 [08/11/2006]
* Improved Outlook support
(a popup is shown when an Outlook contact calls)
* Enabled input of IP-Adresses for P2P-connections with standard phone keys
* Fixed an issue that occured when entering the serial number
* Fixed an issue concerning P2P-connections

Ver.1.38 -> Ver.1.40 [07/25/2006]
* corrected a security issue with possible buffer overflow
* improved voice quality (ptime = 10)
* faster startup
* Multilanguage support improved

Ver.1.37 -> Ver.1.38 [07/05/2006]
* Automatic selection in case of multiple network interfaces
* Multilanguage support improved
* Volume control improved

Ver.1.36 -> Ver.1.37 [06/14/2006]
* added German language interface
* UPnP connectivity improved
* corrected some other problems

Ver.1.35 -> Ver.1.36 [05/02/2006]
* improved NAT traversal algorithm
* improved SIP stack
* corrected some other problems

Ver1.34 -> Ver1.35 [03/20/2006]
* improved UPnP stack
* corrected some other problems

Ver1.33 -> Ver1.34 [03/03/2006]
* improved NAT traversal algorithm
* recording of comlpete communication is now possible

Ver1.32 -> Ver1.33 [02/06/2006]
* increased support of IP telephone servers for use within extension numbers
* supports calls from an increased number of network environments

Ver1.31 -> Ver1.32 [01/17/2006]
* increased number of supported IP Telephone Services
* improved NAT Traversal Engine

Ver1.30 -> Ver1.31 [12/28/2005]
* increased number of UPnP supporting routers
* when shutting down, program now waits until communication is completely closed
* corrected problem that in a specific environment dialing could not be canceled

Ver1.24 -> Ver1.30 [12/22/2005] (Major Version Upgrade)
* improved the SIP stack
* native implementation of UPnP
* added calling rule function (call melodies)

Ver1.23 -> Ver1.24 [10/20/2005]
* added support for serverless calls
* added Click to Reply, Click to Disconnect Function
* corrected some problems

Ver1.22 -> Ver1.23 [09/28/2005]
* altered answering machine mode so that it remembers ready status on shutdown
* corrected problem that with some services an error occurred when calling
* corrected some other problems

Ver1.21 -> Ver1.22 [09/01/2005]
* added RFC3581 Function
* added NAT Mode

Ver1.20 -> Ver1.21 [08/25/2005]
* improved UPnP connectivity

Ver1.13 -> Ver1.20 [08/03/2005]
* added Skin Switching Function
* improved the SIP stack

Ver1.12 -> Ver1.13 [07/26/2005]
* improved GUI drawing management
* added audio visualizer
* added function for backing up settings

Ver1.11 -> Ver1.12 [07/07/2005]
* improved the SIP stack

Ver1.10 -> Ver1.11 [06/15/2005]
* English support
* official support of USB Internet Phone (Model:IP-001)
* improved address book function
* improved quality of RFC2833

Ver1.04 -> Ver1.10 [05/24/2005]
* added vibration mode
* improved connectivity to IP telephone service providers
* improved NAT traversal algorithm
* improved DTMF send quality

Ver1.03 -> Ver1.04 [04/26/2005]
* made communication with AGEphone (PDA version) possible in internal mode (P2P mode)
* improved NAT traversal algorithm

Ver1.02 -> Ver1.03 [03/24/2005]
* improved DTMF send method
* now displays status of NAT traversal and register status of each service
* corrected problem of one way only communication with some audio devices

Ver1.01 -> Ver1.02 [03/14/2005]
* voice of communication partner can now be recorded
* enabled communication up from a 29.2kbps band
* corrected problem of one way only communication with some audio devices

Ver1.00 -> Ver1.01 [03/03/2005]
* support of call progress tone
* corrected problem that audio device could not close normally
* added function to edit phone numbers entered in phone book

System Requirements

OS	Windows 2000 / Windows XP / Windows Vista
CPU	Pentium Processor or equivalent 133Mhz or above
Memory	64MB or above
HDD	10MB or above
Connection	Up- and Downstream of each 80kbps or above
Other	Speakers and microphone, or headset LAN-port (100BASE-TX): 1 port Display: VGA or better, high color support In case of a NAT environment, a router with UPnP support is recommended (If the NAT environment is symmetric, it cannot be used)