FVN Softphone - portable SIP softphone for Windows OS.
It is designed to be used in an office with the existing FVN Office Phone system (or any SIP server) as an extension or replacement for your desktop phone.
![pc1](https://fvn-alliance.com/images/sp/pic1.png)
Softphone can have your main extension (if used without desktop phone) like 204 or separate like 224 with 204 desktop phone.
Incoming calls to 204 will ring both : on your desktop phone and Softphone. You can pick up anyone.
We recommend to use Logitech USB headset with our Softphone.
![pic2](https://fvn-alliance.com/images/sp/pic2.png)
FVN Softphone can be installed on any version of Windows and download from our website by next link:
fvn-alliance.com/prg/FVNSP3.zipSoftphone configuration:
![https://fvn-alliance.com/images/sp/sh1.png](https://fvn-alliance.com/images/sp/sh1.png) | Open menu on the right-top corner of Softphone.
And select ADD ACCOUNT |
![https://fvn-alliance.com/images/sp/sh2.png](https://fvn-alliance.com/images/sp/sh2.png) | Correct information in the fields according to your credentials.
Account name better to have - OFFICE
SIP Server, SIP Proxy and Domain - need to be your Phone System
IP
Display Name - can be yours (example: Manager Ivan).
Username and Login - your extension number (example: 224)
Password (example: ext224)
Voicemail Number: *98
Transport: UDP
And check mark - Publish Presence
Then click on SAVE
|
Setting normally you do not
need to change
| ![https://fvn-alliance.com/images/sp/sh3.png](https://fvn-alliance.com/images/sp/sh3.png)
|
When you making a call - new
window will be up with ability
to:
- Transfer call
- Make a conference
- End Call
| ![https://fvn-alliance.com/images/sp/sh4.png](https://fvn-alliance.com/images/sp/sh4.png)
|
Incoming call you will see
- Name
- Phone Number
With ability to Decline call or
answer.
| ![https://fvn-alliance.com/images/sp/sh5.png](https://fvn-alliance.com/images/sp/sh5.png)
|
Softphone use
Softphone always start minimized in tray. Right click on tray icon lets you access account and application settings.
Used to enter numbers with a mouse or sending DTMF signals. Also you can enter numbers with a physical keyboard, in this case you can enter letters, specify custom domain and port. Examples:
13455674657, buddy, buddy@sip.com, buddy@sip.com:5043, buddy@192.168.1.43, sip:192.168.1.55, etc.
"Presence subscription" allows you to use BLF functionality - pickup incoming calls of other users.
VideoSupported H.264 and H.263+ (other name H.263-1998) video codecs. Default codec - H.264, video
format - 640x480 @ 30 fps, outgoing bitrate 512 kbit/s. H.264 encoding requires significant CPU
resources. Recommended dual core processor, multimedia extensions like MMX will be used if is present.
Video capture and video rendering uses DirectX and Direct3D (with hardware acceleration).
Because hardware acceleration is used, video calls will not work with remote desktop session (RDP).
If you have serious problems with performance:
- update video adapter drivers
- install/reinstall DirectX
Frequently asked questionsQ: I launch Softphone but nothing happens.
A: Check for the Softphone icon in the system tray.
Q: How to setup account?
A: Right click on the Softphone icon in the system tray (near clock:).
Q: How to add contact?
A: Right click on blank white area in Contacts tab.
Q: How to achieve the best voice quality?
A: Voice quality depends on the audio codec that was selected in negotiation for the current call session.
In extended mode Softphone will show you what codec was selected for the session.
We recommended using : G.711@8kHz (PCMU), G.722@16kHz and G.729@8kHz in this sequence.
Sound latency is caused by a set of dynamic buffers on the path of audio. Average value - 200 ms (one way). There is no way to reduce latency significantly.
FVN Softphone Specifications:● small footprint (>2.5MB) and RAM usage (>5MB) - written in C and C++ with minimal possible system resources usage
● usability - user friendly in daily usage
● functionality - voice; video H.264 and H.263+, VP8; SIMPLE messaging (RFC 3428) and presence (RFC 3903, 6665); DTMF In-band, RCF2833, SIP-INFO.
● compatibility - conform to SIP standards
● voice quality - supports best voice codecs: Opus@24kHz, G.711 A-law (PCMA), G.711 u-law (PCMU), G.722@16kHz, G.723@8kHz, G.729@8kHz, iLBC@8kHz, GSM@8kHz, AMR@8kHz, AMR-WB@16kHz, Speex@8,16,32kHz, SILK@8,12,16,24kHz and Linear PCM@44kHz, including stereo.
● WebRTC echo cancellation algorithm and voice activity detection
● privacy - configurable encryption TLS / SRTP for control and media
● portability - has no additional dependencies and stores setting in ini file
● multilanguage and RTL support,