The PortSIP Pangolin Softphone is built on the PortSIP VoIP SDK, The PortSIP VoIP SDK features are all available for PortSIP Pangolin Softphone.
 
      PortSIP VOIP SDK Data Sheet
      PortSIP IVR SDK White Paper
 
 
Standard Features
 
Support platforms: Windows 2000/XP/Vista/7, Windows Mobile 5/6, Nokia S60 3rd FP2.
Support servers: Cisco CallManager, Kamailio, OpenSIPS, Asterisk, PortaOne, Radvision, Nortel, Rainbow, Avaya and other SIP Platforms.
Support development tools:
MS Visual Basic 2005/2008
MS C#
MS Visual C++(8.0/9.0)
CodeGear Delphi 2007
Javascript/HTML
Audio call: G.711 aLaw/uLaw, GSM, iLBC, G723.1, G729.
Video call: H263, H263-1998, H264.
Call transfer: Attended transfer, Blind transfer.
Call forwading
Call hold, mute speaker, mute microphone.
Do not disturb(DND), Auto answer(AA).
Audio record: record audio as wave file.
Video record: record video as AVI file.
Support SVGA, XVGA, VGA, CIF, QCIF video resolution.
Support TLS/SRTP(usually use to avoid SIP blocking)
Support access incoming aduio stream directly.
Support access incoming SIP message directly.
Support access incoming video stream directly.
Support play AVI file to remote side.
Support play wave file to remote side.
Support adding custom SIP header.
Support modify SIP header.
Audio conferencing, support maximum 8 parties audio conferencing.
Video conferencing, support maximum 3rd video conferencing.
Support send INFO and OPTIONS message.
Support P2P call without SIP proxy server
IM Support: SIMPLE(Presence, Subscribe, Pager message) and XMPP.
Message waiting Indicator(MWI)
Authentication: HTTP Basic, Digest Authentication.
DTMF support: Send DTMF tone(RFC2833 and SIP INFO method), detect DTMF tone(RFC2833 and SIP INFO method).
Multiple Call
P2P call without SIP server
Audio TuningWizard
Video Tuning Wizard
Microphone & Speaker Device Selector
Microphone & Speaker Volume control
Important contacts from Outlook.
Acoustic Echo Cancellation
Automatic gain control
Comfort Noise Generation
Voice Activity Detector
STUN support
Outbound proxy server support
Jitter buffer
Free product version upgrades: one year free upgrades.
 
Exclusive to Customization Softphone
 
Skins
Call logs
Address book
Call Timer
Speed dial, redial.
Auto update
Credit Balance Display
Make installation packet
Auto provisioning
DNS SRV
GUI customization
More customize requirements as customer request.

 

Latest news

Jan 25, 2010


PortSIP released the PortSIP VoIP SDK v5.5


Frequently Asked Questions

"Most of mobile computers have an embedded microphone and speaker. Make sure that your microphone is good enough by trying to record your speech. In order to get better audio quality and avoid echo, you may use headset, Blue Tooth, and wired headset etc; We strongly recommend you to install the latest version of operating system, sound and network drivers, and DirectX (check with dxdiag.exe, it should be 9.0 or higher). Check your Internet connectivity by pinging your party's IP address. A proper response time is below 100 ms unless the party you are calling is from Down Under. If the ping time seems too large, you Internet connection is too busy now. Try to stop any other activity. If the problem persists, ask your Internet provider or system administrator to fix it. For the best voice quality, use the G.711 codec on broadband, on dialup try the G729, iLBC, or other codecs."