- Support platforms: Windows
2000/XP/Vista/7, Windows Mobile 5/6/6.5.
- Webphone: Support IE and Firefox,
Chrome.
- Support servers: Cisco CallManager,
OpenSER, Kamailio, OpenSIPS, Asterisk,
Radvision, Nortel, Avaya and other SIP
Platforms.
- Support development tools: MS Visual
Basic, MS Visual C#, MS Visual C++,
CodeGear Delphi, Javascript/HTML
- Audio call: G.711 aLaw/uLaw,
G.722.1, G.722, SPEEX, SPEEX-WB, AMR, AMR-WB,
GSM, iLBC, G723.1, G.729.
- Video call: H.263, H.263-1998,
H.264.
- Audio record: record audio as wav
and ogg file.
- Video record: record video as AVI
file.
- Support SVGA, XVGA, VGA, QVGA, CIF, QCIF,
720P video resolution.
- Support play AVI file to remote
side.
- Support play wave file to remote
side.
- Allows send PCM data to remote
instead of microphone input.
- Acoustic Echo Cancellation
- Automatic gain control
- Comfort Noise Generation
- Voice Activity Detector
- Call transfer: Attended transfer,
Blind transfer.
- Call forwarding
- Call hold, mute speaker, mute
microphone.
- Do not disturb(DND), Auto
answer(AA).
- Audio conferencing, support more
than 100 parties audio conferencing.
- Video conferencing, support more
than 100 parties video conferencing.
- Support P2P call without SIP proxy
server.
- STUN support
- IPv6 support
- Outbound proxy server support
- Support QoS.
- 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 adding custom SIP header.
- Support modify SIP header.
- Support send INFO and OPTIONS
message.
- IM Support: SIMPLE(Presence,
Subscribe, Pager message) and XMPP.
- Message waiting Indicator(MWI)
- DTMF support: Send DTMF tone(RFC2833
and SIP INFO method), detect DTMF
tone(RFC2833 and SIP INFO method).
- Multiple Call
- Audio Tuning Wizard
- Video Tuning Wizard
- Microphone & Speaker Device Selector
- Microphone & Speaker Volume control
- VPN integration.
- Important contacts from Outlook.
- Skins.
- Call logs.
- Address book.
- Call Timer.
- Speed dial, redial.
- Auto update.
- Credit Balance Display.
- Make installation packet.
- Auto provisioning.
- DNS SRV.
- GUI customization.
- Yealink USB Phone support.
- Multiple language support.
- More customize requirements as
customer request.
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Resource Center |
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PortSIP Products License Agreement
SDK Data Sheet
Technical Tip
Question –
How to get the best voice quality?
Answer – 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, G.722.1 codec on
broadband, on dialup try the G.729, iLBC, or other codecs;
For more information, read
the FAQ.
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FEATURES