WINDOWS FORMS SOFTPHONE IN .NET v9.2.0

WINDOWS FORMS SOFTPHONE IN .NET
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Last Update:2012-02-02 - Minor Update: New feature: Webphone development support (Flash and Silverligt)
Requirements:PC, .NET compatible Development Kit, minimum .Net 3.5. SP1, SIP account (or a SIP PBX)
Type:Shareware
Limitations:20 Trial
Category:Communications::Telephony
Language:English


OS



Ozeki .Net VoIP SIP SDK for developing Windows Forms softphone to make/receive multiple VoIP phone calls. The Ozeki Windows Forms softphone sample has basic telephony functions, handles DTMF signals and customizable with further effective functions & company logo or brand name.

It delivers crystal clear sound due to the extended codec support. It has all the functions for establishing phone calls like make/receive call, sending/receiving DTMF signals to navigate in IVR systems and display call events on GUI. Ozeki VoIP SIP SDK has been developed to ensure excellent sound quality.

For this purpose various codecs are supported: G.711 aLaw/uLaw, G.729, iLBC, GSM, G.722, SPEEX, SPEEX-WB. To make/receive VoIP phone calls, the WPF softphone is compatible with almost all PBX systems like Asterisk PBX, Cisco UCM, 3CX Phone System, Kamailio PBX, etc.

Ozeki SIP SDK also works with almost all VoIP service providers (Skype Connect, Gafachi, Callcentric, SureVoIP, VoIPtalk, etc.) due to that it is fully standard compliant. Information can be gathered about the phone line status with the PBX like successful or unsuccessful registration, registered or not registered.

The source code can be downloaded freely to use it or modify it. Due to the flexibility of Ozeki SIP SDK, the developed softphone can be further improved with features like: Jitter buffer, Voice Activity Detection, Acoustic Echo Cancellation, Packet Lost Concealment (PLC), Call timer, Customized with Company Logo/Name, Unlimited User License, Full HD video, Multiple lines for multiple concurrent calls, Conferencing, Call history, Quick calling and re-dialing, etc.

). Tools: SIP protocol implementation, RTP implementation, RTCP implementation, Audio device support (mic, speaker), Audio encoding/decoding (audio codecs), Voice quality improvement tools (AEC, AVG, NR, Jitter buffer), Video device support (camera and playback), Video encoding/decoding (video codecs), Implemented network communication protocols: UDP, TCP  





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Simon Robert, info(at)voip-sip-sdk(dot)com, Phone: +36 52 532 731