VoIP SIP SDK 2.1
VoIP SIP SDK 2.1 Ranking & Summary
VoIP SIP SDK 2.1 description
VoIP SIP SDK 2.1 is designed as a powerful tool which can help you easily build voip softphone to dial and receive phone calls features in your software application.
VoIP SIP SDK provides a powerful and highly customizable solution to quickly add SIP based dial and receive phone calls features in your software applications. It accelerates the development of SIP compliant softphone with a fully-customizable user interface and brand name.
The VoIP softphone sdk contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software). It enables a worldwide communication over the internet or intern networks either by speaking and delivers superior voice quality by voip softphone. It supports DTMF, adaptive silence detection, adaptive jitter buffer! Also using our voice conversation sdk you can work through firewall or NAT.
Voip sdk is based on IETF standards (SIP, STUN, etc.), so it should be compatible with other standard based products such as Asterisk, Open SER other.
- Dynamically loadable codecs
- Registrar support
- Play wav files into conversation
- Record conversation into file
- Hold/Retrieve call
- Forward Call (Blind Call Transfer)
- Transfer Call (Attended Transfer)
- Mute Sound
- VPN support
- Authorization Id
- Noise reduction
- Auto gain
- Jitter buffer parameters
- Samples on Delphi, C#, VB, VB.NET, C++ 2005, C++ 6.0, HTML (SIP ActiveX)
- Windowless samples on C++ and .NET
- Adaptive silence detection
- Adaptive jitter buffer
- STUN support
- Comes as ActiveX control
- Royalty free licensing
- No Yearly/Monthly fee
- IM interface
- Codec disabling
- Playback of different formats of WAV files
- Software volume control
- Available codecs:
- G.726 (16k 24k 32k 40k)
- Speex (Narrow, Wide)
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