Our brand-new SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in your software applications. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.
This software contains a high performance VoIP conferencing client capable of delivering crystal clear sound 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/or by text messages and delivers superior voice quality by integrating advanced configurable digital voice processing features including auto gain controller, acoustic echo cancellation, noise cancellation, reverb cancellation and Voice activity detection.

· Easily make and receive SIP (Session Initiation Protocol) based phone calls through any SIP gateway or SIP compliant IP-Telephony service provider.
· VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A-Law, G711 U-Law, iLBC Codec)
· Registration on SIP Server (SIP Registrar).
· Instant text messaging.
· Microphone and Speaker Visualization support.
· Microphone and Speaker Volume with Mute support.
· Audio device selection.
· Fully-customizable user interface.
· Packetloss resistant (by using iLBC codec).
· Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
· Works with all kind of Internet connections.
· Very easy to incorporate
· AGC (auto gain controller).
· Acoustic echo cancellation or suppression.
· Noise cancellation or suppression.
· Reverb cancellation or suppression.
· VAD (Voice activity detection).

Requirements:
· Visual Basic .NET
· Visual C++ .NET
· Visual C# .NET
· Delphi .NET
· Visual Basic
· Visual C++
· Delphi
· Visual Basic 6.0, all development environments with a ActiveX support

Limitations:
· 30 days trial period

What's New in This Release:
· g729 and g723 Codec´s support
· Multiple and single Codec selection support
· Failure codes support (get SIP Message Response Code, SIP Message Response Text)
· RTP/RTCP Port setting (for inbound RTP traffic)
· Reduce audio latency and audio latency settings (properties: MinPrefetchCount, MaxPrefetchCount, MaxRTPPackets)
· Media status (Events: OnLocalMediaStarted, OnLocalMediaStoped, OnRemoteMediaStarted, OnRemoteMediaStoped)
· Get used codec per line
· Custom Ringtone (play wav) support (property: RingtoneFile)
· Play wav to a selected phone line (methods: StartPlayingAtLine, StopPlayingAtLine)
· Redirect Call to other phone line
· Load and Save Configurations (methods: LoadConfiguration, StoreConfiguration)
· Complete new, re-written and updated samples with source code
Additional:
· Fixed: Blind (unattended) transfer and Consultative (attended) transfer
· New methods such as get_CallState, get_LineResponseCode, get_LineResponseText, get_LocalIP, get_NegotiatedCodecName, get_NegotiatedPayloadType, get_Net...