Download a fully functional version of SIP SDK 3.6 and enjoy your unlimited free trial.

Our brand-new SIP Client 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 and websites. It accelerates the development of SIP/RTP compliant soft phone with a fully-customizable user interface and brand name.

The VoIP SIP Client 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 and delivers superior voice quality by integrating digital voice processing features including auto gain controller (AGC), acoustic echo suppression (AES) and noise suppression. It supports multiple lines, multi-party voice conference, call hold, call forwarding and transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more!

VoIP SIP Client SDK is based on IETF standards (SIP, RTP, STUN, TURN, ICE etc.), so it should be compatible with other standard based products such as: SER, Sip EXpress, OpenSER and Asterisk.

 New features of the VoIP SIP Client SDK:

• 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
• and much more!

 Here is a list of the main features of the VoIP SIP Client SDK:


• 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, Speex, Speex-wb, GSM6.10, iLBC, L16 and g729 & g723 Codec
• Open standards-based and interoperable with all of the major equipment vendors
• UDP and TCP support
• Multi-party voice conference support/ Conference split and join, locally mixed conferences
• Multi-line support (multiple simultaneous calls)
• SIP Instant/Chat Messaging with send/receive controlling
• Integrated STUN, TURN and ICE support
<• Comes with new sample SIP Proxy Server    to provide in bundle with the SIP    Client ActiveX a ready up own SIP VoIP    and Instant Messaging network solution.
• P2P support for directly connections    between 2 SIP clients without SIP Server
• Outbound proxy server support
• Encrypted SIP account settings (encrypted    SIP account settings in your webpage)
• Line Hold/Un-hold support
• Call forwarding and rejection
• Call transfer support
• Select media input/output devices
   on-the-fly - also during a conversation/    conference)
• Mute microphone/speaker + level indicator
• Auto-answer
• DND (Do Not Disturb)
• Adaptive Jitter buffer
• PLC (Packet Lost Concealment)
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or    suppression)
• Noise cancellation or suppression
• DTMF tones support (generation/detection)
• Recording voice conversation into PCM    WAVE (.wav) file
• Playing PCM WAVE (.wav) files to the    remote end
• Audio file memory cache
• Extended SIP URL functions
• Dynamically loadable codec support (coming soon)
• Comes as ActiveX control (Web demo with ready-up signed CAB included)
• Registration on SIP Server (SIP Registrar)
• Log file on/off setting
• Microphone and Speaker Volume with Mute support
• Keep-alive packets to NAT/firewall
• Fully-customizable user interface
• Microsoft Authenticode Certificate
• Works with all kind of Internet connections
• Friendly to NAT and other firewalls
• Keep-alive packets to NAT/firewall
• Royalty free licensing
• No Yearly/Monthly fee
• Very easy to incorporate
• Fully sample applications for various programming languages such as sample source code    for C#, VB.NET, JavaScript (Webdemo), VB 6.0 and Delphi
• For .NET framework as well and all development environments with ActiveX support

Easy, familiar, event-driven call control ActiveX
• Easy to use; quick development
• Support for .NET framework and all development environments with ActiveX support
• Very easy to incorporate

Rich call control feature set
• Multi-party voice conference support (Conference split/ join, locally mixed conferences)
• Multi-line support (multiple simultaneous calls)
• SIP Instant messaging
• Locally mixed conferences
• Hold/Mute
• Call transfer
• Call forwarding and rejection

Industry leading SIP support
• RFC3261 compliant SIP stack
• RFC 2833 out-of-band DTMF signaling
• Integrated STUN, TURN and ICE support

Comprehensive configuration support
• Select media input/output devices (on-the-fly as well during a conversation/conference)
• Configurable ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE)
• SIP proxy

Advanced digital voice processing features
• AGC (auto gain controller)
• AES (Acoustic echo cancellation or suppression)
• Noise cancellation or suppression

… and much more!

Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a SIP softphone, IVR solution, teaching tool, live support, voip chat, meeting tool or any other type of application which requires users being able to talk and type messages to each other.

For VoIP SIP clients to be able to interact with each other they must connect to a SIP gateway or SIP based IP-Telephony service provider.

Just relax!
Please, don't hesitate trying our VoIP SIP Client SDK at once and get yourself, as well as your customers, the exciting experience of easy, fast and high quality standard applications which VoIP-enable your application and website.

We hope you enjoy the new VoIP SIP Client SDK – A powerful and highly versatile VoIP SDK to accelerate development of SIP applications and websites.


 The contents of VoIP SIP Client SDK and the supported development  environments include:


The VoIP SIP Client SDK provides the documentation, samples and related libraries you need to integrate with other applications or websites.

VoIP SIP Client SDK includes a ActiveX that can be used from any programming language like Visual Basic, .NET (VB and C#), C++, Visual Basic, Delphi, ASP, JSP, PHP, JavaScript, VBScript, etc. The VoIP SIP Client SDK is designed to be used by Automation clients.

The contents of VoIP SIP Client SDK and the supported development environments include all of the necessary software components for building systems based on VoIP SIP Client SDK including documented operational software applications, examples (with source code), explanations as well as necessary service programs, libraries and components.

The supported development environments include:

• Visual Basic .NET
• Visual C++ .NET
• Visual C# .NET
• ASP.NET
• ASP, JSP, PHP
• JavaScript/HTML
• Visual Basic
• Visual C++
• Borland Delphi
• and all development environments with ActiveX support

System requirements:
Operating system: Windows XP, 2000, 2003, 2008, Vista, Windows 7

The VoIP SIP Client SDK has received Awards from well-known Download and Shareware sites including Five Star and Editor Choice Awards!

7 If you have any questions regarding the product, purchasing, licenses etc, please don't hesitate to contact us at support@sipvoipsdk.com.