VoIP SIP Client SDK 1.3


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Publisher Description



VoIP SIP Client SDK for .NET and ActiveX - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

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 conaito 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 transfer, DTMF, Packet Loss Concealment (PLC), adaptive jitter buffer, record and playing WAV and much more!

Key features

* 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, GSM6.10, iLBC, L16 g723 and g729 Codec)
* Multi-party voice conference support
* Multi-line support (Multiple concurrent calls)
* Line Hold/Retrieve support
* Call Transfer support
* Mute microphone/speaker
* Auto-answer
* Do Not Disturb (DND)
* 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 and play voice conversation into PCM WAVE (.wav) file

Try it today!

VoIP SIP Client SDK is a free trial software application from the Components & Libraries subcategory, part of the Development category. The app is currently available in English, German and it was last updated on 2010-07-08. The program can be installed on WinXP, WinVista, WinVista x64, Win7 x32, Win7 x64, Windows2000, Windows2003, Windows Vista Ultimate, Windows Vista Ultimate x64, Windows CE, Pocket PC, Unix, Linux, Java, Windows Vista Starter, Windows Vista Home Basic, Windows Vista Home Premium, Windows Vista Business, Windows Vista Enterprise, Windows Vista Home Basic x64, Windows Vista Home Premium x64, Windows Vista Business x64, Windows Vista Enterprise x64, Windows Mobile 2003, Windows Mobile 2005, Linux Console, Linux Gnome, Linux GPL.

VoIP SIP Client SDK (version 1.3) has a file size of 9.63 MB and is available for download from our website. Just click the green Download button above to start. Until now the program was downloaded 152 times. We already checked that the download link to be safe, however for your own protection we recommend that you scan the downloaded software with your antivirus.

Program Details



General

 
Publisher conaito Technologies
Released Date 2010-07-08
Languages English, German

Category

 
Category Development
Subcategory Components & Libraries

System requirements

 
Operating systems linux, other, unix, windows, windows-mobile

Download information

 
File size 9.63 MB
Total downloads 152

Pricing

 
License model Free Trial
Price $999.00

Version History



Here you can find the changelog of VoIP SIP Client SDK since it was posted on our website on 2015-04-28. The latest version is 1.3 and it was updated on soft112.com on 2017-07-01. See below the changes in each version:

version 1.3

posted on 2010-07-08

* g729 and g723 Codecs 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

version 1.2

posted on 2009-11-17

*Easily make and receive SIP based phone calls
* Multi-Line, Multi-User conference support
* Make and receive SIP based phone calls
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (G711 A/U, Speex, GSM6.10, iLBC, L16, g729)
* STUN, DTMF and Call Transfer






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Downloads last week: 1
Release Date: 2010-07-08
Publisher: conaito Technologies
Operating System: linux, other, unix, windows, windows-mobile
Type: Free Trial