About VoIP SIP Client SDK
VoIP SIP Client SDK - A powerful and highly versatile VoIP SDK. Our SIP SDK provides a powerful and highly versatile solution to add quickly SIP (Session Initiation Protocol) based dial and receive phone calls features in...
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!
Previous Versions
Here you can find the changelog of VoIP SIP Client SDK since it was posted on our website on 2015-04-28 03:00:00.
The latest version is 1.3 and it was updated on 2024-04-22 20:40:04. See below the changes in each version.
VoIP SIP Client SDK version 1.3
Updated At: 2010-07-08
Changes: * 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
VoIP SIP Client SDK version 1.2
Updated At: 2009-11-17
Changes: *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
Disclaimer
External Download
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