SIP Softphone ActiveX 1.1
Our brand-new SIP Softphone ActiveX 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 SIP Softphone ActiveX 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 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
...and much more. Try it today!
* 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
* 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
* Single Product/Service based Licensing - per Developer
* Single Website based Licensing - per Website Domain/URL
* Support and Maintenance
* Unlimited Redistribution/Royalty Free License
Our support team welcomes the opportunity to answer any questions you may have.
Educators and educational institutions can purchase the conaito VoIP SIP Client SDK at discounted prices. Please contact us for more information.
Single Product/Service based Licensing
We provide a `Single License` to `Single Service/Product` Licensing per Developer. You are limited to use the Single License under your Single Product or Service per Developer. So, if you have two different Products/Services, then you would have to buy two license keys.
But you are free to use a single license for as many copies of the Products/Services for which you have purchased the License Key. The exclusive selling of the conaito VoIP SIP Client SDK without it being integrated into Product/Service/Website is not permitted.
More Information about our reseller program can you find on our website.
Single Website based Licensing - per Website Domain/URL
We provide a `Single Website License` to a `Single Website`. You are limited to use the Single Web License under the Single Website Domain name. So, if you will use the conaito VoIP SIP Client SDK on two different Websites, then you would have to buy two license keys.
But you are free to use a Single Website License for unlimited users of the website for which you have purchased the License Key. The exclusive selling of the conaito VoIP SIP Client SDK without it being integrated into Product/Service/Websites is not permitted.
More Information about our reseller program can you find on our website.
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