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TOPS
SMSC Relay ActiveX SDK library is a set of optimized ActiveX components enabling quick creation of applications implementing the communication with the SMSC through TCP/IP protocol.
Changes in version 2.3
- Asynchronous message submit in SMPP protocol (SubmitMessageAsunc function).
- Unbind functionality in SMPP protocol (FinalizeSession function)
- ASCII and GSM frontend functionality added to the SMPP protocol.
- InquiryMessage function in UCP component interface.
- DeleteMessage function in UCP component interface.
- Added SubmitData function for SMPP protocol.
- ISO 8859-1 (Latin-1), ISO 8859-5 (Cyrylic) and ISO 8859-8 (Hebrew) encoding.
- Performance improvements.
- Minor tweaks and fixes.
The library consists of four components, each of them implements one communication protocol. Supported protocols are SEMA (version 8.1), CIMD2 (version 1.2), UCP (version 2) and SMPP (version 3.4). Key features:
- Multithreading construction ensures maximum efficiency of the sending and receiving messages process
- Receiving messages sent by users to special (short) numbers (contests, surveys, voting services etc.)
- Possibility to send and receive SMS and EMS messages, messages containing images, animations, logos, sounds etc.
- Possibility to send and receive seven-bit text messages, eight-bit messages with binary data as well as text messages encoded in Unicode (UCS2) standard
- Full information about the message delivery status
- Support for messages containing user header (UDH) and for encrypted messages
- Possibility to send messages in Direct Display mode (FlashSMS)
- Possibility to attach an alphanumeric signature to messages
- Network connection keep-alive functionality
- Possibility to use in .NET environment and all environments that support ActiveX (also in ASP environment.)
- Tested in cooperation with many SMSC configurations
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conaito Technologies
VoIP SIP SDK for .NET and ActiveX - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.
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 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 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 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 suppression, noise suppression, reverb suppression and mute microphon
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)
* NEW in v2.0! Multi-line (up to 512 simultaneous calls) support (Multiple Concurrent calls)
* NEW in v2.0! Call Hold support
* NEW in v2.0! Call Transfer support
* NEW in v2.0! Instant text messaging (MIME) support and typing indication
* NEW in v2.0! Mute microphone/speaker for each line
* NEW in v2.0! DNS SRV resolution for SIP servers (RFC 3263)
* NEW in v2.0! Stereo codec (L16)
* NEW in v2.0! RTCP
* NEW in v2.0! Auto-answer
* NEW in v2.0! Do Not Disturb (DND)
* NEW in v2.0! Adaptive jitter buffer
* NEW in v2.0! Adaptive silence
* Advanced configurable digital voice processing features
...and much more. Try it today!
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conaito Technologies
VoIP SIP SDK - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.
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 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 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 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 suppression, noise suppression, reverb suppression and mute microphon
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)
* NEW in v2.2! Multi-User conference support
* NEW in v2.0! Multi-line (simultaneous calls) support (Multiple Concurrent calls)
* NEW in v2.0! Call Hold support
* NEW in v2.0! Call Transfer support
* NEW in v2.0! Instant text messaging (MIME) support and typing indication
* NEW in v2.0! Mute microphone/speaker for each line
* NEW in v2.0! DNS SRV resolution for SIP servers (RFC 3263)
* NEW in v2.0! Stereo codec (L16)
* NEW in v2.0! RTCP
* Auto-answer
* Do Not Disturb (DND)
* Adaptive jitter buffer
* Adaptive silence
* Advanced configurable digital voice processing features
...and much more. Try it today!
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conaito Technologies
SIP Phone DLL - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.
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 and websites. It accelerates the development of SIP/ RTP compliant soft phone with a fully-customizable user interface and brand name.
The conaito SIP Phone DLL 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 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 suppression, noise suppression, reverb suppression and mute microphon
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)
* NEW in v2.0! Multi-line (up to 512 simultaneous calls) support (Multiple Concurrent calls)
* NEW in v2.0! Call Hold support
* NEW in v2.0! Call Transfer support
* NEW in v2.0! Instant text messaging (MIME) support and typing indication
* NEW in v2.0! Mute microphone/speaker for each line
* NEW in v2.0! DNS SRV resolution for SIP servers (RFC 3263)
* NEW in v2.0! Stereo codec (L16)
* NEW in v2.0! RTCP
* NEW in v2.0! Auto-answer
* NEW in v2.0! Do Not Disturb (DND)
* NEW in v2.0! Adaptive jitter buffer
* NEW in v2.0! Adaptive silence
* Advanced configurable digital voice processing features
...and much more. Try it today!
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BigSpeed Computing Inc.
BigSpeed Video Chat SDK is a set of two ActiveX controls (Client and Server) that
lets you set up a private video chat network for secure messaging.
The transfered data are scrambled using on-the-fly 128-bit AES encryption. The security is provided in two different modes: symmetric encryption with shared secret key and asymmetric encryption with 1024-bit public/private keys.
The server is responsible for keeping information on and authentication of online peers.
All peers connected to a particular server build a single private community.
There is adjustable video frame quality and motion detection for better video compression
A lot of badwidth can be saved by not sending any frame which is not quite different than the last one.
Ability is provided to conduct one-to-one or or one-to-many voice/video sessions.
In addition, the average bitrate is reduced thanks to the silence compression. In the encoder, a Video activity detector with an adjustable trigger level is used to distinguish between regions with normal speech activity and those with silence or background noise.
Significant echo suppression is achieved using a robust double talk detector.
Capability to mute recording and mute playback is implemented.
The incoming and outgoing audio data are available in raw PCM format for visualization in the application.
Two special events notify about signal overloading (possible distorsions) and transmission delays (network congestions).
Two different kind of text information can be exchanged between peers: alert and chat messages.
File transfer capability is implemented for direct file exchange between peers.
BigSpeed Video Chat SDK utilizes an elegant event-driven paradigm for easy integration into Windows applications.A special attention is paid to provide responsive user interface while avoiding multithreading problems.
Sample applications are included in Visual Basic 2005 and Delphi 7.
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