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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)
* Encrypt SIP account settings (protect your SIP account settings in websites)
* Secure Weblicensing (protect your license in websites)
* Multi-User conference support
* Multi-line (simultaneous calls) support (Multiple Concurrent calls)
* Call Hold support
* Call Transfer support
* Instant text messaging (MIME) support and typing indication
* Mute microphone/speaker for each line
* DNS SRV resolution for SIP servers (RFC 3263)
* Stereo codec (L16)
* 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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Multiple award winning email server with ultimate features and more than years history. Provides a wide range of optional components and services including secured mail services (SMTP, POP3, IMAP), AJAX webmail (Outlook replacement), anti-spam (SpamAssassin, SPF, Bayes, SURBL, Quarantine, DKIM, Greylisting, Reports), anti-virus, instant messaging (XMPP supporting AOL, ICQ, MSN, Yahoo, GoogleTalk gateways), groupware (with access from webmail, Outlook, CalDAV, SyncML), FTP (server with many security standards and the system integration), load balancing with ODBC, monitoring and report facilities (charts and reports), developers API, account types (groups, mailing lists, list servers, executables, catalogs, static routes, remote accounts, users, domains), system alert tools, remote and web administration. IceWarp eMail Server is a reliable, easy to use, and affordable communication solution that is used by more than 80 million users worldwide. Available on Windows and Unix based systems.
VoIP SIP SDK with DLL, ActiveX and .NET: http://www.conaito.com/trialdownloads/conaitoVoIP_SIP_SDK.zip
IceWarp eMail Server Software: http://www.icewarp.com/download/merak.zip |
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VaxVoIP SIP SDK provides tools and components to quickly add SIP (Session Initiation Protocol) based IP-Telephony make and receive phone calls feature in your web pages and software applications. It accelerates the development of SIP based soft phone with your own GUI (graphical user interface) and brand name.
It delivers superior voice quality by integrating advanced digital voice processing features including acoustic echo cancellation, noise cancellation and adaptive jitter buffering.
In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK.
- ACOUSTIC ECHO CANCELLATION OR SUPPRESSION
In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK. Hands-free or Internet telephony imposes several problems. The principal one is due to the coupling between loudspeaker and microphone. The loudspeaker signal is echoed back to the microphone and transmitted back to its origin. As a result the far-end participant perceives this as an echo.
- NOISE CANCELLATION OR SUPPRESSION:
VaxVoIP SIP SDK offers advanced Noise Cancellation technology that allows significant suppression of any background noise and provides high quality of output speech.
- ADAPTIVE JITTER BUFFER
Jitter buffers are used to smooth delay variations in received audio by buffering the packets and adjusting their rendering. The result is a smoother delivery of audio to the user.
- PACKET LOSS CONCEALMENT
Packet Loss Concealment (PLC) is a technique used to mask the effects of lost or discarded packets. PLC is generally effective only for small numbers of consecutive lost packets, for example a total of 20-30 milliseconds of speech, and for low packet loss rates.
- NAT AND FIREWALLS FRIENDLY
User can set SIP outbound proxy inorder to make and receive phone calls behind the NAT/firewall.
VaxVoIP SIP activeX SDK: http://www.vaxvoip.com/vaxvoipsip/VaxSIPUserAgentSDK.zip
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