Best voip sdk software for free download

The conaito VoiceMail SDK is a powerful solution for recording and sending voice messages from your websites to any existing Email client software. The recipient receives the usual Email message with a link to Mp3 audio file. He/she can download Mp3 file and listen to it using any existing Mp3 playback program or play it direct from server.

Key features:
Record, send by Email and play mp3 file direct from website (without any need for an additional player), Recording Level Indicator, NAT (Router) & Firewall friendly (upload recorded Mp3 with HTTP post), Comes with ready-to-run sample, Freely changeable Interface design, Supports multi-language environment, Works on the most popular web browsers and web server, Works with all kinds of Internet connections, Comes with valid Microsoft Authenticode Certificate, No Yearly/Monthly fee, Very easy to incorporate.

The conaito VoiceMail SDK is compatible with Windows 98/2000/2003/ME/XP/Vista, MacOS and with Microsoft Internet Explorer 4.0 and up, Firefox, Safari, Opera, Mozilla and Netscape 7 browsers.

Potential applications of VoiceMail SDK:
Private and/or company websites can be enhanced by adding add voicemail capabilities so that visitors or customers can leave voice messages add voicemail capabilities to your website so that your visitors or customers can leave voice messages, Email business card can be embedded in emails so that recipients can click on your "Leave a voice message" link/button and send you a voicemail, Voice overs can be sent for intros to media such as a text document or even flash slide show and these can be personalised for each recipient, Greeting cards can be enhanced by adding recorded voice messages, eBay auction inserts can have "Ask us" buttons to enable customer to send queries in an simple and easy fashion, web based voicemail services can be set up for customers, VoIP services can be enhanced with voice messaging

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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.
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 cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here is a list of the main features of the conaito VoIP SIP client:
* 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 and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).

And much more! Try it today.

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

Our brand-new SIP Phone DLL 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)
* 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)
* RTCP
* Auto-answer
* Do Not Disturb (DND)
* Adaptive jitter buffer
* Adaptive silence
* PLC (Packet Lost Concealment)
* AGC, VAD, AEC, Noise Concealment

...and much more. Try it today!

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SIP DLL - A powerful and highly versatile VoIP SDK to accelerate development of SIP applications.

Our brand-new SIP DLL 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.
It accelerates the development of SIP/ RTP compliant soft phone with
a fully-customizable user interface and brand name. The SIP 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 cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here is a list of the main features of the conaito SIP DLL:
* 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 and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).

And much more! Try it today.

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SIP 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 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!

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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.
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 cancellation, noise cancellation, reverb cancellation and Voice activity detection.

Here is a list of the main features of the conaito VoIP SIP client:
* 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 and iLBC Codec).
* Registration on SIP Server (SIP Registrar).
* Instant text messaging.
* Microphone and Speaker Visualization support.
* Microphone and Speaker Volume with Mute support.
* Audio device selection.
* Fully-customizable user interface.
* Packetloss resistant (by using iLBC codec).
* Supports OLE Automation (scripting) by providing IDispatch interface and custom interfaces for C/C++ developers.
* Works with all kind of Internet connections.
* Royalty free licensing
* No Yearly/Monthly fee
* Very easy to incorporate
* VAD (Voice activity detection), Reverb, Echo and Noise cancellation or suppression, AGC (auto gain controller).

And much more! Try it today.

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conaito VoIP Evo SDK - A simple way to VoIP-enable your applications and websites.

The conaito VoIP Evo SDK contains a high performance VoIP conferencing client capable of delivering crystal clear sound even for both low and high-bandwidth users. conaito VoIP Evo users are arranged in a tree-structure where each node is a room/channel where users can talk, send instant messages and share files. More than 40 users can be in each room/channel and participate in a conference while resource usage, both memory-wise and CPU-wise, is still kept at an acceptable level for gaming and other types of resource intensive tasks.

Here is a list of the main features of the conaito VoIP Evo VoIP client:
* Arrange users in rooms/channels with topics and passwords.
* Private and public instant text messaging.
* Share files among users in the same channel.
* Push-to-talk and voice activation to enable voice transmission.
* Individual volume for each user.
* Position users using 3D sound.
* Denoising and gaining of outgoing audio.
* 8 KHz, 16 KHz and 32 KHz voice recording each with multiple voice quality levels.
* Throughput between approx. 600 to 6000 bytes/sec. depending on voice quality.
* Packetloss resistant.
… and much more!

Having the above features available makes it simple to develop any type of VoIP-enabled application, like e.g. a teaching tool, live support, chat, meeting tool, a multiplayer game or any other type of application which requires users being able to talk, type messages and send files to each other. For conaito VoIP Evo clients to be able to interact with each other they must connect to a conaito VoIP Evo server which keeps track of where users are located and forwards voice-data packets between clients. The conaito VoIP Evo server is able to serve 1000 simultaneous clients and can be hosted in both LAN and internet environments.

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The most important new feature is the ability for the admin to record conversations to files. The admin can now also stream .wav files to clients. Volume levels can now be controlled by AGC (automatic gain control) so all users have the same audio level.

* Switched to Unicode clients
* Improved AGC (auto gain controller) to ensure all users have the same volume level
* Record VoIP conferences to WAV and MP3
* Admin can store all conversations to disk (wav or mp3)
* Play (Stream) WAV to conferences (for IVR solutions)
* Added ability to do stereo playback
* Ignore audio, private text messages and channel text messages
* Fully updated examples with source code
* Phased out the OCX (but still available as v2.2)

Here is a list of the main features of the conaito VoIP EVO SDK - Enterprise/ Secure Edition:

* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (Speex Codec)
* Full Duplex - VoIP conferencing solution (multi-user conferences)
* Comes with Windows as well Linux server (running as service).
* Work with DirectSound as well with Windows Audio.
* Remote console server administration (e.g. use telnet to control the server)
* Administrator EVO Server API (VoIP EVO Admin COM control) - 2 SDK versions (VoIP EVO Client SDK and VoIP EVO Admin SDK) are available.
* Record conferences to WAV and MP3 - Voice conferencing recording (*.wav and *.mp3 format)
* Cryptography (SSL for TCP data transmission and BLOW FISH for UDP data transmission) is used for Encryption/Decryption (only by Secure Edition).
* Arrange users in rooms/channels with topics and passwords.
* Full rooms/channels control including protection.
* User status control (participant available, away) including status messages.
* Get user talking function - Check whether a user is talking.
* Private and public instant text messaging.
* Broadcast server messages - sending text messages from server to all connected users.
* Channel Messages as well Message of the Day support.

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conaito VoIP SDK ActiveX for developers of VoIP audio applications and webpages, such as voice chat, conference, VoIP, providing real-time low latency multi-client audio streaming over UDP/IP networks. Includes efficient components for sound recording, playback, encoding, decoding, mixing, resampling, reading, and writing wave files, mixer volume controls access. Provides UDP/IP server and client components for peer-to-peer, multi-user, and broadcast audio communications. Really easy to use! Working with Firewall and NAT! conaito VoIP SDK is the best way to add voice conference and text conference into your application and web pages. conaito VoIP SDK includes voip activex, voip ocx, voip com, voip dll, voip lib and voip cab, so you can use the one you like the most!

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The conaito Mp3 Voice Recording Applet SDK is a powerful solution for recording and sending voice messages from your websites to any existing Email client software. The recipient receives the usual Email message with a link to Mp3 audio file. He/she can download Mp3 file and listen to it using any existing Mp3 playback program or play it direct from server.

Key features:
Record, send by Email and play mp3 file direct from website (without any need for an additional player), Recording Level Indicator, NAT (Router) & Firewall friendly (upload recorded Mp3 with HTTP post), Comes with ready-to-run sample, Freely changeable Interface design, Supports multi-language environment, Works on the most popular web browsers and web server, Works with all kinds of Internet connections, Comes with valid Microsoft Authenticode Certificate, No Yearly/Monthly fee, Very easy to incorporate.

The conaito Mp3 Voice Recording Applet SDK is compatible with Windows 98/2000/2003/ME/XP, MacOS and with Microsoft Internet Explorer 4.0 and up, Firefox, Netscape 7 browsers.

Potential applications of Mp3 Voice Recording Applet SDK:
Private and/or company websites can be enhanced by adding add voicemail capabilities so that visitors or customers can leave voice messages add voicemail capabilities to your website so that your visitors or customers can leave voice messages, Email business card can be embedded in emails so that recipients can click on your "Leave a voice message" link/button and send you a voicemail, Voice overs can be sent for intros to media such as a text document or even flash slide show and these can be personalised for each recipient, Greeting cards can be enhanced by adding recorded voice messages, eBay auction inserts can have "Ask us" buttons to enable customer to send queries in an simple and easy fashion, web based voicemail services can be set up for customers, VoIP services can be enhanced with voice messaging

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Vax Voice activeX is the easiest way to add PC-to-PC voice chat and text chat over the LAN, WAN or Internet in your applications and Web pages. You can add many new voice and text chat related features and discover just how much better your application and web pages can be, with:

- ENCRYPTED VOICE & TEXT CHAT:
It is the best feature of VaxVoice activeX components. Using this feature, Vax Voice controls send/receive encrypted voice and text data over the Internet.

- MULTIUSER VOICE CONFERENCING:
Using this feature, three or more people can have a real-time voice conference.

- VOICE CONVERSATION RECORDING:
Using this feature, during the conferencing, you will be able to record the conversation into wave (.wav) file for later play back.

- TEXT CHAT:
During the conference, you will be able to have text chat.

- VAD (VOICE ACTIVITY DETECTION):
When you enable this feature, Vax Voice activeX starts sending voice data only when you speak.

- GSM 6.10 BASED VOICE COMPRESSION:
The VaxVoice component uses the GSM 6.10 voice compression codec; thereby reducing the required data rate to as low as 1.6Kbytes/Second for a single voice session.


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