Best voip software for free download

Make FREE pc-to-pc, pc-pc and cheap pc-to-phone, pc-phone, pc-to-fax, pc-fax, phone-to-pc, phone-pc, fax-to-pc, fax-pc calls over Internet/VoIP or ISDN/PSTN telephone networks. Use your computer as internet telephone, USB phone, USB handset, ISDN telephone, ISDN fax, answer phone, fax software, voicemail, caller ID, CLIP/CLIR, audio conferencing, call answer, call transfer, call forwarding, call waiting, call accounting, call recording, telephone exchange/PBX, etc. Works with all Firewalls, NAT and LAN routers - nothing to configure.

Related software: telephone, phone, fax, internet, free, calls, isdn, usb, voip, mouse, pc-telephone, pc-phone, pc-to-phone, pc-pc, pc-to-pc, pc-fax, pc-to-fax, internet-telephone, free-pc-to-phone-calls, free-phone-calls, isdn-phone, isdn-fax, usb-phone, usb-voip-phone, usb-voip-mouse, …

Designed for networking and telecom engineers, admins and communication technology educators and students, this VOIP technology quick guide covers all related technologies: SIP, H.323 and xGCP signaling architecture and technologies, VOIP transport and QoS technologies, and video and audio codecs.

A comprehensive VOIP technology guide for networking and telecom professionals.

An easy to use training reference for telecom and data networking technology students to get an overall picture of VOIP communications.

VOIP network architecture and communication protocols.

Covers all major VOIP signaling technologies, such as SIP, H.323 and xGCP (H.248).

Presents VOIP transport (and QoS technologies.

Detailed information on commonly used video and audio codecs.

Comprehensive glossary and dictionary for the VOIP technologies and applications.

Designed by experts with decades of experience in data and tele- communication industries.

Related software: voip, protocols, architecture, technologies, signaling, voip protocols, voip architecture, voip technologies, voip signaling, …

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 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 v2.4! Encrypt SIP account settings (protect your SIP account settings in websites)
* NEW in v2.3! Secure Weblicensing (protect your license in websites)
* NEW in v2.2! 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!

Related software: sip, sdk, sip/, rtp, activex, lib, dll, softphone, voip, voice, conference, text, component, com, chat, conversation, sip sdk, sip/ rtp, sip activex, sip lib, sip dll, sip softphone, voip sdk, voice conference, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip chat, voice conversation, …

Uplink connects SIP protocol VoIP calls to the proprietary Skype phone network. It works in both directions.
The SIP protocol has become the industry standard for VoIP. Thousands of telephone companies, IP phone manufacturers and virtual IP based PBX systems use this protocol to connect calls. The problem is the proprietary IP phone 'Skype' has an almost cult following in the youth market and sometimes the call rates for SkypeOut are discounted. This software lets you connect calls between the two systems.

Typical Applications:
~ Call centers who want to be able to advertise a Skype number for customers.
~ Businesses running a PBX like the Axon PC or Asterisk Linux Based Phone System who want to advertise a Skype number.
~ To use SkypeOut on your dial plan to make those calls to towns where Skype rates are cheaper.
~ To use Sip Based Voicemail Systems or Sip Based IVR Systems (for example information lines) on Skype calls.

Features:
~ Connects both incoming and outgoing calls to the Skype network.
~ Can be used to make SkypeOut calls from a SIP device or SIP PBx.
~ Can receive SkypeIn calls and direct them to your SIP extension.
~ Fully complies with the RFC3261 for SIP signalling.
~ Quick and easy operation.

Related software: sip, skype, adapter, converter, voip, asterisk, sip to skype, skype adapter, skype to sip, skype to sip converter, voip adapter, asterisk skype, …

Functional empire dual-mode phone for a small price

The Pirelli DP-L10 combines cellular and Voice over IP and offers ease of handling. When a WLAN hotspot available, notify the Pirelli DP-L10 the availability of cost-effective and allows talks on Internet telephony. With numerous features, this dual-mode phone with a high quality and reliable device for the use of mobile Voice over IP at a low price. The Pirelli DP-L10 will SiPHome without cellular contract sold, and is thus all the customers of T-Mobile, Vodafone, E-Plus and O2 use.

Internet and mobile telephony in the highest voice quality

The Pirelli DP-L10 is a GSM-Triband-Handy and hence as In the United States usage. In WLAN-Modus supports the WLAN standard. This is the wireless connection to the wireless access point, depending on the configuration manual on the Verschlüssellungs- standards WEP, WPA and WPA2 secured. With the support of current VoIP voice codecs, the DP-L10 even at lower bandwidth highest voice quality and has Sprachqualitäts- features like CNG (Comfort Noise Generation), VAD (Voice Activity Detection) and echo suppression.

Numerous features

In addition to the many features, such as the phone book for 500 entries, contact synchronization with Outlook, SMS, MMS, WAP 2.0 and e-mail functionality, 25 polyphonic ringing tones and an integrated VGA camera, meets the Pirelli DP-L10 all current high standards of the current mobile generation.

Power and comfortable call termination

The battery provides a comfortable talktime of up to 4 hours, and a standby time of up to 180 hours. Before each interview, the user via radio button to choose which Telefonart he wants to make phone calls. Moreover, even GSM-Telefonate executed whe

Related software: voice, over, siphome, telefon, business, wlan, utilities, tools, telefone, telefonanlagen, telefonanlage, lösungen, dect, betriebssysteme, adapter, enum, domain, telefonie, internet, domains, voip, tarif, sip, features, feature, voice over ip, …

Implement a professional voice response system into your business quickly and efficiently. CT Developer Studio helps you develop and host automated call handling systems which can be tailored to individual needs. Develop complex IVR software or implement a simple inbound call center, set up automatic dialing or create a fully automatic call management system in Windows. Setting up an IVR system becomes fast and straightforward using CT Developer Studio.

Get started in minutes using one of the many supplied demonstration templates. Quickly modify the details to suit your own requirements thanks to CT Developer Studio graphical interface. Drag and drop components and draw connections with a mouse onto a flow chart to specify the route for your calls. This visual development approach closely mimics the "What You See Is What You Get" approach of modern text processors, allowing you to model your resulting voice application in a straightforward way. SQL database support ensures fast and error-free operation under heavy loads even in the busiest environments. CT Developer Studio is designed to handle hundreds of phone lines and concurrent phone calls on a regular Windows PC.

The intuitive graphical IVR designer can create your IVR system in a matter of minutes, and the rich functionality and the ease with which CT Developer Studio can be customized provides you with the means of creating the call handling system you want. CT Developer Studio intuitive, icon-based interface removes the complexity of developing advanced call centers, IVR scripts and call routing applications by seamlessly integrating telephony and computers. Complex functions are built into simple graphical icons.

CT Developer Studio is an ideal tool for developing, hosting and managing advanced voice applications. CT Developer Studio supports a wide variety of telephony cards such as Intel/Dialogic, NMS, and Ai-Logix.

Related software: interactive, voice, response, system, telephone, answering, ivr, mail, voicemail, communication, sip, skype, voip, sdk, application, development, auto, dialer, dial, phone, caller, call, recording, multi, line, pstn, isdn, logix, nms, dialogic, intel, board, center, centre, outbound, incoming, display, interactive voice response system, interactive voice response, voice mail, communication system, voip ivr, voice application development, auto dialer, phone dial, caller id, call recording, answering system, multi line, phone line, ai logix, intel phone board, call center, call centre, outbound call, incoming call, call display, call system, …

conaito VoIP EVO SDK for Pocket PC and Windows Mobile - A simple way to VoIP-enable your Pocket PC applications

Conaito VoIP EVO SDK for Pocket PC contains a C-API DLL and a COM control of the VoIP EVO client that can be used from any programming language C++ and .NET Compact Framework. All client implementations provide pretty much the same interface. .NET Compact Framework users will probably be most comfortable with the COM control whereas C++ developers are most likely to prefer the C-interface DLL.

Features Overview:
* VoIP conferencing with crystal clear sound even for both low and high-bandwidth users (Speex Codec)
* VoIP conferencing (multi-user conferences) with crystal clear sound even for both low and high-bandwidth users (Speex Codec)
* Support Pocket PC and Windows Mobile (ARMv4 and ARMv4I)
* Comes with a C-API DLL (C++) and a COM control (.NET Compact Framework) of the VoIP EVO client
* Fully compatible with the clients of the VoIP EVO SDK and VoIP EVO Enterprise SDK
* Comes with Windows as well Linux server (running as service).
* Remote console server administration (e.g. use telnet to control the server)
* 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.
* File Transfer - Share files among users in the same channel.
* Ban/ Un-Ban/ Kick Clients as well as set Client connection number limits.
* Set Clients to Operator or Administrator for different rights.
* Listing of participants and rooms/channels.
* Push-to-talk (set Keyboard shortcuts) to enable voice transmission.
* VAD (Voice Activity Detection) to enable voice transmission.

and much more!

Try it today!

Related software: voip, sdk, voice, conference, instant, text, component, activex, lib, com, ocx, chat, conversation, udp/ip, gsm, 6.10, internet, telefonie, pocket, voip sdk, voice conference, conference instant voice, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip ocx, voip chat, voice conversation, gsm 6.10, internet telefonie, pocket pc, …

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 v2.4! Encrypt SIP account settings (protect your SIP account settings in websites)
* NEW in v2.3! Secure Weblicensing (protect your license in websites)
* NEW in v2.2! 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!

Related software: sip, sdk, sip/, rtp, activex, lib, dll, softphone, voip, voice, conference, text, component, com, chat, conversation, sip sdk, sip/ rtp, sip activex, sip lib, sip dll, sip softphone, voip sdk, voice conference, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip chat, voice conversation, …

Quorum is software which runs as a phone conference server on any Windows PC. Callers dial into the server to join telephone conferences.
Callers can connect to a conference call using either standard or VoIP lines on external telephone numbers or through an office phone system. The number of people who can join a conference call is limited only by your bandwidth.

When the caller selects the option to create a new conference Quorum will assign and read back the unique and secure conference number. The conference creator then gives this number to each conference participant. Whenever any person dials the server and enters this conference number, they will join in the current conversation with the other members of the conference.

Features:

~ Runs on almost any Windows PC.
~ Handles unlimited simultaneous conferences. The number of calls and conferences is limited only by your bandwidth.
~ Can connect via VoIP using the international standard SIP protocol. If VoIP is used no special hardware is needed.
~ Can connect to ordinary phone lines (or analog PBX extensions) directly using telephony devices.
~ Can connect to one Skype Number (so a Skype user can connect) using the Uplink Skype to SIP Adapter.
~ Easily allocate new conference IDs on the phone.
~ Join any pre-defined conferences by using the allocated conference number.
~ Voice prompts are easily changed via the user interface.
~ VoIP calls can be made using free VoIP SIP Service Providers saving call costs if they are an issue.
~ Installs and ready to run conferences in just minutes.
~ Very simple user interface for day-to-day easy business use.

Related software: quorum, telephone, conferencing, teleconference, business, meetings, phone, conference, calling, call, telephone conferencing, business meetings, phone meetings, conference calling, conference call, …

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:
* 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)
* NEW in v2.1! Multiple-Instances support
* 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 MP3 - Voice conferencing recording (*.mp3 format)
* Cryptography (SSL for TCP data transmission and BLOW FISH for UDP data

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 can be hosted in both LAN and internet environments.

Related software: voip, sdk, voice, conference, instant, text, component, activex, lib, com, ocx, chat, conversation, udp/ip, gsm, 6.10, internet, telefonie, voip sdk, voice conference, conference instant voice, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip ocx, voip chat, voice conversation, gsm 6.10, internet telefonie, …

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, MacOS and with Microsoft Internet Explorer 4.0 and up, Firefox, 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

Related software: voicemail, sdk, mp3, recording, applet, voip, voice, conference, text, component, chat, conversation, internet, telefonie, over, conaito, developer, toolkit, voicemail sdk, mp3 recording, recording applet, mp3 sdk, voip sdk, voice conference, text conference, voip conference, voip component, voip chat, voice conversation, internet telefonie, voice over ip, voip developer toolkit, voip toolkit, …

Skype Recorder is a surprisingly easy-to-use tool for recording Skype audio conversations. Skype Recorder can record Skype calls automatically. The software has a number of additional handy features such as an automatic record encoding into MP3/WMA/OGG/WAV format, all Skype versions support, conversations in either single or dual audio track recording, hidden mode work, a built-in audio player, a built-in FTP-client and an easy-going interface.

It is really simple to create a new recording. All you need is to run Skype Recorder.
The software will start the recording by itself as soon as you start talking on Skype. The conversation over, Skype Recorder will encode the recording to the specified format and will save the ready file on your hard drive. Due to the built-in audio player, you will be able to listen to the recording in no time and do whatever you like - delete it, move or upload to the site.

Related software: skype, recorder, record, call, conversation, voip, plugin, skype recorder, record skype, skype call, skype conversation, voip recorder, record voip, record skype call, record skype conversation, skype plugin, …

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.

Related software: sip, sdk, sip/, rtp, activex, lib, dll, softphone, voip, voice, conference, text, component, com, chat, conversation, sip sdk, sip/ rtp, sip activex, sip lib, sip dll, sip softphone, voip sdk, voice conference, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip chat, voice conversation, …

MAX Phone is expanding the soft phone experience by making it even easier for users to make VoIP (Voice over Internet Protocol) and Video over IP calls.
MAX Phone supports SIP industry standard, and is fully inter-operable with any RFC SIP 3261 compliant servers.

MAX Phone delivers SIP-based communications and services for PC-to-Phone, Phone-to-PC and PC-to-PC services.

Standard Telephone Features:

MAIN Telecom Multimedia Communicator MAX Phone has all the standard telephone features, including:

* 4 lines with ability for extension.
* Caller ID.
* Call rejection.
* Call waiting.
* Call forward & blind transfer.
* Hold.
* Call record using ASTERISK engine.
* Microphone and Speaker volume control.
* Mute Microphone.
* Redial, Auto answer & Do not disturb.
* Direct IP to IP Calling.
* Video call.

Enhanced Features and Functions:

MAX Phone also supports the following VoIP features and functions:
* Import contacts list from comma-separated values (CSV) files, CSV standard is supported by Microsoft Outlook, Microsoft Excel ...etc.
* Export contacts list to comma-separated values (CSV) files.
* Separate Calls history for each registered user.
* Separate Contacts list for each registered user.
* Audio & Video tuning wizard.
* Acoustic echo cancellation, redundant audio coding, dynamic jitter buffer and adjustment, automatic gain control, voice activity detection.
* Support for the following audio codecs: G.711, G.722.1, G.723, GSM, DVI4 and SIREN
* Support for the following video codecs: H.261 and H.263.
* Automatic selection of the best codec based on the other party’s capability, available bandwidth, and network conditions, MAX Phone switches the codec within a call in response to changing network conditions.
* UPnP-enabled NAT traversal.
* Support for DTMF.
* Live Update.
* Private Labeling and User Interface customization.

Related software: max, softphone, soft, phone, sip, voip, maintelecom, asterisk, voice, video, record, internet, telephony, …

SIP Phone DLL 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 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 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 v2.4! Encrypt SIP account settings (protect your SIP account settings in websites)
* NEW in v2.3! Secure Weblicensing (protect your license in websites)
* NEW in v2.2! 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!

Related software: sip, sdk, sip/, rtp, activex, lib, dll, softphone, voip, voice, conference, text, component, com, chat, conversation, sip sdk, sip/ rtp, sip activex, sip lib, sip dll, sip softphone, voip sdk, voice conference, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip chat, voice conversation, …

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.

Related software: sip, sdk, sip/, rtp, activex, lib, dll, softphone, voip, voice, conference, text, component, com, chat, conversation, sip sdk, sip/ rtp, sip activex, sip lib, sip dll, sip softphone, voip sdk, voice conference, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip chat, voice conversation, …

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 v2.3! Secure Weblicensing (protect your license in websites)
* NEW in v2.2! 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!

Related software: sip, sdk, sip/, rtp, activex, lib, dll, softphone, voip, voice, conference, text, component, com, chat, conversation, sip sdk, sip/ rtp, sip activex, sip lib, sip dll, sip softphone, voip sdk, voice conference, text conference, voip conference, voip component, voip activex, voip lib, voip com, voip chat, voice conversation, …