Sip and Uplink Skype To Sip Adapter Software for Windows

SIP Phone DLL
$999 - conaito Technologies

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

Here is a list of the main features of the conaito SIP Phone 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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VoIP SIP SDK
$999 - 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.
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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VoIP Provider - SiPHome
$0 - SiPHome

Jeder Nutzer, der SiPHome zum ersten Mal testet, kann seinen ersten Anruf kostenlos in mehr als 200 Länder weltweit führen.

* Erwerben Sie mehr SiPHome-Guthaben, um in der ganzen Welt zu günstigen Minutentarifen ins
normale Telefonnetz und Handynetz anzurufen.
* Bei einem Wechsel zu Classic Plus erhalten Sie gratis eine Rufnummer von SiPHome.

Bei SiPHome erhalten Sie eine Rufnummer, über die Sie vom Fest- und Mobilfunknetz aus zum Ortstarif erreichbar sind.

Das bietet unser kostenloser Tarif - Classic ( mtl. Grundgebühr = 0 Euro) :

* Familienangehörige, Freunde und Kollegen können Sie nun von jedem Telefon aus auf SiPHome erreichen.
* Sie können Sie zum Ortstarif anrufen.
* Sie brauchen noch nicht einmal SiPHome installiert zu haben.
* Ihre SiPHome-Nummer folgt Ihnen, egal wo Sie sich gerade in der Welt aufhalten.
* Sie können Anrufe entgegennehmen, wenn Sie bei SiPHome angemeldet sind (oder sie zu einem
anderen Telefonanschluss oder Mobiltelefon weiterleiten).
* Sie können zu weltweit verschiedenen Standorten auf beliebig viele Telefonnummern Ihre
20 SIP-Adressen bzw. 20 IAX-Adressen weiterleiten.
* Mit der Enum-Unterstützung können Sie SiPHome-Anrufe auf Ihre Handy- und Festnetznummer
auch dann entgegennehmen, wenn Sie nicht online sind.
* ENUM= Empfangen Sie Ihre Anrufe, egal wo Sie sind ? ob einfach nur unterwegs oder auf großer Reise am anderen Ende der Welt ohne Internetzugang.

Mit SiPHome SMS können Sie schnell und günstig mit Freunden Kontakt aufnehmen, die offline, beschäftigt oder unterwegs sind.

SiPHome SMS

* Lassen Sie den Kontakt zu Ihren Freunden nie abreißen, egal ob sie offline oder unterwegs sind.
* Praktisch ? geben Sie lange Texte einfach und bequem über Ihre Tastatur ein.
* SMS in alle Handynetze schon ab 12 Ct.

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Uplink Skype to Sip Adapter
$27.5 - NCH Swift Sound Software

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.

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VaxVoIP SIP activeX SDK
$1500 - VaxSoft Inc

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.

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Telepati SIP Phone Freeware
$0 - Research Lab Inc

Telepati SIP Phone Freeware allows you to make PC-PC phone-phone calls over the Internet. Developed using Research Labs VOIP SIP Phone SDK, this free soft phone brings SIP protocol support for ActiveX. With this phone once you set the gatekeeper proxy with the username and password from your providers, you can connect and start speaking with anyone on the internet. Now VOIP open source projects have made it easier to implement a customized VOIP Solution eliminating costly PSTN lines. Though this implementation might turn out fairly complex, our VOIP Implementation Team at Research-Lab will guide you remotely for the same. Tested and works with 75% of the providers.

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