VaxVoIP SIP Phone SDK provides tools and components to quickly add SIP (Session Initiation Protocol) based dial and receive phone calls feature in your web pages and software applications. It accelerates the development of SIP softphone, webphone and web dialer having your own GUI (graphical user interface) and brand name. VaxVoIP SIP SDK also provides sip tunneling server software that makes your VaxVoIP SDK integrated softphone and webphone work in VoIP/SIP blocked countries.



VaxVoIP SIP SDK includes SIP activeX , SIP OCX , SIP DLL , SIP Lib and SIP webplugin, so you can use the one you like the most. It is really easy to incorporate VaxVoIP SIP Phone SDK in your web pages and/or applications. Softphone and webphone sample source codes are available to download.

VaxVoIP SIP SDK can also be used to develop softphone and webphone for MS Windows OS, Apple MAC OS, Apple iPhone, Apple iPad and other Hand-Held devices.

It delivers superior voice quality by integrating advanced digital voice processing features including acoustic echo cancellation, noise cancellation and adaptive jitter buffering. For more details, please visit the features link.

SIP TUNNELING SERVER SUPPORT

VaxVoIP SIP Phone SDK provides SIP tunneling Server which makes the SIP and VoIP communication possible and let VaxVoIP SDK integrated softphone and webphone users dial and receive VoIP phone calls in VoIP blocked countries.

DEVELOP CALL CENTER LIVE CALL COACHING FEATURE

VaxVoIP SIP Phone SDK provides functionality to develop call center coaching services. In which supervisor instructs to the agent in real-time. But customer does not hear the voice of the supervisor.

VOICE CHANGER SUPPORT

VaxVoIP SIP SDK supports Voice Changer, which works in real-time and let you sound like a robot, a chipmunk, a drunk grandpa, a teen boy or someone who just inhaled helium. Such feature is only available for MS Windows and iOS SDK.

ANSWERING MACHINE DETECTION SUPPORT

VaxVoIP SIP SDK exports functionality to develop interactive intelligence based answering machine detection feature. Please run sample code and demo application/web phone for more details. Such feature is only available for MS Windows SDK.

IE, FIREFOX, GOOGLE CHROME, SAFARI AND OTHER WEB BROWSERS SUPPORT

Webphone developed by VaxVoIP SIP SDK works with the latest versions of almost all web browsers like: Microsoft's Internet Explorer, Fire Fox, Google Chrome, Safari and other Mozilla based web browsers.

DEVELOP SOFTPHONE FOR iPHONE, iPAD, iPOD & HAND-HELD DEVICES

It is really easy to develop softphone for Android based devices (HTC, SAMSUNG, XPERIA etc.), Apple iOS based devices (iPhone, iPad, iPod etc.) and other Hand-Held device. Please download the sample codes and SDK for more details.

SIP PROXY AUTHENTICATION

VaxVoIP SIP SDK enables to register with the SIP proxy server by providing Login Id and Login password.

DEVELOP SIP BASED INSTANT MESSENGER

One can easily add SIP based Instant Messiging and Presence feature in its VaxVoIP integrated softphone. VaxVoIP SDK supports SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) protocol.

SIMPLE is the SIP protocol extention to send and receive SIP based chat messages and status (online, offline, away, on the phone etc). So, it really easy to add and develop SIP based chat feature, please see the sample code and demo for more details.

Chat feature is only available for MS Windows SDK, it is under development for MAC, iPhone and other SDKs.

DIAL/RECEIVE PHONE CALLS

You can dial and receive phone calls through any SIP based server, gateway or ITSP (Internet Telephony Service Provider).

MULTI-LINES SUPPORT

VaxVoIP SIP SDK enables to initialize the component with a user-define specific number of lines. You will be free to start the component with 4, 8, 10, 20, 40, 80 or more number of lines.

Such feature is use to start conference call, consult call transfer, dial/receive multiple phone calls and for many other purposes.

MULTI-PARTY VOICE CONFERENCE

User can dial and receive multiple calls to start conference call.

LINE HOLD

During the call session, user can put any line on hold.

CALL FORWARDING

Forward an incoming call to other phone number, user name or sip account.

CALL TRANSFER

Transfer a call to other phone number, user name, sip account or sip uri.

ENCRYPT SIP ACCOUNT SETTINGS

If you hard-code the SIP account settings in your webpage then any user can easily view those settings by viewing the source of that webpage.

To prevent such situation, VaxVoIP allows you to encrypt the hard-coded/static SIP account settings and then use them in your webpages. For more details, please see Encrypt SIP account settings in the documentation section.

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 produce high quality of output speech.

AGC (AUTO GAIN CONTROLLER)

We support AGC (auto gain controller). AGC is a mechanism by which input voice gain/volume is adjusted automatically based on input signal level.

RECORD CONVERSATION INTO WAVE (.WAV) FILE

During the phone call, you will be able to record the conversation into wave (.wav) file for later play back.

PLAY WAVE (.WAV) FILE TO THE REMOTE END

VaxVoIP SIP SDK export methods to play wave (.wav) file to the remote end.

FRIENDLY TO NAT AND OTHER FIREWALLS

User can set SIP outbound proxy inorder to make and receive phone calls behind the NAT/firewall.

In some cases, ITSP (Internet Telephony service provider) support outbound proxy. Outbound proxy is a way to let the NAT/firewall user make and receive phone calls.

If the NAT/firewall router does not support SIP pass-through, you need to consult your ITSP if they support SIP outbound proxy. Since different NAT router vendor implement NAT differently. Typically ITSP may provide SIP outbound proxy to resolve NAT pass-through issues.

STUN is not a good idea to support NAT pass-through, because STUN does NOT support symmetric NAT type, symmetric NAT is more secure and widely use for commercial purposes. Almost all branded routers support symmetric NAT type, even Microsoft windows SERVER 2000 & 2003 built-in NAT is also base upon symmetric NAT type. Please see STUN RFC for more details.

KEEP-ALIVE PACKETS TO NAT/FIREWALL

VaxVoIP SIP SDK support keep alive feature. When you enable it, VaxSIP component starts sending keep-alive packets and keeps the port open at firewall ends.

NARROWBAND & WIDEBAND VOICE CODECS

VaxVoIP SIP SDK supports for both narrowband and wideband codecs that's why it works with all type of Internet connections.

Home:

http://www.vaxvoip.com/phonesdk.html