SIP PHONE SDK

VaxVoIP SIP Phone SDK exports functions that enable the activation of SIP-based video conferences.

VaxVoIP SIP Phone SDK offers functionality for the development of call center coaching services, enabling real-time instructions from supervisors to agents while ensuring that customers do not hear the supervisor's voice.

VaxVoIP SIP SDK supports a Voice Changer feature, allowing real-time voice modulation to sound like various characters such as a robot, a chipmunk, a drunk grandpa, a teen boy, or someone who has inhaled helium.

VaxVoIP SIP SDK provides functionality to develop an interactive intelligence-based answering machine detection feature. For in-depth information, run the sample code and demo application. Please note that this feature is exclusively available for the MS Windows OS SDK.

Creating Softphones for Various Hand-Held Devices: Developing softphones for Android-based devices, Apple iOS-based devices (including iPhone, iPad, and iPod), and other handheld devices is a straightforward process. Download the sample codes and SDK for comprehensive details.

VaxVoIP SIP SDK allows registration with the SIP VoIP Server using a Login Account and Password.

Seamless Integration of SIP-Based Instant Messaging and Presence: VaxVoIP SIP SDK simplifies the addition of SIP-based Instant Messaging and Presence features to your integrated softphone.

Our SDK fully supports the SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) protocol, an extension of SIP that facilitates the exchange of SIP-based chat messages and status indicators (such as online, offline, away, on the phone, etc). Adding SIP-based chat functionality is a straightforward process; please explore the sample code and demo for comprehensive details.

Effortless Dialing and Receiving Phone Calls: With VaxVoIP SIP SDK, you have the capability to initiate and accept phone calls seamlessly via any SIP-based server, gateway, or ITSP (Internet Telephony Service Provider).

This versatility empowers your VoIP applications to connect and communicate across the SIP network with ease.

Customizable Line Initialization: VaxVoIP SIP SDK allows you to configure the component with a user-defined number of lines. You have the flexibility to start with as few as 4, 8, 10, 20, or even more lines, tailoring your VoIP application to your specific requirements.

This feature is instrumental for initiating conference calls, facilitating consultative call transfers, managing the dialing and receiving of multiple phone calls, and serving various other purposes..

Users have the capability to initiate and accept multiple calls, enabling the initiation of conference calls.

During the SEC session, users have the ability to place any active line on hold.

In a SIP environment, you can forward an incoming call to another phone number, user name, or SIP account.

Effortlessly transfer a call to another phone number, user name, SIP account, or SIP URI.

To eliminate acoustic feedback in the VaxVoIP SIP SDK, an echo canceller is introduced.

Hands-free or internet telephony introduces several challenges, with the primary issue being the interaction between the loudspeaker and the microphone. The sound from the loudspeaker reflects back into the microphone and is then sent back to its source. This results in the far-end participant perceiving the reflected sound as an echo.

The VaxVoIP SIP SDK includes advanced Noise Cancellation technology, which effectively reduces or suppresses background noise to enhance the quality of the outgoing speech, ensuring clear and high-quality communication.

We offer support for AGC (Auto Gain Controller), a feature that automatically adjusts the input voice gain or volume based on the incoming signal level. This ensures that the audio remains consistently clear and at an appropriate level during communication.

While on a phone call, you have the capability to record the conversation into a wave (.wav) file for future playback.

The VaxVoIP SIP SDK provides methods for playing wave (.wav) files to the remote end of the call. This functionality allows you to transmit audio files to the other party during the call.

You can use a SIP outbound proxy to facilitate making and receiving phone calls from a device located behind a NAT (Network Address Translation) or firewall. This outbound proxy acts as an intermediary between your device and the SIP server, helping to traverse the NAT or firewall and enabling seamless communication.

Many Internet Telephony Service Providers (ITSPs) offer support for NAT SIP pass-through, which is a common solution for addressing NAT traversal issues in SIP (Session Initiation Protocol) communication. To ensure proper NAT pass-through, it's advisable to consult your specific ITSP for guidance and configuration details. In many cases, ITSPs provide NAT-friendly SIP servers to help resolve NAT pass-through challenges and ensure smooth communication in network environments with NAT or firewall configurations.

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 commercial routers support symmetric NAT type, even Microsoft Windows SERVER OS built-in NAT is also base upon symmetric NAT type. Please see STUN RFC for more details.

The VaxVoIP SIP SDK includes support for the keep-alive feature. When enabled, the VaxSIP component initiates the transmission of keep-alive packets, which helps in keeping a port open at the firewall endpoints. This functionality ensures that the SIP communication remains active and responsive even in network environments with firewalls or NAT devices.

The VaxVoIP SIP SDK supports both narrowband and wideband codecs, allowing it to work effectively with a wide range of Internet connections. This versatility enables high-quality audio communication across different network environments, ensuring compatibility and performance regardless of the type of Internet connection being used.

Jitter buffers play a crucial role in mitigating variations in audio delay by buffering received packets and adjusting their playback timing. This process helps to smooth out the delivery of audio to the user, resulting in a more consistent and seamless audio experience.

Packet Loss Concealment (PLC) is a technique used to alleviate the effects of lost or discarded packets in audio or video transmission. PLC is particularly effective when dealing with a small number of consecutive lost packets, such as a cumulative loss of 20-30 milliseconds of speech, and in situations with low packet loss rates. It helps maintain audio or video quality by filling in the gaps created by packet loss, providing a smoother and more continuous playback experience.

VaxVoIP SIP enables applications to generate DTMF (Dual-Tone Multi-Frequency) tones. This functionality allows your applications to produce the specific audio tones associated with telephone keypad digits, making it possible to send DTMF signals during VoIP calls for various purposes such as navigating automated systems or interacting with telephony services.

In addition to generating DTMF tones, VaxVoIP SIP SDK also supports DTMF tone detection. This means that the SDK can recognize and interpret DTMF tones received during a call, allowing applications to respond to or process these tones as needed. This two-way DTMF support makes it versatile for applications that require interaction with telephony services or automated systems.

VaxVoIP SIP SDK supports the Do Not Disturb (DND) feature. This feature allows applications to enable or disable DND mode, which prevents incoming calls or notifications from disturbing the user. It's a valuable feature for controlling call interruptions and ensuring privacy or uninterrupted workflow when necessary.

Users can directly control the microphone and speaker volume when using VaxVoIP SIP SDK. This feature provides users with the ability to adjust the audio input (microphone) and output (speakers) levels according to their preferences, ensuring a comfortable and tailored audio experience during calls.

After purchasing the license key, you will receive one year of access to new product versions and upgrades free of charge. This ensures that you can stay up to date with the latest features, improvements, and security enhancements for your software.

Please download a demo application or any of our sample codes and try them out for up to 30 days.

This trial period allows you to explore our software and its features, helping you evaluate its suitability for your needs before making a commitment.

You have the option to develop your SIP-based softphone with a custom graphical user interface (GUI) and branding quite easily.

This flexibility allows you to create a unique softphone application tailored to your specific requirements and brand identity. Sample source codes for VB, VB.Net, C#, Delphi and VC++ are available on our website, please click the SAMPLES link to find out more.

Yes, video conferencing is supported. Please visit SAMPLES for the trial and testing of video conferencing feature.

Yes, Voice Changer is supported, 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.

Yes, answering machine detection with interactive intelligence is supported. For testing, please download the sample code or demo application and run it.

Yes, it is really easy to develop such feature by using VaxVoIP SDK exported methods. In which supervisor instructs to the agent in real-time. But customer does not hear the voice of the supervisor.

Yes, it is really easy to develop softphone for iPhone, iPad and other android based phone and tablet devices. Please have a look at demo and sample codes available on the website.

Yes, one can easily develop SIP based Instant Messenger. Please visit the FEATURES link for more details.

Yes, it is supported, Please visit the FEATURES link for more details.

Our SIP SDK works without any problem with Asterisk and all other SIP based SERVER and SERVICE providers.

G711 A-Law, G711 U-Law, G.729, iLBC and GSM 6.10 are supported.

Please visit the FEATURES link for more details.

Yes, it is supported. Please visit the FEATURES link for more details.

Yes, it is supported. Please visit the FEATURES link for more details.

Yes, it is supported. Please visit the FEATURES link for more details.

Yes, phone conversation recording feature is supported. Please visit the FEATURES link to find out more.

Yes, DTMF generation and detection both are supported. Please visit the FEATURES link for more details.

Yes, we provide customization of our SDK. Please send us your requirements and a sales person will be contacting you soon.

Please click the FEATURES link for more details.

There is no difference, except 'Evaluation Copy' message box and expiry.

Yes, our SDK's source code can be shared exclusively with user rights. Furthermore, we offer training sessions tailored to help users fully leverage the capabilities of the source code.