VOIP SIP SDK

Software Screenshot:
VOIP SIP SDK
Software Details:
Version: 1.9.12
Upload Date: 3 May 20
Developer: VOIP SIP SDK
Distribution Type: Shareware
Price: 697.00 $
Downloads: 42
Size: 118317 Kb

Rating: nan/5 (Total Votes: 0)

Ozeki VoIP SIP SDK is a software development kit, that allows you to make VoIP calls using SIP protocol. It can be easily included in any software developed in .Net. When you integrate Ozeki VoIP SIP SDK, your application will be capable of delivering crystal clear sound even for low and high-bandwidth users and SIP compatible devices (hardware and software). Superior voice quality is also guaranteed due to the digital voice processing features and the extended codec support.
The latest inbuilt features of Ozeki VoIP SIP SDK also provide support for video streaming. This means, that you can create VoIP SIP solutions that are capable of video phoning too.
Ozeki VoIP SIP SDK is based on IETF standards (SIP, RTP, STUN, TURN, ICE etc.), so it is compatible with other standard based products (such as PBXs like Asterisk, 3CX, Cisco, etc.).
After downloading Ozeki VoIP SIP SDK you can create a C# softphone or a VB.net softphone in minutes, or you can build SIP VoIP call services easily and quickly. Incorporating Ozeki VoIP SIP SDK in your application is really easy.
Ozdeki VoIP SIP SDK supports all the basic and advanced telephony features, and all the codecs used in VoIP communications. It also has extended audio and video features to ensure high voice and video quality for VoIP SIP phone calls.
The trial version comes with a fully-functional SIP softphone demo program, and can be easily upgraded with your purchased license to full version without any uninstallation.

What is new in this release:

  • New feature (Camera / Browser camera supprot): Browsre camera support was developed to support HTML5 access to the camera.
  • Improvement (Camera / MJPEG Streaming): The MJPEG streaming capabilities has been improved greatly. Now it allows switching resolution during streaming, and the resource usage is reduced.
  • Improvement (VoIP / relayed calls): Voice calls work well even if all parties are behind firewalls. Now the caller, the callee, and the pbx can operate behind NAT using private IP addresses. In the previous versions the relay calls did not work reliably if all parties were behind firewalls, because some SIP clients (e.g. Bria softphone on Android) were not able to provide appropriate IP address information. This situation is no fixed, the PBX can determines its own public ip using STUN, or this address can be entered manually on the config form, and the PBX can determine the IP addresses of the SIP clients if neccessary using the source adddress of the incoming packets instead of the often unreliable information provided in the SDP data.
  • Improvement (Alphanumeric SIP addressing): The PBX can operate using alphanumeric addresses instead of simply numbers. You can setup extensions using phone number and/or names if you develop a PBX
  • Improvement (Alphanumeric SIP dial plans): If you build a VoIP PBX, you may setup call routing using alphanumeric addressing.

Requirements:

.NET compatible Development Kit

Limitations:

Limited functionality

Supported Operation Systems

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