Ozeki VoIP SDK - Product Guide
This article is a brief introduction about RTP implementation in relation with Ozeki VoIP SIP SDK. After reading through this page you will be fully familiar with all the essential terms concerning RTP implementation in Ozeki VoIP SIP SDK and what you will need for creating your own solution using Ozeki VoIP SIP SDK.
The Real-Time Transport Protocol (RTP) is one of the technical foundations of Voice over IP. RTP defines a standardized packet format for delivering audio and video over IP networks. It provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulating data, over multicast or unicast network services. It is commonly used in communication and entertainment systems that involve media streaming, such as telephony, video teleconference applications, television services and web-based push-to-talk features.
RTP implementation is inevitable when you would like to develop a VoIP based software product. By using Ozeki VoIP SIP SDK you do not need to think about how to implement the Real-Time Transport Protocol (Figure 1) to your application, because this SDK already includes it. That is why Ozeki VoIP SIP SDK can be your greatest help in developing any VoIP solutions.
Ozeki VoIP SIP SDK contains an RTP implementation as a base of the SDK. You can not directly reach the RTP implementation in your programs, but if you need some information about it, you can check the related developer guide page.
This article introduced you the basic knowledge about RTP protocol implementation and showed how Ozeki VoIP SIP SDK can help you to fulfill your wishes about this topic. If you have read through this page carefully, you already have all the knowledge you need to start on your own solution.
As you are now familiar with all the terms concerning this topic, now it is time to take a step further and explore what other extraordinary solution Ozeki VoIP SIP SDK can provide to you.
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