| Recently, it has become a new trend to provide audio service to customers using IP-based and packet-based internet instead of PSTN. However, as far as we know, IP network can only provide "best-efforts" service, which means that it can not guarantee whether the media stream is correctly transferred to the endpoint. So, it is necessary to research on the network topology, user device and software which will be a great support to VoIP service. Here, the thesis introduce the basic concept of VoIP with key technologies, and emphasis on how to enhance the QoS of VoIP with providing more comfortable audio quality to customers and offer a new mechanism to guarantee the audio quality by selecting codec dynamically according to the change of the network.Firstly, the thesis introduces the background of VoIP. And there is a presumptively introduction of VoIP network, with analyzing the network characteristics, system structure, connection mode, main strong points, along with introduction to elements, structure and development of NGN. On the other hand, I will explain the problem we meet in the process of migration from VoIP to NGN and offer a solution to the problem in the evolvement.Secondly, the thesis introduces the signaling technology what is widely used in VoIP, especially SIP (Session Initiation Protocol). There will be a complete specification about SIP, such as the unit, message, operation etc, with a comparison between SIP and H.323 which is another important signaling technology used in VoIP and pointing out the strongpoint and drawback of both signaling technologies. At the same time, I will introduce SDP (Session Description Protocol) briefly, which is used with other... |
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