Compression Codec Change Mechanisms During a VoIP Call

This paper explores the opportunities of using not standardized compression codecs while transmitting the voice data over the existing network infrastructure of VoIP (Voice Over Internet Protocol) providers based on SIP (Session Initiation Protocol) in the Internet. Telecommunication networks offer a large set of services aimed at satisfying the ever increasing needs of users. One of them is a VoIP service implemented through network IP/TCP or IP/UDP stack. One of the most important determinants of this process is the need to use data compression. Codec implementation allows you to manage the data compression degree by changing the width of bandwidth usage and voice quality modeling. These features have priority when using VoIP on mobile devices on pre-paid cards such as smartphones where constraints are the size of the packet, the area signal strength in your area and the computing power of the device. Therefore, it is reasonable to develop new technology that provides flexibility in this regard. First of all, implementation of new codecs and ability to put them into VoIP calls according to SIP signaling standards influence on lowering the cost of deployment and then final costs of calls from subscribers located anywhere around the world with access to the Internet even at low bandwidth. The second advantage is the lower network load affecting the reduction of the costs of its maintenance by the owners of the network or service providers.