Pruebas de escenarios avanzados de Telefonía IP e IP TV

Aquest ítem conté el treball original, defensat públicament amb data de 2 de maig de 2017, així com una versió millorada del mateix amb data de 4 de desembre de 2017. Els canvis introduïts a la segona versió són 1) correcció d'erradesThe aim of this project is to test different real advanced scenarios of video and audio network-based delivery in the laboratory, in order to use these scenarios for teaching. Nowadays the great capacity of today's networks have allowed more and more users able to enjoy multimedia content over the network. Video and audio streaming have become one of the most used implementations for this purpose. However, streaming over HTTP/TCP has problems because it does not adapt to the network state in case of delays or losses. In these situations TCP reacts by decreasing the transmission window, thus downgrading the capacity down the video playback stops. A new standard was born to correct to these network problems called MPEG-DASH (Dynamic Adaptive Streaming over HTTP). This new universal standard allows the client to automatically choose the quality of the content being played and adapt it to the available bandwidth, thus allowing uninterrupted playback of the video. One of the main issues of the internet is the fact that sometimes the content location is so far from the client that affect speed and latency. Furthermore in some cases the content can traverse through various networks adding latency. To avoid this, a possibility is to use an overlay network that brings content closer to the user according to the geographic area, achieving a better performance of their connection. This type of network is called CDN (Content Delivery Network). This evolution has been not only in the video transmission telephony has also suffered an absolute revolution in the world of communications with the invention of IP telephony. Every day more companies use this technology for their communications, allowing better management of telephone resources with a cheaper price, in addition to have greater ease of use and implementation. The IP telephony makes possible making a call from any place where there is internet access. Despite its many advantages, there are still drawbacks about the quality of service of VoIP network as a consequence of latency, jitter and packet loss. Apart from these technological limitations, there is also the problem of NAT (Network Address Translation) traversal. When the connection goes through this NAT, the user IP address and port changes and causing problems when receiving the RTP audio the other user. In summary, this document will detail the results obtained from the tests with the technologies previously mentioned and the installation details in order to be implemented in the laboratory