| dc.description.abstractenglish | The voice service requires protocols for signaling (in the control plane) and protocols for data transfer (in the userplane). Protocols used for signaling of the QoS-enabled VoIP in NGN are SIP and Diameter, which work on application layer regarding the OSI (Open System for Interconnection) protocol layering model (both protocols are covered in Chapter 7). For the voice data transfer the QoS-enabled
VOIP uses RTP/UDP/IP protocol stack. Above the RTP [[2]], which belongs to transport layer together with the UDP, is the VoIP application that provides voice codec functionalities.
RTP is designed to provide end-to-end transport functions for provision of real-time services in Internet, such as audio (e.g., VoIP), video (e.g., IPTV - Internet Protocol Television), and multimedia data, over unicast (e.g., for VoIP) or multicast networks (e.g., for IPTV or multimedia streaming services). However, RTP is typically present at end hosts that have an established session, and hence it does provide QoS guarantees and resource reservation for the real-time services (they have to be implemented by the network, such as NGN, that is used to transfer real-time data). RTP is used together with RTP Control Protocol (RTCP) protocol, where RTP is used to carry media streams (e.g., audio, video, multimedia, etc.) while RTCP is used to provide control feedback between two end points of an RTP session regarding the transmission statistics, QoS, and synchronization of multiple streams. So, RTP standard, defines a pair of protocols, namely RTP and RTCP. Both | spa |
