QoS (Quality of Service) is one of the major challenges for the emerging services based on the RTP (Real-Time Transport Protocol) to deliver multimedia content. In order to synchronize audio and video components of multimedia content, RTP timestamps have to be related. Periodic RTCP SR (Real-Time Transport Control Protocol Sender Report) packets, which associate the RTP timestamps with a common clock at the transmitter, are associated with each information stream. Thus, the frequency and the timing of the RTCP SR packets often contribute to the delay before audio and video components are rendered, not just to their synchronization, because many clients will not render anything before the synchronization has been established. Recently, the concern has been expressed that the synchronization delay is especially problematic in case of network congestion. The doctoral thesis proposes a solution to this problem, which is based on the idea of isolating the RTCP packets, which carry synchronization informations, from the RTP data packets by assigning them to a higher priority class which ensures the average reduced minimum RTCP interval on the recipient’s side defined with 360 divided by the session bandwidth in kilobits per second, in accordance with IETF RFC 3550. An experimental verification of the solution proposed, in comparison with the existing priority assignment schemes, is performed using NS-2 (Network Simulator version 2). The obtained results are statistically processed using Analyse-it software add-in for Microsoft Excel. Assigning higher priority to the RTCP packets over RTP data packets, provides the average reduced minimum RTCP interval on the recipient’s side, in accordance with IETF RFC 3550, sufficient for inter-media lip-synchronization of the RTP multicast sessions regardless of network congestion while maintaining the accuracy of statistics measured by RTCP.