 VoIP accompanied with real-time bidirectional video (e.g. video telephony, or video conferencing), which also requires bounded delay but the requirements are higher due to higher traffic demands of the video (especially for better picture resolu- tions) than voice and the requirements for the synchronization between the audio and video streams because both media types (audio and video) need to be reproduced with the same time reference. That is accomplished by using timestamp approach in RTP, together with its accompanied control protocol, RTCP, which provides reporting func- tions between the client and server side of RTP/UDP/UP communication [2]. Additionally, all real-time services including VoIP, video telephony (i.e. video VoIP) or conferencing, video/multimedia streaming (typically by notion of video streaming one assume video with one or multiple audio streams, which can be called also multimedia streaming), all require low jitter (i.e. delay variation). Why? Because, the reproduction of audio or video streams at the receiver's side must be continuous (i.e. without stoppage of the audio or freezing of the video, once the playing has started), which requires playout buffering. The playout buffering should compensate all one-way delay and jitter (for both, real-time audio and video) in order to have smooth reproduction of the content (either that is speech in the earphone for VoIP, or watching video stream on the screen by using a given video player). In the case of only streaming one-way audio or video, the delay plays a minor role (as long as the user is prepared to accept a couple of seconds delay for their playout, while the jitter buffer is filled). The streaming video services have increasing importance as they have become the biggest segment of Internet traffic around 2020. According to the predictions [3], by 2021 around 68% of the total traffic will be served via fixed broadband Internet access (Figure 7.2), 17% will be served via mobile broad- band networks (3G, 4G, and 5G), while 15% will be mana