out any notable buffering), propagation delay (due to limited speed of the signals, which is always around the speed of light on any transmission media), and SO on. In all-IP environments, different packets may enter buffers with different queue length, which may introduce different delays to IP packets from the same end-to-end connection. The average delay variation is also referred to as jitter. To compensate for the jitter, the voice has a playback point, similar to video. However, audio is more tolerant to errors than video due to the human ear, which is more error resistant (up to several percent of the error ratio). Audio must synchronize with video in multimedia communication, such as real-time video streaming (with audio typically synchronized with the video content), video telephony (voice as audio traffic, with synchronized video, in both directions), etc. Because voice is a two-way continuous audio streaming with strict delay requirements, it is typically carried by using RTP/UDP/IP protocol stack. The voice is a most typical example of real-time traffic. However, there are differences in how the voice is treated when provided by telecom operators with guaranteed QoS and when it is being provided as any other data by OTT voice service providers (e.g. Skype, Viber, WhatsApp) over the public Internet. 2.3.2 Video Traffic Characterization Most of the video traffic is based on video information which is statistically compressed. The statistical compression is performed by source coding in the video codec (includes video coder on the sending side and video decoder on the receiving side). The most used video codecs nowadays are based on the MPEG-4 standard. Before this came its prede- cessors, MPEG-2 targeted to DVD-quality video, which was also used for transmission 2.3 Internet Traffic Characterization over telecommunication networks, and MPEG-1, which was used for recording video on storage media only (not suitable for transmission over telecommunication networks). The MPEG standards, from MPEG-1 and furt