Adaptive PlayoutAlgorithm For VoIP • Voice Over IP (VoIP) is a common technology for performing voice calls over the Internet. • While the Internet isn’t designed for real-time data transfer, packets of information will arrive to their destination with “jittering”. • Therefore comes the need to handle and reorganize those packets before playing them to the client – Which is done by the Jitter Buffer . • Applying adaptive algorithms will enable us to react to the changing network’s behavior over the session, providing high quality conversation. • Modeling the entire system by a continuous time Markov chainenable us to test and determine the favorite algorithm.
Network Packets Transfer Scheme Time Network 1 3 2 4 1 3 1 Participant Packets Network J. Buffer Possible Network states, each with a different average load 1 Participant Talkspurt
Adaptive Playout Algorithms • Finding the best delay time is a trade-off between minimal delay and packet-lost • Applying adaptive algorithms on various generated scenarios may give us the best quality of call - MOS (mean opinion score) Network Buffer Client
The “Analytic Algorithm” seeks for optimal values 0f buffer length and packets delay – for getting maximum MOS score • Auto-regressive based algorithms gives different weight factors for past estimation and current measured packets arrival time variance in order to revaluate the next packet delay
Different algorithms may give the best MOS score for various network and participant scenarios, but on almost all cases it is advised not to use a simple – constant delay algorithm, but one that learns the network and adjust to occurring changes.