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Piotr Wilk

Metody eliminacji efektu jittera (zmienności opóźnienia pakietów) w systemach VoIP


Evaluation of jitter eliminating methods in VoIP systems


Opiekun pracy dyplomowej: dr inż. Stanisław Krawczyk
Praca dyplomowa ife obroniona 2007-10-12
Streszczenie pracy dyplomowej:
In the past few years, voice traffic (voice communication) was being provided only by the Circuit Switched Networks (CSN, '), commonly known as PSTN, whereas the data traffic was mainly sent over Packet Switched Networks (PSN). Voice over IP (VoIP) is a technology, which arises from the merging telecommunications networks with Internet technologies, and for voice transport it uses PSN. In CSN, before the call begins, dedicated circuit is being established and resources for the communication are allocated. The circuit remains constant for the duration of the call. Due to that fact, the quality of connection is also stable during the whole conversation. IP telephony uses Packet Switched Network for voice transmission. However, due to the nature of the IP networks, it introduces such problems as delay, jitter and packet loss. These are widely known problems and exist in all kind of Packet Switched Network. Since the VoIP is Real Time Application, it is very vulnerable to such impairments and they are the source of the biggest downgrade in the perceived audio quality. Thus, elements which are trying to mitigate the effect of these impairments play very important role in the design of VoIP systems. Lots of study has been performed to minimize the effect of jitter, delay and packet loss. And since it is very difficult to eliminate them on the network side, mechanism that deals with them has to operate on the receiver side. For that purpose Buffer component was developed. It plays a crucial role in the real time voice applications. It is responsible for receiving the voice packets and delaying the playout time. It also has implemented features for handling loss packets. In other words, it is trying to minimize all crucial factors which have the biggest influence on the call connection quality. Since there exists several different methods used for designing the Buffer, it makes it very interesting issue for detailed analysis. This thesis is focused on the buffer and all the mechanisms associated with it and is organized as follows. First chapter describes technology in more details by explaining all procedure of work all modules from which VoIP system is built and illustrates all the implication associated with the VoIP telephony and reasons of such situation. Chapter two is focused on the different playout buffer algorithms. Even though TSM (Time Scale Modification) and SDA (Spike Detection Algorithm) are parts of the Buffer, separate sections for them are created. This is caused by the fact that different SDA and TSM can be used together with all kind of Playout Buffer algorithms. Chapter three tells about SDA (Spike Detection Algorithm) which is a technique that deals with extremely high increase in the packet delay, called spike. Chapter four describes about Time Scale Modification techniques, which are another way of mitigating previously mentioned drawbacks of the IP telephony. In chapter five one can find comparison and conclusion concerning all algorithms mentioned in this thesis. Last chapter presents general conclusions and future directions.