2596

2631پ

2755

كارشناسي ارشد

مخابرات

اصفهان: دانشگاه صنعتي اصفهان، دانشكده برق و كامپيوتر

1383

[الف]، نه، 124ص .: مصور، جدول، شكل، نمودار

مسعودرضا هاشمي

فرامرز هندسي

08/04/84

1396/10/26

كتابنامه

برق و كامپيوتر

مهندسي برق و كامپيوتر

ID2755

به فارسي و انگليسي : قابل رويت درنسخه ديجيتالي

Abstract Voice over IP VoIP uses Internet protocol IP to transmit voice as packets over an IPnetwork VoIP can be achieved on any data network that uses IP like Internet Intranet andLocal Area Networks VolP can provide convenient low cost telephone calls over theInternet and has the potential to supercedeanalog telephone systems In this thesis a new VoIP system has been proposed This system employs a multiratespeech codec which can vary its rate on a frame basis depending on the network channelconditions For monitoring channel conditions interarrival jitter is calculated using thetime stamp information in the RTP header for every frame We have estimated theavailable bandwidth by monitoring jitter integral in a sliding window The basic idea is thatone way delays of the periodic packet stream show increasing trend when the stream rate islarger than available bandwidth We then developed the multirate speech codec mode assignment algorithms for VolPusing the estimated channel conditions as follows 1 Decreasing the rate if jitter integral shows increasing trend or a packet loss happens 2 Increasing the rate if it is possible A method has been developed for the second part in which a number of packets with the higher rate are sentand the arrival times are monitored Experimental results demonstrate the superiority of the VoIP with the multirate speech codec over the conventional VolP systems d I f Ci 1 lr v VIJ I

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