Performance of Various Codecs Related to Jitter Buffer Variation in VoIP Using SIP
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منابع مشابه
Performance of Various Codecs Related to Jitter Buffer Variation in VoIP Using SIP
Briefly speaking, there are two popular Voice over Internet Protocol (VoIP) standards, H.323 and Session Initiation Protocol (SIP). The first standard was designed by ITU and has become the basis for the widespread implementation of VoIP systems although it was not specifically designed for it. The second standard, SIP, was proposed by IETF and it is designed to connect, communicate and exchang...
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The QoS standard of a VoIP session degrades if its stringent time requirements are not met. Low end-to-end delay of the voice packets and low packet loss must be maintained. Jitter between voice packets must also be within tolerable limits. Jitter hampers voice quality and makes the VoIP call uncomfortable to the user. Very often, buffers are used to store the received packets for a short time ...
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With the rapid growth of the Internet, customers are demanding multimedia applications such as telephony and video on demand, to be available on the internet. Voice over Internet Protocol (VoIP) is a rapidly emerging technology for voice communication that uses the ubiquity of IP-based networks. The greatest challenge is the provisioning of Quality of Service (QoS) over limited bandwidth. Multi...
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The paper deals with an influence of network jitter on effective packet loss in dejitter buffer. We analyze behavior of jitter buffers with and without packet reordering capability and quantify the additional packet loss caused by packets dropped in buffer on top of the measured network packet loss. We propose substitution of packet loss parameter Ppl in ITU-T E-Model by effective packet loss P...
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ژورنال
عنوان ژورنال: Jurnal Teknik Elektro
سال: 2010
ISSN: 1411-870X,1411-870X
DOI: 10.9744/jte.8.2.103-108