Integration Protocols for Voice and Data Traffic
Increasing demands for multimedia services offer integration of multimedia traffic as a hot issue in the future research areas. As a result, in the literature, many multiplexing schemes have been proposed. However, most of them have been implemented with a high complexity, others may be non-effec...
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Main Author: | |
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Format: | Thesis |
Language: | English English |
Published: |
1997
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Online Access: | http://psasir.upm.edu.my/id/eprint/10009/1/FK_1997_9_A.pdf http://psasir.upm.edu.my/id/eprint/10009/ |
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Summary: | Increasing demands for multimedia services offer integration of multimedia
traffic as a hot issue in the future research areas. As a result, in the literature, many
multiplexing schemes have been proposed. However, most of them have been
implemented with a high complexity, others may be non-effective to satisfy the
multiplexing performance criteria, while the rest are still not subjected to a wide
range of analysis. Therefore, there is a critical need for comparing some of the
recommended multiplexing schemes as well as developing a simple and effective
integration protocol while still achieving reasonable bandwidth utilization.
This thesis is intended to examine integration protocols for multimedia traffic,
with primary focusing on voice-data integration. Firstly, a survey of the existing
multiplexing schemes and related issues are presented. Next, an Adaptive Round
Robin (ARR) protocol is proposed, as an alternative for voice-data integration, and
extensively simulated. Finally, further comparisons, based on computer simulations,
are carried out for various multiplexing schemes including Strictly Priority Servicing
(SPS), Fixed Round Robin (FRR), Dynamic Bandwidth Allocation/(T1, T2) and
Queue Length Threshold (QLT).As a contribution of the thesis, the proposed protocol tries to avoid the
drawbacks of the previous multiplexing schemes besides satisfying the multiplexing
performance criteria. The protocol differs from the others in that, it gives a limited
priority for voice over data, it organizes the incoming packets to the single First-in
First-out (FIFO) output buffer rather than the only outgoing scheduling, i.e., all data
sources are polled in order according to the adaptation policy; however, before a data
source can send a packet, all active voice sources are polled in order. Thus it
provides an improvement in voice delay performance without significant effect on
data delay performance over previous protocols. In addition, simulation comparisons
between various multiplexing schemes have been discussed. In these simulations
voice packets are assumed to be generated from on-off sources (talkspurt-silence
calls), which is closer to reality and which is not considered in most of the
performance analyses of previous schemes. |
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