There is a tremendous demand on real-time
multimedia delivery over wireless Internet due to the dramatic increase in
wireless communication and the growth of the Internet. However, real-time
multimedia over wireless Internet poses many challenges. First of all, the
inherent best-effort characteristics of packet-switched networks make it
difficult to provide guaranteed Quality of Services (QoS) for real-time
multimedia delivery. Secondly, wireless channels have much higher packet-loss
rate, bit-error rate, and channel instability compared with wired channels
due to the noise, multi-path, shadowing fades, user mobility, etc., which
results in fluctuating communication channel statistics.
Thirdly, real-time communication demands very strict time
limitations on the network end-to-end delay and the delay jitter.
The image shown below is a typical Internet
telephony system structure. The signaling flow via SIP or H.323 protocol suite
is used to set up and control the logical connection between the speaker and
receiver. The real-time voice packets are sampled, processed and then
transported through the RTP/UDP/IP channel to the receiver side. The
receiver does the reverse procedure to recover and then play back the voice.
In this research, we propose an intelligent
application architecture and several QoS improvement mechanisms to timely
estimate and predict the wireless network traffic statistics and dynamically
take appropriate actions to improve the overall performance of a real-time
wireless Internet telephony system. The proposed intelligent application
framework will actively evaluate the current
network situation, collect necessary statistics and give some predictions based
on time-series analysis. Based on this real-time updated
information, the application's sender side can take some adaptive actions such
as voice codec selection, forward error-correction schemes for packet-loss
concealment, or header compression to improve the QoS under currently available
network resources. Also, the application's receiver side will use these
updating traffic statistics to dynamically adjust receiver jitter
buffer size, the voice play-out time and the local packet-loss concealment
algorithms.
This work is a collaborative effort with Dr. Kevin McNeill, BAE Systems, Inc.
Publications:
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Mingkuan Liu, Jeffrey J. Rodriguez, and Kevin M. McNeill, "An Adaptive
Jitter Buffer Play-Out Scheme to Improve VoIP Quality in Wireless
Networks," Military Communications Conf. (MILCOM), October
23-25, 2006, Washington, D.C. [ PDF ]
