17,511 research outputs found
Optimal packetisation of MPEG-4 using RTP over mobile networks
The introduction of third-generation wireless networks should result in real-time mobile
video communications becoming a reality. Delivery of such video is likely to be facilitated by the realtime
transport protocol (RTP). Careful packetisation of the video data is necessary to ensure the
optimal trade-off between channel utilisation and error robustness. Theoretical analyses for two basic
schemes of MPEG-4 data encapsulation within RTP packets are presented. Simulations over a GPRS
(general packet radio service) network are used to validate the analysis of the most efficient scheme.
Finally, a motion adaptive system for deriving MPEG-4 video packet sizes is presented. Further
simulations demonstrate the benefits of the adaptive system
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Effective video multicast over wireless internet
With the rapid growth of wireless networks and great success of Internet video, wireless video services are expected to be widely deployed in the near future. As different types of wireless networks are converging into all IP networks, i.e., the Internet, it is important to study video delivery over the wireless Internet. This paper proposes a novel end-system based adaptation protocol calledWireless Hybrid Adaptation Layered Multicast (WHALM) protocol for layered video multicast over wireless Internet. In WHALM the sender dynamically collects bandwidth distribution from the receivers and uses an optimal layer rate allocation mechanism to reduce the mismatches between the coarse-grained layer subscription levels and the heterogeneous and dynamic rate requirements from the receivers, thus maximizing the degree of satisfaction of all the receivers in a multicast session. Based on sampling theory and theory of probability, we reduce the required number of bandwidth feedbacks to a reasonable degree and use a scalable feedback mechanism to control the feedback process practically. WHALM is also tuned to perform well in wireless networks by integrating an end-to-end loss differentiation algorithm (LDA) to differentiate error losses from congestion losses at the receiver side. With a series of simulation experiments over NS platform, WHALM has been proved to be able to greatly improve the degree of satisfaction of all the receivers while avoiding congestion collapse on the wireless Internet
The QUIC Fix for Optimal Video Streaming
Within a few years of its introduction, QUIC has gained traction: a
significant chunk of traffic is now delivered over QUIC. The networking
community is actively engaged in debating the fairness, performance, and
applicability of QUIC for various use cases, but these debates are centered
around a narrow, common theme: how does the new reliable transport built on top
of UDP fare in different scenarios? Support for unreliable delivery in QUIC
remains largely unexplored.
The option for delivering content unreliably, as in a best-effort model,
deserves the QUIC designers' and community's attention. We propose extending
QUIC to support unreliable streams and present a simple approach for
implementation. We discuss a simple use case of video streaming---an
application that dominates the overall Internet traffic---that can leverage the
unreliable streams and potentially bring immense benefits to network operators
and content providers. To this end, we present a prototype implementation that,
by using both the reliable and unreliable streams in QUIC, outperforms both TCP
and QUIC in our evaluations.Comment: Published to ACM CoNEXT Workshop on the Evolution, Performance, and
Interoperability of QUIC (EPIQ
Performance evaluation of MPEG-4 video streaming over UMTS networks using an integrated tool environment
Universal Mobile Telecommunications System (UMTS) is a third-generation mobile communications system that supports wireless wideband multimedia applications. This paper investigates the video quality attained in streaming MPEG-4 video over UMTS networks using an integrated tool environment, which comprises an MPEG-4 encoder/decoder, a network simulator and video quality evaluation tools. The benefit of such an integrated tool environment is that it allows the evaluation of real video sources compressed using an MPEG-4 encoder. Simulation results show that UMTS Radio Link Control (RLC) outperforms the unacknowledged mode. The latter mode provides timely delivery but no error recovery. The acknowledged mode can deliver excellent perceived video quality for RLC block error rates up to 30% utilizing a playback buffer at the streaming client. Based on the analysis of the performance results, a self-adaptive RLC acknowledged mode protocol is proposed
Congestion Control using FEC for Conversational Multimedia Communication
In this paper, we propose a new rate control algorithm for conversational
multimedia flows. In our approach, along with Real-time Transport Protocol
(RTP) media packets, we propose sending redundant packets to probe for
available bandwidth. These redundant packets are Forward Error Correction (FEC)
encoded RTP packets. A straightforward interpretation is that if no losses
occur, the sender can increase the sending rate to include the FEC bit rate,
and in the case of losses due to congestion the redundant packets help in
recovering the lost packets. We also show that by varying the FEC bit rate, the
sender is able to conservatively or aggressively probe for available bandwidth.
We evaluate our FEC-based Rate Adaptation (FBRA) algorithm in a network
simulator and in the real-world and compare it to other congestion control
algorithms
Fuzzy Logic Control of Adaptive ARQ for Video Distribution over a Bluetooth Wireless Link
Bluetooth's default automatic repeat request (ARQ) scheme is not suited to video distribution resulting in missed display and decoded deadlines. Adaptive ARQ with active discard of expired packets from the send buffer is an alternative approach. However, even with the addition of cross-layer adaptation to picture-type packet importance, ARQ is not ideal in conditions of a deteriorating RF channel. The paper presents fuzzy logic control of ARQ, based on send buffer fullness and the head-of-line packet's deadline. The advantage of the fuzzy logic approach, which also scales its output according to picture type importance, is that the impact of delay can be directly introduced to the model, causing retransmissions to be reduced compared to all other schemes. The scheme considers both the delay constraints of the video stream and at the same time avoids send buffer overflow. Tests explore a variety of Bluetooth send buffer sizes and channel conditions. For adverse channel conditions and buffer size, the tests show an improvement of at least 4 dB in video quality compared to nonfuzzy schemes. The scheme can be applied to any codec with I-, P-, and (possibly) B-slices by inspection of packet headers without the need for encoder intervention.</jats:p
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