EE 368C Project Multi-stream Audio Transmission with Path Diversity

Slides:



Advertisements
Similar presentations
Streaming Video over the Internet
Advertisements

Playback-buffer Equalization For Streaming Media Using Stateless Transport Prioritization By Wai-tian Tan, Weidong Cui and John G. Apostolopoulos Presented.
Receiver-driven Layered Multicast S. McCanne, V. Jacobsen and M. Vetterli University of Calif, Berkeley and Lawrence Berkeley National Laboratory SIGCOMM.
1 Improving VoIP Transfer Rate over Internet Syed Misbahuddin, Dr. Engg. Department of Computer Science and Software Engineering University of Hail, Saudi.
Yi Liang Department of Electrical Engineering Stanford University April 19, 2000 Loss Recovery and Adaptive Playout Control for Packet Voice Communications.
Chapter 6 outline r 6.1 Multimedia Networking Applications r 6.2 Streaming stored audio and video m RTSP r 6.3 Real-time, Interactive Multimedia: Internet.
Measurements of Congestion Responsiveness of Windows Streaming Media (WSM) Presented By:- Ashish Gupta.
June 3, A New Multipath Routing Protocol for Ad Hoc Wireless Networks Amit Gupta and Amit Vyas.
Bernd Girod. Joint Source-Network Coding for Real-time Media 1 Joint Source-Network Coding for Real-time Media Bernd Girod Information Systems Laboratory.
Rate Distortion Optimized Streaming Maryam Hamidirad CMPT 820 Simon Fraser Univerity 1.
ACM Multimedia October 4, 2001 Real-time Voice Communication over the Internet Using Packet Path Diversity Yi Liang, Eckehard Steinbach, and Bernd Girod.
A New Adaptive FEC Loss Control Algorithm for Voice Over IP Applications Chinmay Padhye, Kenneth Christensen and Wilfirdo Moreno Department of Computer.
Distributed Video Streaming Over Internet Thinh PQ Nguyen and Avideh Zakhor Berkeley, CA, USA Presented By Sam.
Yi Liang Multi-stream Voice Communication with Path Diversity.
Adaptive Playout Scheduling Using Time-scale Modification Yi Liang, Nikolaus Färber Bernd Girod, Balaji Prabhakar.
Traffic Sensitive Active Queue Management - Mark Claypool, Robert Kinicki, Abhishek Kumar Dept. of Computer Science Worcester Polytechnic Institute Presenter.
Yi Liang Aug. 2, 01 Low-latency Internet media streaming using packet path diversity - prior and future work.
Adaptive Delay Aware Error Control for Internet telephony Catherine Boutremans Jean-Yves Le Boudec IP Telephony Workshop’2001 Institute for computer Communication.
1 Emulating AQM from End Hosts Presenters: Syed Zaidi Ivor Rodrigues.
Oct. 18, 2000 Subjective Tests Results Yi Liang Degradation Category Rating of Scaled Speech Three short network traces with different jitter statistics.
Nov. 3, 2000 Adaptive Playout Scheduling in Packet Voice Communications.
Influence of File Size Distribution on Legacy LAN QoS Parameters Nikolaus Färber Nov. 15, 2000.
Adaptive Playout Scheduling Using Time- scale Modification in Packet Voice Communications Yi J. Liang, Nikolaus Farber, Bernd Girod Information Systems.
CS640: Introduction to Computer Networks
SHEAU-RU TONG Management Information System Dept., National Pingtung University of Science and Technology, Taiwan (R.O.C.) YUAN-TSE.
Computer Networks: Multimedia Applications Ivan Marsic Rutgers University Chapter 3 – Multimedia & Real-time Applications.
1 Multimedia Communication Multimedia Systems(Module 5 Lesson 2) Summary: r Internet Phone Example m Making the Best use of Internet’s Best-Effort Service.
Network Instruments VoIP Analysis. VoIP Basics  What is VoIP?  Packetized voice traffic sent over an IP network  Competes with other traffic on the.
1 Measuring Congestion Responsiveness of Windows Streaming Media James Nichols Advisors: Prof. Mark Claypool Prof. Bob Kinicki Reader: Prof. David Finkel.
Wireless communications and mobile computing conference, p.p , July 2011.
Ch 6. Multimedia Networking Myungchul Kim
NUS.SOC.CS Roger Zimmermann (based in part on slides by Ooi Wei Tsang) Rate Adaptations.
Flow Control in Multimedia Communication Multimedia Systems and Standards S2 IF Telkom University.
L Subramanian*, I Stoica*, H Balakrishnan +, R Katz* *UC Berkeley, MIT + USENIX NSDI’04, 2004 Presented by Alok Rakkhit, Ionut Trestian.
A Comparison of RaDiO and CoDiO over IEEE WLANs May 25 th Jeonghun Noh Deepesh Jain A Comparison of RaDiO and CoDiO over IEEE WLANs.
Path Diversity for Media Streaming The Use of Multiple Description Coding J. Apostolopoulos, M. Trott and W. Tan Presented by Xiaoyuan GUO.
OverQos: An Overlay based Architecture for Enhancing Internet Qos L Subramanian*, I Stoica*, H Balakrishnan +, R Katz* *UC Berkeley, MIT + USENIX NSDI’04,
NUS.SOC.CS Roger Zimmermann (based in part on slides by Ooi Wei Tsang) Rate Adaptations.
Congestion Control in Data Networks and Internets
Accelerating Peer-to-Peer Networks for Video Streaming
Chapter 7 Multimedia Networking
The Transport Layer Congestion Control & UDP
Instructor Materials Chapter 6: Quality of Service
Neha Jain Shashwat Yadav
Empirically Characterizing the Buffer Behaviour of Real Devices
HCF and EDCF Simulations
MDC METHOD FOR HDTV TRANSMISSION OVER EXISTING IP NETWORK
Traffic Engineering with AIMD in MPLS Networks
Rate Adaptations.
Rohit Kapoor, Ling-Jyh Chen, M. Y. Sanadidi, Mario Gerla
Introduction to Networking
Khiem Lam Jimmy Vuong Andrew Yang
Congestion Control, Internet transport protocols: udp
© 2008 Cisco Systems, Inc. All rights reserved.Cisco ConfidentialPresentation_ID 1 Chapter 6: Quality of Service Connecting Networks.
Multimedia networking: outline
CSE679: Multimedia and Networking
CprE 458/558: Real-Time Systems
A New Multipath Routing Protocol for Ad Hoc Wireless Networks
Packet loss concealment using audio morphing
EE 122: Quality of Service and Resource Allocation
Streaming MPEG video over wireless link
AN ANALYTICAL MODEL OF MPEG-4 MULTIPLE DESCRIPTION VIDEO SOURCES
Performance Evaluation of Computer Networks
Multimedia networking: outline
Performance Evaluation of Computer Networks
Lightweight but Powerful QoS Solution for Embedded Systems
Reinhard Laroy BIPT European Parliament - 27 February 2012
Quality-of-Service ECE1545.
Project proposal Multi-stream and multi-path audio transmission
Modeling and Evaluating Variable Bit rate Video Steaming for ax
Presentation transcript:

EE 368C Project Multi-stream Audio Transmission with Path Diversity

The Incentives Best-effort services vs. strict QoS requirements of real-time speech communication, e.g. latency, loss, delay variation etc. Low data rate of the voice stream Why alternative path/multi-path? Exists a superior alt. path in 30-80% cases [Savage, 99’] Path diversity – network behavior averaged; burst loss converted to isolated loss; outage probability decreased [Apostolopoulos, 01’] Multi-path – independent jitter behavior Realization: explicitly path selection using relay servers D 1 Relay Relay 2 Data traffic Data traffic Voice traffic S The bottleneck

Outline Background on media playout scheduling algorithms Adaptive playout for multiple streams Measurements over the Internet and results Ns simulation and results Performance analysis of multi-stream transmission

Playout Scheduling Algorithms Tradeoff between delay and loss, wish to reduce both The adaptive playout scheme for single stream – jitter adaptation Fixed playout deadline late loss Adaptive playout Audio processing, refer to 6th ref in proposal.

Multi-stream Audio Playout Can always take the packet with lower delay Adaptive playout and speech scaling make seamless switching between streams possible; question: setting playout schedule? Multiple description coding; question: audio quality?

Determine the Playout Schedule To minimize the Lagrange cost function C = delay + 1  p (packet from both streams lost) + 2  p (one packet lost)  (audio quality degrad. when losing one packet) = d + 1  lp1  lp2 + 2  [lp1 (1- lp2)+lp2 (1- lp1)]  SNRdegrad Maintaining history for both streams; loss probability determined by delay and past history (order statistics) Freq d Greater 1 results in lower loss rate at the cost of higher delay Greater 2 results in both lower loss rate and better audio quality, at the cost of higher delay Increasing 1 without big 2 leads to lower loss rate, but not necessarily better sound quality Small 1 and 2 result in low delay lp1 Stream 1 ds1 lp2 Stream 2 ds2

MDC over Multiple Streams Multi-stream with MDC Stream 1: Even samples: quantized in finer resolution (8-bit) Odd samples: quantized in coarser resolution (4-bit) Stream 2: the other way [Jiang, 00’] SNR degradation s1 E O E O E O E O s2 O E  O E O E Packets in multi-streams Multiple description coding (MDC): generates multiple descriptions of equal importance for the same source signal Stream 1   Stream 2 SNRdegrad (dB) -21 -27 (after conceal.)

Comparison: Single-stream with FEC Stream sent FEC protected single-stream For fair comparison Primary copy: quantized in finer resolution (8-bit) Secondary copy quantized in coarser resolution (4-bit) SNR degradation Same data rate as multi-stream MDC 1 2 1 3 2 3 4 Stream received with packet loss  1 2 1 3 3 4 Stream reconstructed 1 2 3 4 Packets protected with FEC Primary   Second - SNRdegrad (dB) -24 -27 (after conceal.) FEC: adds redundancy by sending multiple copies of the source signal in the following packet(s) [Bolot, 96’]

Experiments over the Internet (I) (45) (5) (5) MIT 18.184.0.50 BBN Planet Exodus Comm. Netergy networks 192.84.16.176 (40) (5) Qwest Harvard 140.247.62.110 Path 1 (direct): Netergy – MIT Path 2 (alternative): Netergy – Harvard – MIT Direct path: 30ms UDP packets sent from source to dest.; routes determined by routing algorithm Alt. path: packets sent from source to relay server, then forwarded to dest.

Results of Experiment I (1) Delay – loss curve obtained by varying 1 while keeping 2 small and fixed. Mean delays (ms): 72.4/60.3 Link loss rate: 0.02%/0.85% Observed significant reduction in delay and loss rate by using multiple streams. Total/burst loss rate greatly reduced since jitter averaged. Burst loss: no matter caused by link or high jitter, is isolated. Path 1 (direct): Netergy – MIT Path 2 (alt): Netergy – Harvard - MIT

Results of Experiment I (2) Results obtained by varying 2 while keeping 1 fixed With higher delay: better chances to play both descriptions Observed lower playout rate variation by using multiple streams Jitter averaged; lower STD of min(di , dj)

Results of Experiment I (3) Playout of packets from multiple streams

Experiments over the Internet (II) Harvard 140.247.62.110 (10) Germany 131.188.130.136 (7) VBNS IP Backbone Service (40) DANTE Operations (5) (5) UUNET Tech. AT&T New Jersey 165.230.227.81 Path 1 (direct): N. J. – Germany Path 2 (alternative): N. J. – Harvard – Germany

Results of Experiment II (1) Mean delays (ms): 61.3/65.0 Link loss rate: 0.6%/1.1% Burst loss rate can still be reduced by more than 3%, since jitter averaged. Path 1 (direct): N. J. – Germany Path 2 (alternative): N. J. – Harvard – Germany

Results of Experiment II (2)

More Comments on Our Experiments Measurements by us: Transmission and receiving over multiple paths made at the same time One-way delay; UDP packets; 20 or 30 ms sampling rate Can observe jitter behavior for different paths Collected data valuable for further study of streaming multimedia Measurements by [Savage, 99’]: Not all data collected at the same time; conclusions made on averaged large aggregate of data Round trip time; ICMP packets; sampling frequency not high enough Not able to observe jitter behavior for streaming multimedia study

Simulations Using Network Simulator Ns: packet by packet event driven simulator Simulation parameters Link BW: 10Mbps Switch buffer: 100k byte/port Prop. delay on each link: 20ms TCP window size: 16k byte N: # of data sources attached to each intermediate hop Load: amount of traffic sent by each host Voice traffic model: CBR 64kbps Data traffic model: log normal, based on “Workload Characterization of the 1998 World Cup Web Site” [Arlitt, 99] D H3 H6 H2 H5 N H1 H4 S The simulation topology

Loss Reduction Losses of each path increase as network load goes up Multiple paths: loss rate reduced; burst loss isolated Multiple paths: loss/burst loss rate increases more gracefully as traffic load goes up Link loss rate vs. N

Delay Reduction =38 =26 Average delay reduction from stream i defined as: E[di - min(di , dj)] Most gain from multi-path when prop. delays are close Delay reduction increases as delay STD goes up

Conclusions and Future Work Multiple streams with path diversity: Reduces loss rate – due to averaged jitter, isolated burst loss and outage period Reduced delay by taking packet with lower delay from multiple streams Smoothed delay variation MDC works well with multi-stream adaptive playout, and makes audio quality scalable Performance gain affected by prop. delay and delay STD Multi-stream transmission can be realized by future peer-to-peer frame Applying the scheme onto more loss-sensitive applications, such as streaming video Improving topology and traffic model in ns simulation

Coupling of Multiple Paths D This topology studies using two paths that are not completely independent One link is shared by two streams, over which loss/delay behavior is not independent H7 H3 H6 H2 H5 N H1 H4 S Simulation topology B

Losses over Coupled Paths Topology A Topology B