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Realizing High Quality Digital Video Over Internet July 16 th 2008 TEIN2 NOC Workshop Kazunori Sugiura (

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Presentation on theme: "Realizing High Quality Digital Video Over Internet July 16 th 2008 TEIN2 NOC Workshop Kazunori Sugiura ("— Presentation transcript:

1 Realizing High Quality Digital Video Over Internet July 16 th 2008 TEIN2 NOC Workshop Kazunori Sugiura (

2 Digitized Video Transport System How can we transport live multi-media contents?

3 MPEG-2 H.264 DV Raw-HD WAV MP3 WMV RM MOV TXT Meta-Data Analog Ogg HardwareMedia Format

4 New Generation Internet: GLOBAL COMPUTING ENVIRONMENT To fulfill the IT requirements of every social and living activities in Asia, This project aims new generation Internet infrastructure by inter-connecting every activities. Global Computing Environment in Asia Core Technology for New Asia Internet Digital Broadcast Contents on Demand Metro, Regional Transportation Aerospace and Satellite Sea, and Air Transportation Cars, Bikes Social Services Houses, Apartments, Buildings Education Society Digital Media and ContentsLife style and LivingsSocial and Public Activities

5 Once upon a time

6 Streaming ?! 1996 Internet World Exposition


8 Rally Raid Mongolia 1996 Internet World Expo

9 Frame rate and Resolution fps (frames per second) – NTSC 30fps (29.97fps) – PAL 25fps – Films 24fps Resolutions ( dot resolutions) 1 Second 30picts. 30fps 1 Second 30 Picts 30fps

10 Quality and Contents By means of media contents – Quality ( SD to QHD ) – Bandwidth ( 1.5M to 4.5G up) – Encoding Characteristics ( High Computation, Delays…)

11 802.11n 100BaseT 1000BT HDDVD Blu-ray Digital BS We are here DV World (720x480) HDV World (MPEG2 720p 1080i) HD World (MPEG2 H p) Real HD World Uncompressed (1920 x 1080 p) Digital Cinema (MJ 4096x2160, 2048x1080) QHD (3840x2160) 16HD(SHD) (7680x4320) 64HD(VSHD) (15360x8640) IPTV IPv4/IPv6 Embedded Multicast FEC TCP Friendly DRM DVD We are here 10G NGN g 10G Stack

12 Resolution and Bandwidth ResolutionPixel Dot 32Bit 1Frame 30fps byte Bandwidth bps 640x480307, K35.16M281M 800x600480, K54.93M439M 1024x768786,4323M90M720M 1280x10241,310,7205M150M1200M 1600x12001,920, M219.6M1.72G 1920x10802,073, M237.3M1.85G 1920x12002,304, M263.4M2.06G 3960x24009,504, M1087.5M8.50G

13 Summary Media streaming over Internet is a challenge – Bandwidth – Computational power – Resolution and video quality Advances in media compression technique Advances in packet transport technology

14 Compression Technique

15 Frame based Compression Method – Motion JPEG – DV Inter Frame Compression Method – MPEG2 – MPEG4 – H.264

16 JPEG/Motion JPEG JPEG – Lossy data compression picture format – DCT(Discrete Cosine Transform) algorithm Cutting High frequency Motion JPEG – Combining JPEG pictures to make frame based video

17 Compress even more Using higher compression method – Motion based lossy compression No movement – background Movement – Target Using the recently used Inter Frame compression

18 MPEG Video CODEC – MPEG1 – MPEG2 – (MPEG3 is gone ) – MPEG4 (H.264) Data standard – MPEG7 Data handling – MPEG21

19 MPEG1 Specifications – Bandwidth: 1.5Mbps – Standards for Video CD MPEG1 uses SIF Format – Non Interlace – 360×240×29.97Hz(NTSC)

20 GOP(Group Of Picture) structure GOP – Frame combination – I picture (Intra picture) – P picture (Predictive picture) – B picture (Bi-directional predictive picture) IP BBBB P BB P BB P BB 15Picture/1GOP I PictireB PictireP Pictire Forward prediction Bi-directional prediction

21 MPEG2 Higher quality encoding compared to MPEG1 – Broadcast Quality Examples – DVD – HDV – Digital Broadcast

22 Resolutions for MPEG2 1920×1152 ×60 4:2:0 1920×1152 ×60 4:2:2 1440×1152 ×60 4:2:0 1440×1152 ×60 4:2:0 1440×1152 ×60 4:2:2 720×576×30 4:2:0 B 720×576×30 4:2:0 720×576×30 4:2:0 720×576×30 4:2:2 352×288×30 4:2:0 352×288×30 4:2:0

23 MPEG4 Higher Compression method than MPEG – Object based encoding For low bandwidth MPEG-4 AVC (Part 10 ) H.264

24 DV DV(Digital Video) Mbps – IEC61834 (1999) Resolution 720x480(NTSC) Mbps Audio 1.536Mbps – 48kHz/16bit 2 channel – 32kHz/12bit 4 channel – Frame Compression using DCT DV Cassette mini-DV Cassette

25 HDV Canon, Sharp, Sony, Victor (2003) Resolution – 1280x720 (720p) 19.7Mbps – 1440x1080 (1080i) 25Mbps 1080/25p 1080/30p 1080/24p – Audio 48kHz/16bit 2 channel MPEG1 Audio Layer2 (384Kbps) – MPEG2 Inter Frame Compression – DV, mini-DV

26 DVCPRO Panasonic (1996) – DVCPRO 480i(NTSC) 25Mbps – DVCPRO i(NTSC) 50Mbps – DVCPRO P 480p(EDTV-II) 50Mbps – DVCPRO HD 1080i/720p 100MBps – Interoperability with DV Format DVCPRO Cassette

27 What is Streaming? Streaming – Receive Data while playing – Does not store data as a file ( DRM (Digital Rights Management) perspective Download Playback – Download contents as a file, and playback afterwards Download Playback Streaming Internet

28 Streaming Technology To absorb characteristics of the Internet Some of the software technologies – Buffering ( Delays) – Session Management – Multiple Bit-rate ( Multiple Quality) – Bandwidth reservation, QoS – Error Correction – Contents Protection

29 History in Streaming 1994 – StreamWorks1.0(First Commercial streaming applications) – VIC, VAT, RAT developed by UCL (University College London) 1995 – Real Audio – Internet World Exposition IWE96 – RTP(RFC1889) 1997 – MS NetShow2.0

30 History in Streaming (contd.) 1999 – QuickTime4.0( ) – VLC (Video Lan Client) – DVTS 2002 – Helix( Open source ) 2003 – Windows Media 9

31 Streaming Theory

32 TCP and streaming TCP – Retransmission method – Packet receiving order guarantee – Congestion Management Using TCP for streaming – How can we manage the real-time when Retransmission occurs? Congestion Management happens?

33 UDP Streaming UDP – No Retransmission – No Packet order guarantee – No Congestion Management To maintain some sort of packet transport guarantee to End-toEnd – RTP/RTCP

34 RTP/RTCP RTP Real-time Transport Protocol RTCP Real-time Transport Control Protocol Standardized in RFC1890 Standardization to transport real-time data through the network

35 RTP Real-time Transport Protocol RFC1889 Standard protocol for streaming Common information required to send real-time data –Sequence Number –Timestamp Dedicate RFC for data dependent part –Many payload format dependent on their compression method and media formats RTP itself does not reserve resources or manage and guarantee the QoS

36 RTP Packet Marker pv X Extension CC CSRC Count Payload Type Sequence Number Time Stamp Synchronization Source (SSRC) Contribution Source (CSRC)

37 RTP related RFCs RTP Basic Specifications RFC1889RTP: A Transport Protocol for Real-Time Applications. RFC1890RTP Profile for Audio and Video Conferences with Minimal Control. Other Payload Specifications RFC2198RTP Payload for Redundant Audio Data. RFC2793RTP Payload for Text Conversation. RFC2833RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals. RTP Generals RFC2508Compressing IP/UDP/RTP Headers for Low-Speed Serial Links.

38 RTP Payload Format Payload Format RFC2029Sun's CellB Video Encoding. RFC2032H.261 Video Streams. RFC2035JPEG-compressed Video. RFC2038MPEG1/MPEG2 Video. RFC2190H.263 Video Streams. RFC2250MPEG1/MPEG2 Video. RFC2343Bundled MPEG EXPERIMENTAL RFC2429the 1998 Version of ITU-T Rec. H.263 Video (H.263+). RFC2431BT.656 Video Encoding. RFC2435JPEG-compressed Video. RFC2658PureVoice(tm) Audio. RFC2862Real-Time Pointers. RFC3016MPEG-4 Audio/Visual Streams. RFC3047ITU-T Recommendation G

39 RTP Payload Type Payload TypeEncodingAudio / VideoFrequency (Hz) Channel ( ) 0PCMUA A G.721A GSMA DV14A DV14A LPCA PCMAA G.722A L16A L16A MPAA G.728A CelBV NvV H.261V90000

40 RTCP(Realtime Control Protocol) RFC1889 – To Control RTP Packet flow control, clock transmission between sender and receiver – Quality report by receiver Jitter Lost packet RTCP Sender Report – Reports to receiver or the 3 rd application for status report on sending conditions

41 RTCP Packet Report block comes afterwards Pv=2 Payload Type SR=200 Length Sender Synchronization Source SSRC NTP Timestamp Fore 32bit (Send time for reports) NTP Timestamp Lower 32bit RTP Timestamp Sender Packet count Sender Octet Count RC Header Sender Info

42 RTCP Method RTCP Receiver Report – Packet Loss rate, Sum of packet losses, Sequence number of received packet, jitter, last timestamp of sender report (LSR), and Delay between LSR (DLSR) BYE APP – Application Extension

43 RTCP Method SDES – Receiver information – CNAME Receiver identifier – NAME Receiver name – Mail address – PHONE Phone No. – LOC Location – TOOL Application name – NOTE User condition – PRIV Application extension

44 Summary Streaming application over Internet uses – UDP / IP – RTP – RTCP Ensure real-time isochronous transmission

45 DVTS Design

46 Video format used in IEEE1394 DV Format A.K.A DV Camera Frame base compression 30Mbps 720 x 480 Pixels (NTSC) 720 x 576 Pixels (PAL) HDV Format MPEG2-TS(PS) compressed High Definition DV Inter Frame compression ( = Latency ) Somewhat different from HDTV – 1080i (1440 x 1080)25Mbps, (1920 x 1080 ) – 720p (1280 x 720) 19Mbps Less Bandwidth compared to DV 30 > 25

47 Simple Mechanism Packet Losses What happens when packet losses occur: – Reuse the past frame data Present frame Packet losses Past frame Past frame data Is used Reused frame

48 Frame rate reduction and Bandwidth in DV Format Bandwidth (Mbps) Frame Rate

49 Network Friendly streaming Packet shaping and burst-less transmission time Packets


51 Packet Flow Control

52 Drop Gateway / Router Congestion mechanism

53 Application End application End application based packet shaping (Avoiding burst traffic) Large packet buffer pool

54 Jitter caused by priority mismatch Application Audio packet Video packet Waiting for audio packet Sender host

55 Separation of packet Application Audio Packet Video packet Video queue Audio queue Priority audio packet Sender host

56 TCP sender UDP sender congestion packets TCP packet UDP packet TCP UDP Traffic

57 TCP sender UDP sender congestion packets TCP packet UDP packet TCP UDP Traffic

58 TCP sender UDP sender packets Decrease packets to avoid congestion TCP packet UDP packet TCP UDP Traffic

59 TCP sender UDP sender packets Decrease packets to avoid congestion TCP packet UDP packet Rate will not change (No congestion avoidance) TCP UDP Traffic

60 TCP sender UDP sender packets Decrease packets to avoid congestion Rate will not change (No congestion avoidance) TCP packet UDP packet TCP UDP Traffic

61 summary DVTS is developed with software technology to – Adapt into various condition of network – Try to be friendly with other packet / networks Avoiding bursty packets Using TCP Friendly algorithms Using past packet / loss of packets

62 DVTS Update

63 DVTS supports HDV streaming Support Video Format – MPEG2-TS 1080i and 720p Support Operating-System – Linux (kernel or later and 2.6.x) Newest Libraw1394 driver is required. – Windows XP – VLC TransmitIEEE1394 Output Display Output Recording DVTS for Linux DVTS for Windows Video LAN Client

64 DVTSng Rev.2 Merge DVTS and HiDVTS applications on one package – DV -> use DV mode application – JVC HDV -> use HDV-JVC mode application – Sony HDV -> use HDV-SONY mode application IPv6 multicast (ASM/SSM) update – Rev.1 cannot use IPv6 multicast function getaddrinfo() doesnt run -> re-enable Download URL – URL alias OK other device will be supported on next revision

65 DV Mode Application HDV Mode Application Support IEEE1394 output Not support IEEE1394 output Please use HDVout tool instead!!

66 HiDVTS / Camera Output Tool Functions – HDV(MPEG2) RTP data receive IPv4, IPv6 receive port unicast, multicast (ASM/SSM) – IEEE1394(HDV device) output only support SONY device (maybe) Download URL –

67 Start/Stop running Quit application Select IP version Select HDV device (camera/VCR) If you want to use multicast, Specify multicast address, interface and source address (SSM)

68 DVTS supports Multicast subsets (including IGMPv3 and MLDv2) Linux kernel 2.6.x supports v4/v6 SSM Windows – XP supports only v4 SSM(IGMPv3) – Vista supports v4/v6 SSM(IGMPv3,MLDv2) IGMPv3MLDv2 DVTS for Linux DVTS for Vista DVTS for Windows XP Video LAN Client

69 MacOS Update gldvshow was released – No Audio Support – FEC Support – OpenGL error monitor Support DVTS1.0g was released – Support MacOS 10.5 Leopard

70 cbrmaker(1/3) Constant Bit Rate UDP stream making stream in Application Layer – You can check Bandwidth Transmission delay Packet arrival interval time Packet loss You can get – Support platform – Linux i386 only

71 cbrmaker(2/3) You can draw graphs from logs of cbrmaker – Transmission delay – Packet arrival interval time – Packet loss

72 cbrmaker(3/3) Parameter – bps (bit per second) – pps(packet per second) – Packet size – usleep time between packets Example of the utility – OS / network driver test – Network bandwidth test – Network quality test

73 Global Studio

74 Global Studio Location(2008) Stanford U. US NEW(June,2007)

75 Changes on existing studio Echo canceller installation – To provide better audio environment – Schedule Cambridge(September 07) Yonsei(March 08) Tsinguha(March 08) Replacing video transmission PCs – To reduce audio noise – More CPU power(for HD level video transmission) – Schedule Cambridge(September 07) Updating equipments at Mita campus – Improving audio and video quality Digital Audio for noiseless audio environment HD video – Collaboration with SOI-Asia

76 The Stanford Studio Installed in June 2007 – Collaboration with Stanford university. Location – The Wallenberg Hall Mobile Equipments – 1 st floor learning theater or other conference room

77 Mobile cabinet(inside) Polycom EF2241 Audio echo canceller Cisco Catalyst 2960 Audio Patch panel(All XLR) DVTS PCs(Laptop PC is used after 2007) Mobile cabinet(outside) Handle to carry the cabinet Removable doors Mobile Equipment cabinet Wheels

78 DMC Mita Location: Tokyo, Japan Operated by: Keio University DMC Institute DVTS and Polycom / Multipoint capable IPv4/IPv6

79 Studio at Tsinghua University Location: Beijing, China Operated by: Tsinghua University Time difference: 1 hour

80 Studio at Yonsei University Location: Seoul, Korea Operated by: Yonsei University Connecting to KOREN (Korea Advanced Research Network) Established in February 2006

81 Studio in San Francisco Location: California, USA 15 minutes drive from SFO Airport in Data Center Stanford University Time difference: 17 hours


83 Studio at the University of Cambridge Location: Cambridge, UK Operated by: University of Cambridge Time-zone: -9 hour (JST)

84 Studio in New York Location: New York, USA near United Nations HQ Operated by: Japan Society Time difference: 13 hours


86 Tripod WIDE Conversion Lenz Accessory Kit Video Light Headphone DVTS/HDVTS S L Systems IEEE1394 Long Cable MIC Audio Mixer DVTS/HDVTS S L Systems Projector WIN-XP HD Note-PC ADVC-110 Polycom VSX 7000S Network Camera(SD) Ethernet Switch 8 to 16 port Giga Ethernet RGB / DVI IEEE1394 S-Video Audio Version 1.0 uhyo UPLINK AC TAP Configuration Plan for Cambridge Studio Panel Monitor Sony HVR-Z1J Behringer MX B-U Converter ADVC-110 Behringer FBQ2496 DVTS/HDVTS S L Systems POSTERS

87 DMC Manifesto

88 Mozilla hour World Wide continuous event – Connects Mozilla community member on the earth for collaboration. Different Time zone Distributed location DMC provide the Global Studio as communication environments. – DVTS based conference – Real video streaming Date & Location – September 15 th -16 th – Japan/US/France – SOI-ASIA Closing session at Mita Discussion at Paris

89 Cool Japan Support NHK TV program Cool Japan – In November 2007 Provide DVTS video conference environment – San Francisco, Cambridge, Seoul and Mita On Air – December 12 th – DVD is available Contact kudo Received video at MitaReceived video at Cambridge

90 Other Events The 6 th DMC symposium – Connect Beijing for Prof. Murai – December 2007 Keio/Kyoto MoU ceremony – Connect Mita and Kyoto – September 2007 etc


92 SIGCOMM2007 IPv4/v6 dual stack Multicast Live Streaming using Consumer HD Video

93 outline SIGCOMM2007 live streaming Inter-Domain Multicast Routing Status DVTS for HDV(MPEG2-TS HD) Next step to deploy IPv6 Multicast

94 SIGCOMM 2007 Live Streaming WIDE Project(AS2500) provided Internet connectivity for SIGCOMM2007 participates. We also provided 2 workshops and 3 conferences live streaming to world wide researchers who could receive. We prepared following streams. – IPv4 and IPv6 MPEG2-TS 1080i streaming to world wide researchers. – IPv6 MPEG4 streaming to TEIN2 and SOI-ASIA members.

95 Inter-Domain Multicast Routing Some Asian Academic AS have been enabled IDMR since SIGCOMM2007. – Tein2-SG and Tein2-HK has been connected to Tein2-JP. – Koren has been connected to APAN-JP. – Good content makes us promote IDMR. World Wide IDMR ring was constructed! – Most of Academic AS have multiple AS-Path.


97 KOREN TransPAC2 Abilene AARNET3 TEIN2-SG WIDE APAN-JP TEIN2-HK TEIN2-JP IPv4 Listener AS (Confirmed at SIGCOMM2007)


99 SIGCOMM2007 ASIA IPv6 Multicast Live Streaming 13 Mbps UDL Kyoto, Japan DVTS 30 Mbps AI3/SOI Japan 24 SOI Asia partner 12 Asia countries MPEG4 VideoLAN 1Mbps APAN-JP TEIN2-JP TEIN2-SG MPEG4 VideoLAN 1 and 3Mbps Uninet members INHERENT members RP

100 Backbone is ready! and then… How to deploy IPv6 multicast? Backbone is ready to provide IPv4 and IPv6 multicast. BUT! Users dont care of IP version, they want to receive good contents. We have to prepare the IPv4/v6 transparent multicast environment for the users. In addition, IPv6 has a priority to use IPv4:p

101 DVTS supports HDV streaming Support Video Format – MPEG2-TS 1080i and 720p Support Operating-System – Linux (kernel or later and 2.6.x) Newest Libraw1394 driver is required. – Windows XP – VLC TransmitIEEE1394 Output Display Output Recording DVTS for Linux DVTS for Windows Video LAN Client

102 Minimal Settings Just prepare 1 Camera and 2 PCs. – Optional: HDV Deck. IP Network IEEE1394 UDP/RTP IEEE1394 HDV Camera $2,000 - $10,000 HDV VCR

103 DVTS supports Multicast subsets (including IGMPv3 and MLDv2) Linux kernel 2.6.x supports v4/v6 SSM Windows – XP supports only v4 SSM(IGMPv3) – Vista supports v4/v6 SSM(IGMPv3,MLDv2) IGMPv3MLDv2 DVTS for Linux DVTS for Vista DVTS for Windows XP Video LAN Client

104 Component/1080i RGB-15pin Hall PA Canon Canon XL H1 Speakers Laptop Canon Roland VC300-HD Audio Mixer IEEE1394(DV) IEEE1394(HDV) Roland v-440 HD Component/1080i IPv4/v6 Multicast To Global Multicast Cloud DV RTP To SFC ADVC-110 IEEE1394(DV) CompositeIPv6 Multicast To Tein2 via AI3 SONY HVR-Z1J Component/1080i hdvsend for Linux dvsend for Linux SIGCOMM2007 Live Settings

105 AS2500 WIDE Connect from APAN-JP MBGP AS-Path Table from UNINETT NORWAY AS7660 APAN-JP Live Streaming Equipments

106 SIGCOMM2007 ASIA IPv6 Multicast Live Streaming


108 HD-SDI SD-SDI i-Visto media converter HDTV SDTV SDTV equipment HDTV equipment i-Visto gateway XG i-Visto camera i-Visto product lineup i-Visto eXmedia server IP network (LAN/WAN) HDTV monitor i-Visto manager i-Visto 1.5Gbps HDTV IP HD-SDI IP Gateway eXmedia server

109 Traffic(cont.) HTB(10GE) Incomming from Sapporo NW center (v6-multi) Sapporo NW center(10GE) Outgoing to KDDI Sapporo (v4-uni) KDDI Sapporo(1GE*2) Outgoing to KDDI Sendai (v4-uni)

110 Traffic(cont.) ABC(10GE) Incoming (v4-uni) Outgoing (v6-multi) NTT Otemachi(10GE) Forwarding ( v4-uni&v6-multi)

111 i-Visto(cont.) Packetize each line – Not each frame Configuration – Resolution 1080i or 760p – Color 8bit(1Gbps) or 10bit(1.6Gbps) Network interface: 2 * 1GE – Usually use 10GE NIC – Over subscribe 1GE*2 circuit – Segmentation for each 1GE Testing IPv6 Multicast send/receive at 1470byte(most suitable frame size) – Pre-Test – Packet loss occurred by lack of performance at receiving side

112 Layered Multicast

113 Layered multicast for adaptive modulation Problem statement – With WiMAX multicast, multicast data transmission rate depends on the link capacity of the node whose link quality is worst Nodes with good wireless connection cannot get high-quality multicast data Combination of layered multicast and adaptive modulation – Nodes come to be able to receive the multicast data with proper quality depends on their wireless connection quality Key Technolgies – Layered multicast Network resource aware multicast data transmission method – Multicast group management Dynamic IP multicast group management based on the nodes current physical layer modulation Mapping between an IP multicast group to a modulation based WiMAX multicast group

114 Layered Multicast Hierarchical data structure – Data are divided into several Layer Layer is multimedia data unit Video quality control selecting the number of accepting layers – Layer Base Layer – Base Multimedia data Enhanced Layer – Quality enhancement data Multicast tree structure – Data are deliverd with IP multicast – Senders provide maximum quality data – Intermediate nodes decrease layers to send adapting to the network resource 10/ /10 8 6/ /10 1 6/10 6 Base Enhanced 1 Enhanced 2 Enhanced 3 Enhanced 4 Enhanced n

115 WiMAX electric wave method BPSK ready QPSK ready 16QAM read 64QAM ready

116 Modulation based multicast group management image BS 64QAM + 16QAM + QPSK + BPSK zone 16QAM + QPSK + BPSK zone QPSK + BPSK zone BPSK zone

117 Modulation based multicast group management Signal type distance 64QAM (Enhanced 3) 16QAM (Enhanced 2) QPSK (Enhanced 1) BPSK (Base Layer) Base only Base + Enhanced 1 Base + Enhanced Base + Enhanced

118 Research-Slide

119 Research background: The growth of mixed environment Real-network e a/b/g 10G 40,100G virtual-network Service or user-tool NGN High quality Event relay IPTV Live VoIP Network Collaboration game Digital Cinema security authentication Resource management Priority control download End control End control User requirement AS User requirement Service characteristic Network characteristic MulticastUnicastP2P

120 Motivation Due to the use of high speed Network(1Gbps, 10Gbps, etc), high bandwidth (high video audio quality) streaming will become common on the internet What is a requirement for high-bandwidth streaming?, and impact on other applications? – Ex) What is the tradeoff point on user-oriented application? What is streaming application problem in nearly future ? What is the solution? HD Camera studio

121 Adaptive Real-time Streaming Real-time Streaming – High interactiveness Video conference Online game, 3D graphics contents Need to maintain strict packet interval time – Difficult to maintain best possible quality Network condition change effect on largely – Important to precisely detect a network condition change. Network associated system vs. End-to-End system – The former merit: easy to quickly and precisely detect network condition (ex. Bottleneck ling) possible to adjust according to it Priority control Bandwidth guarantee – The latter merit: not need for specific router or switch implementation Loss based (or RTT-based) adaptation Network Estimation Quality Adaptation Network ControllerEnd-Node controller

122 Research point (1/2) How can various streaming flows conduct best adaptation according to network condition and change?? – Network estimation Various network environment – Various network characteristic Various congestion – The number or type of competing flows – Network load – User-oriented quality adaptation High network utilization – Effectively consume the network resource Contents quality protection – Maintaining quality for end-usage Stability – Frequently quality change would degrade its quality – User-friendliness (impact on other applications) Scalability (Congestion Control) – TCP-fairness high network utilization » Various definition of fairness Network estimation Quality adaptation (Congestion Control) Network Condition

123 Research point (2/2) Quality adaptation – High network utilization Contents quality protection, Scalability The way of changing transmission rate The way of maintaining quality on changeable network condition TCP-friendly could be best solution ? Network estimation – Network indicator What is necessary and effectual? Need new indicator or indicator cooperation – Signaling mechanism Responsiveness accuracy network Media source client Congestion Control Quality protection Network indicator Application/service policy Transmission Protocol Video format Vide/Audio data scalability

124 Previous work Specializing in quality protection – High network utilization – The mechanism of minimizing packet loss Multiple control using FEC – Quality adaptation Rate Control (discarding frame, so depends on video format) Dynamic FEC (Reed-Solomon Code) – Network condition estimation Packet loss rate on a interval time The number of consecutive lost packets ( means a network load) FEC recovery rate Application policy Media source Congestion control Quality protection network client Packet loss rate The number of consecutive loss packets FEC recovery rate Rate Control Dynamic FEC

125 Future work Verification using a simulation – Definite the mechanism tradeoff point High-network utilization, quality protection, stability and scalability Network condition (bandwidth and RTT, etc), the number and type of TCP- flow and UDP-flow R – TFRC effectiveness ? How can it be change on network condition ? – My algorithm effectiveness? Investigate effective adaptation mechanism The mechanism of knowing network condition and change. – Survey – Verification of detection and learning adaptation mechanism

126 Research-Slide Support material

127 The problem of quality reduction According to network condition, pktloss happens – Physical bandwidth or available bandwidth Σ(DVTS traffic + other traffic ) > available bandwidth – Congestion – Quality reducing Sender Receiver Internet DV/RTP packet congestion bursty

128 The existing solution with DVTS(1/2) Reuse the past frame data Rate Control to save the consumption bandwidth Σ(DVTS traffic + other traffic ) bandwidth < available bandwidth

129 The existing solution with DVTS(1/2) Static FEC to packet loss recovery congestion FEC decode!! Packet loss recovery FEC technology DV/RTP packet FEC packet

130 The demerit of existing solutions In the condition of network congestion – Reuse the past frame data and Rate Control Solve the block noise Cannot solve the reducing video quality – Smoothness of frequently moving scene – Static FEC Increasing the total consumption bandwidth – There is the case of quality reducing more !! – Users adapt the transport method by hand What is How to provide the best possible Video and Play quality ???

131 Motivation Dynamic Adaptation – To provide the best possible quality – The user doesnt have to change the DVTS options – According to the network condition change, adaptive DVTS dynamically adapt the transport method DV full rate + FEC 10% ??? DV half rate + FEC 30% ??? DV 1/3 rate + FEC 20% ???

132 approach Rate Control with Dynamic FEC Mechanism – Frame rate control reducing bandwidth in case of fatal bandwidth conditions To save the total consumption bandwidth – Dynamically adapting FEC rate keeping play and video quality Dynamic available bandwidth proving via FEC – What happens when network condition get well ?? – Adapting Frame rate to current available bandwidth » Providing best possible video quality Video frame data Internet sender FEC data Changing each rate

133 Adaptive DVTS design Receive report module Flow control module Transmit Buffer Report transmit module Play Buffer RTP Packet (data, FEC) RTCP RR Packet Network state Network state Current flow type before flow type Network state measurement module pktloss pattern Flow type FEC decode module sender receiver FEC encode module Rate control module tmp buffer

134 Frec(%) = L L (L > L, L0) (L L, L0) 0 Frec: FEC recovery rate Fenc: FEC encoding rate L: (the number of non-recoverd UDP packets within 5 seconds) The number of consecutively lost packet and FEC recovery rate

135 Adaptive DVTS vs Normal DVTS BEST CASE Background bandwith:90M Anticipating available bandwidth

136 Streaming Adaptation – Adjusting Forward Error Correction with Quality Scaling for streaming MPEG, huahui NOSSDAV 2005 ACM, june – A Rate Control Scheme for Adaptive Real-Time Applications in IP Networks With Lossy Links and Long Round Trip Times – A Dynamic Congestion Control Mechanism for Real Time Streams over RTP, Ahmad, N, ICACT 2007, Feb – A Rate Control Scheme for Adaptive Video Streaming over the Internet, Panagiotis, IEEE ICC 2007 proceegings /streaming –End-to-End Internet Packet Dynamics, Vern Paxson LBNL , june23,1997 – Analysis of Audio Packet Loss in the Internet, INRIA B.P 93,Jean Bolot, HUgues Crepin, Andres Vega Garda Sophia-Antipolis Cedex – Improving Accuracy in End-to-End Packet loss Measurement, SIGCOMM2005

137 Network States and Flow Types in Full rate(30Mbps) StateLoss rate Consecutive loss 0%10%20%30% F10 0 FEC0% F20

138 State Loss rateConsecutive loss50%60%70%80%90%100% H10 *************FEC100% Normal-Full H20

139 G8 summit operation using FEC DVTS

140 Overview Encoder FEC DVTS Receiver FEC DVTS Sender converter FEC DVTS Sender FEC DVTS Receiver DV streaming with FEC 30% kanagawa Hokkaido Receiving each G8 national broadcast

141 Backbone Foundry6.otemachi Foundry4.nezu Nec2.yagami Cisco2.fujisawa SFC λ Net /28 Net /27 IMC Net /28 SNet / JGN2plusNTT Business Ether * 4 SFC ITC L2 Backbone Cisco2.notemachiT-LEX Te 3/4 Vlan Gi 7/13-16 Vlan Gi 2/1/ Mbps * 4 1Gbps * 1 1Gbps10Gbps L2 L3 100M 1G 10G

142 Hammers in kanagawa Cisco2.notemachi Cat3750-1Cat3750-2Cat2950-1Cat CNNIBBCDWTVTV-5RTRRAITVYOBI NHK World Gi7/13 Gi7/14Gi7/15 Gi7/16 Gi1/0/1Gi2/0/1Fa0/1 Untag vlan /27

143 Sapporo IMC catalyst3750.sapporo BBC- R tag vlan CNN I-R TV- 5-R DWT V-R RAI- TV-R RTR -R NHK- World -R YOBI 1 YOBI 2 YOBI 3 1/0/11/0/2 1/0/3 1/0/4 1/0/5 1/0/6 1/0/7 1/0/13 1/0/14 1/0/15 Cisco2.notemachi T-LEX Te 3/4 Vlan

144 SFC (backup stream) ax2430.sfc-ramda Cisco2.fujisawa TV5-SDWTV 0/22 0/23 Gi 2/1/

145 System overview in Sapporo NHK NHK WIDE NHK

146 WIDE Rack BBCCNN TV5 YOBI-2 YOBI-3 DW RAIRTR NHKYOBI-1 NHK (to ) WIDE (toJGN) NTSC DV (WIDE) PAL DV (NHK) port assign and IP address 1/0/1: :BBC-R 1/0/2: :CNNI-R 1/0/3: :TV-5-R 1/0/4: :DWTV-R 1/0/5: :RAI-TV-R 1/0/6: :RPR-R 1/0/7: :NHK-World-R 1/0/13: :YOBI-1 1/0/14: :YOBI-2 1/0/15: :YOBI-3

147 Monitor Preview in Sapporo

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