Professional Documents
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Report
Jean-Nol Gouyet, EBU International Training Revised and proof-read by the speakers
EBU Networks 2008 Seminar / 23 - 24 June 2008 Reproduction prohibited without written permission of the EBU Technical Department & EBU International Training
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Opening round table4
1 1.1 1.2 1.3 1.4 2 2.1 2.2 2.3 2.4 3 3.1 3.2 3.3 4 4.1 4.2 4.3 5 5.1 5.2 5.3 6 6.1 6.2 6.3
IP and MPLS ................................................................................................................ 6 Concepts and protocols ................................................................................................ 6 Traffic engineering ........................................................................................................ 7 Video over IP - Maintaining QoS ................................................................................... 8 Practical application ...................................................................................................... 9 Audio Contribution over IP ...................................................................................... 10 The ACIP standard - A tutorial .................................................................................... 10 Implementing and testing ACIP using open source software ...................................... 11 Interoperability in action .............................................................................................. 13 Demos ........................................................................................................................ 14 Video contribution over IP ....................................................................................... 15 Video contribution over IP The EBU project group N/VCIP ...................................... 15 Real-time transport of television using JPEG2000 ...................................................... 16 France 3's new contribution network over IP ............................................................... 17 HD over Networks ..................................................................................................... 18 HD contribution codecs ............................................................................................... 18 Practical thoughts and wishes or whishful thinking! ................................................. 19 Eurovision experience in HD contribution ................................................................... 20 Real world applications: the News .......................................................................... 21 The use of COFDM for ENG applications ................................................................... 21 Getting News to base, or 'Only connect'? ................................................................... 22 How to connect your Video Journalists (VJ) ................................................................ 22 Real world applications - Production and Broadcast............................................. 23 Architecting a fully-networked production environment ............................................... 23 Bringing the BBC's networks into the 21st century ....................................................... 25 DVB-H small gap fillers: home repeaters improving indoor coverage .......................... 26
List of abbreviations and acronyms..28 Table 1: Towards lossless video transport - Deployment scenarios.35 Table 2: Practical measurement and practical network performance...37 Table 3: HD contribution codecs.39 Table 4: The ovewhelming crowd of networks worldwide!......................................................................41
EBU Networks 2008 Seminar / 23 - 24 June 2008 Reproduction prohibited without written permission of the EBU Technical Department & EBU International Training
Notice
This report is intended to serve as a reminder of the presentations for those who came to the seminar, or as an introduction for those unable to be there. So, please feel free to forward this report to your colleagues! It is not a transcription of the lectures, but a summary of the main elements of the sessions. The tutorial-like presentations and tests results are more detailed. The speakers presentations are available on the following EBU FTP site via browser: ftp://uptraining:ft4train@ftp.ebu.ch The slides number [in brackets] refer to the slides of the corresponding presentation. To help "decode" the abbreviations and acronyms used in the presentations' slides or in this report, a list is provided at the end of this report. Web links are provided in the report for further reading.
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Many thanks to all the speakers and session chairmen who revised the report draft. Special thanks to Peter CalvertSmith (Siemens), Andrea Metz (IRT) and Martin Turner (BBC), who kindly allowed us to use their personal presentation notes ( 1.2 - 2.2 - 5.2). Nathalie Cordonnier, project manager, and Corinne Sancosme (EBU International Training) made the final revision and editing.
The Networks 2005 / 2006 / 2007 seminar reports are still available on the EBU Web site: http://www.ebu.ch/CMSimages/en/NMC2005report_FINAL_tcm6-40551.pdf http://www.ebu.ch/CMSimages/en/EBU_2006_Networks_Report_tcm6-45920.pdf http://www.ebu.ch/CMSimages/en/EBU-Networks2007-Report_tcm6-53489.pdf
cf. 2.3 cf. EBU Networks2006 seminar report 2.1and 2.2 and EBU Networks2005 seminar report 2.2 and 2.3
EBU Networks 2008 Seminar / 23 - 24 June 2008 Reproduction prohibited without written permission of the EBU Technical Department & EBU International Training
5 Evolution of the Internet o The Internet may change. Traffic shaping, for example, can affect us. There may be also some tariff on the
Internet - if one day I can't download as many Gigabytes as I want, it will change our way of watching programmes. o And what will come after IP? Which technology will it be and what are we missing today? "Technology is our word for something that doesn't quite work yet" (Danny Hillis, Douglas Adams). New network technology is something that we have, which offers us a lot of opportunites but that don't quite work yet with many challenges to ensure the expected performance, in a number of environments for a number of different reasons. One of the role of the EBU Network Management Committee is to help to make it work, and this seminar should contribute.
EBU Networks 2008 Seminar / 23 - 24 June 2008 Reproduction prohibited without written permission of the EBU Technical Department & EBU International Training
IP and MPLS
IP and MPLS has been the network technology hailed as the method of delivering high QoS media services across both metro and wide area networks for some years now. This first session explores both the theory and the practicalities of actually trying to achieve the business needs of the broadcast and production markets when using this network protocol along with the lessons learned up to now. (Reminder from the Networks 2007 seminar report 3.3) MPLS is short for Multiprotocol Label Switching, an IETF initiative that integrates Layer 2 information about network links (bandwidth, latency, utilization) into Layer 3 (IP) in order to improve IP-packet exchange. MPLS gives network operators a great deal of flexibility to divert and route traffic around link failures, congestion, and bottlenecks. It has been developed since 1999. It is complex to operate and configure for the service provider, but is simple as a "data socket" on the wall for the user. The main features of MPLS are: The MPLS Integrated Services (IntServ) "explicit routing" allows to manually create individual tunnels, taking different paths through the network, for different types of application data. If fast rerouting (FRR) is necessary, either end-to-end back-up tunnels (created manually before the failure occurs) are automatically switched over globally ("global repair"), or backup tunnels are automatically created for each segment of the primary tunnel and switched locally ("local repair"). Differentiated Services (DiffServ) and DiffServ-Aware Traffic Engineering offers a dynamic path selection using OSPF (Open Shortest Path First), the network knowing globally about available resources.
http://www.cisco.com/
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minutes if no protection is applied, and that calculations across several Service Providers show a mean time between core failures affecting video is greater than 100 hours. The impact of outage can be reduced but requires smart engineering. Aiming at lossless video transport, the following deployment scenarios can be implemented, classified in Figure 1 in terms of lossless or not, of cost and complexity of network design and deployment and application infrastructure. The corresponding techniques are detailed in the Table 1 (Annex).
Figure 1: Deployment scenarios for lossless video transport
Network Re-engineering
Increasing Loss
No network Re-engineering
Fast Convergence
Lossless
1 GOP Impacted
8 IP router and broadcast experts need to work together in the design and testing of equipment and systems. IP
measurement is not refined for media use, hence the new EBU N/IPM project group started 1 April 2008. How to install media IP network Plan your network: Physical layer, Capacity, VLANs / ACLs / Address ranges Know your IP network equipment: Codecs, Switches, Routers, etc. Know your Telco and its equipment Know and inform your users (operators) that Audio and Video over IP has limitations Test off-line you will find out the limitations without prejudicing service Ensure the network is very well managed and clean Ensure a QoS structure is set and maintained
Use good planning and change control6 to add or change codecs and network structure. Monitor the network for unauthorised additions.
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A formal method of tracking and recording proposed and actual changes to a system http://www.gen-networks.com/ http://www.gen-networks.com/content/iris/index.aspx
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Guaranteeing availability is part of QoS. It is therefore critical to implement diverse protected circuits and switches and to build redundant Label Switched Paths (LSP) using MPLS and Classification. This allows Fast Reroute (shorter than 50 ms) and simultaneous IP Services (hardware manufacturers, such as Harris and Medialinks, will support receiving multiple IP Flows and do protection in the end video devices). Bandwidth management - An MPLS enabled network allows the creation of Label Switch Paths (virtual circuit) that can tunnel traffic through the network. The Traffic Engineering implements deterministic routing and enables proper bandwidth management. The Network Manager needs to ensure no oversubscription of high-priority flows. An external system, such as IRIS, is required to properly manage broadcasts. This system must: know and understand the network topology, accept and manage all high priority data on the network, understand all future bandwidth that will be placed on the network.
Figure 2: Classifing and prioritizing traffic & Inside the 'cloud'
http://www.tdf.fr/groupe-tdf/filiales/
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Exchanges of programs between regional studios [11], using MPLS, VPLS and RSVP. Backbone network for voice and data services (eg for a Wimax service provider) [12], with MPLS, VPLS and LDP. Distribution of HD programmes to DTT transmitters [13], with MPLS, VPLS, RSVP and FRR. Transport of 18 DVB satellite multiplex to the TDFs teleport uplinks in Nantes (Brittany) [14], with MPLS and RSVP. Digital distribution of radio programs from 24 regional studios to 92 regional FM or T-DMB transmitters [15], with MPLS, VPLS (point-to-point) and and RSVP (point-to-multipoint). Pro WiFi contribution access network for journalists [16]: up to 100 access points, 2 Mbit/s guaranteed, and up to 10 Mbit/s in best effort for the access, MPLS, VPLS and LDP used. Connecting remote cameras, placed on the transmitters sites, to the studios for live weather illustration or for regional use [17]; with MPLS and VPLS. Transport of COFDM ENG services from the receiving HF point to the play-out studio [18], with MPLS and VPLS.
The next step is to extend this multi-service network to the European level [19].
Audio over IP equipment from one manufacturer has until now not been compatible with another manufacturers unit. This session covers applications and real-life use of audio over IP with demonstrations and hands-on.
The transport protocol used is RTP on top of UDP [8], as in many real-time IPTV systems. For the session management, 3 protocols are used: SDP: Session Description Protocol (RFC4566), SIP: Session
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See also EBU - TECH 3329 'A Tutorial on Audio Contribution over IP'. May 2008 http://www.ebu.ch/CMSimages/en/tec_doc_t3329-2008_tcm6-59851.pdf Lars JONSSON and Mathias COINCHON 'Streaming audio contributions over IP' EBU Technical Review - 2008 Q1 http://www.ebu.ch/en/technical/trev/trev_2008-Q1_jonsson%20(aoip).pdf 11 http://www.ebu-acip.org
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EBU - TECH 3326 (revision 3) 'Recommendation for interoperability between Audio over IP equipment'. April 2008 http://www.ebu.ch/CMSimages/en/tec_doc_t3326-2008_tcm6-54427.pdf 13 http://www.aptx.com
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Initiation Protocol (RFC3261) and SAP: Session Announcement Protocol (RFC2974). For the signalling, SIP, commonly used for Voice over IP, allows to establish, maintain and end the session [13].The hosts listen to messages and respond. SIP also negotiates the audio coding parameters. To facilitate interoperability, SIP servers 14 associate names with IP addresses and allow a reporter to be found, whatever the location, whatever the device! Audio Contribution over IP should rely on managed (private) IP networks, allowing QoS. QoS can be expressed in Service Level Agreement (SLA) contracts with providers, but requirements must be clearly expressed concerning: transmission performances (latency, jitter...), network availability (99.9% 99.99%), provisioning delay (1 week? 1 month?). The definition of the measurement method is also important (EBU N/IPM group is working on profiling, measurement). A variety of 'last mile' access methods are available with different performances:
Fiber optic High quality but expensive Copper with xDSL SDSL: Symetrical uplink / downlink Bit errors lead to packet losses Mobile (3G/UMTS, Wimax, LTE) Increasing bit rates but unreliable No solutions with guaranteed QoS nowadays HSDPA / HSUPA shared channel Satellite Long delays, often shared bandwidth Inmarsat BGAN, DVBRCS providers (Eurovision in the future?) Wireless Wifi: no guaranty due to frequency sharing
'Audio Contribution over IP' Recommendations has already been implemented by many audio codecs manufacturers [16]. A recent test (February 2008) between 9 manufacturers proved that earlier incompatible units can connect with professional audio formats using the standard. Some are still premature prototypes, and not yet EBU compliant. Marketing is very aggressive. Units are still under development. An open source reference implementation by IRT / BBC R&D is in development ( 2.2).
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AEQ (Spain) http://www.aeq.es/eng/index.htm , AETA (France) http://www.aeta-audio.com , AVT (Germany) http://www.avtnbg.de/homepage_engl/start.htm , Digigram (France) http://www.digigram.com , Mayah (Germany) http://www.mayah.com/ , ORBAN (Germany) ) http://orban-europe.com, Prodys (Spain) http://www.prodys.net , Telos (U.S.A.) .) http://telos-systems.com, Tieline (Australia) http://www.tieline.com/ip/index.html
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Advantages: Easily see full traffic between both codecs under test and SIP communications with accurate timing of data Only one protocol analyser required at each of the five test station positions RTP travels directly between codecs, via hub No involvement in testing or ensuring that router configuration is correct Disadvantages: Codecs and Config PCs have to move around testing station positions minimal movement regime determined Potentially, other test stations could see communication traffic to/from SIP server from codecs not involved in test in practice, not a problem The testing consisted of 11 rounds of 45 permutations of devices under test for each of 4 mandatory codecs. More 18 than 180 tests were undertaken including reruns of some tests. G.711 was tested using an Asterisk SIP server; G.722, MPEG Layer II and Linear PCM were tested without the SIP proxy. All rounds were recorded (2 GB of data 19 stored) using the open source Wireshark protocol analyser [15] [16]. The test results in short: Of the four mandatory audio coding formats: most units were capable of connecting G.711 audio, some units were also capable of connecting with G.722, MPEG Layer II and Linear PCM. 137 total connections successfully made (G.711: 45 total = 39 audio + 6 failed / G.722: 36 total = 30 audio + 6 failed / MPEG Layer II: 36 total = 29 audio + 7 failed / Linear PCM: 20 total = 10 audio + 10 failed). Minor improvements in software versions were undertaken and reruns operated in this case but the time was limited for too many. Some encountered problems, concerning: Protocols' usage: SIP - some connection closures due to poor command handling. SDP (re-invite, options request) - mandatory attributes not supplied, and ignoring media stream port in SDP message. Codecs: G.711 - only -Law handled by some. G.722 - audio not received by some units, some related to protocol usage. MPEG Layer II - bit rate and channel limitations; poor channel handling. Linear PCM - no support; packet size handling now mandated to 4ms. The lessons learnt with the SIP Gateways. Although not strictly part of the interoperability document, are a necessary part of infrastructure requirements for broadcasters. Asterisk was used, although others are also available, but with a number of unexpected behaviours (for example: INVITE queries not relayed correctly; media stream sometimes altered, promotion of G.711 to G.722; unexpected disconnections; odd behaviour of BYE request; response by codecs handling Gateway codec presence checking is codec still there?). The SIP infrastructures need to be able to handle Audio over IP, cell phones and Voice over IP units and Video over IP as well? Some ongoing work for broadcasters required in this area. At the end January 2008 the BBC Festival of Technology took place at BBC Television Centre, London, with the first public demonstration of the reference software with 'real' devices [23]. So, what could we do with this in the future? Some thoughts: Allow video codecs. Decide on good broadcast (hopefully Open Source) codecs for audio, even codecs not in the EBU standard. Include surround sound codecs. Could produce a stand alone SIP client, with the following needs: o Use broadcast quality codecs. o Able to communicate to an EBU SIP server (if implemented). o Able to communicate with most Hardware codecs. o Both for on-the-road usage and in-studio usage. o Maybe Java-client? (What about latency?) o Ease of use. In the ultimate situation the following one-button control performs the following: o Confirm connection to SIP server. o Set up temporary communication to other end (studio?).
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13 o o o o o o o
Test line speed to other end. Choose appropriate codec. Reconnect using new codec. Automatic Fallback to lower bandwidth codec if bad line during transmission. Built in simple Audio mixer, for microphone and playback of files. Open text chat window with studio for easy communication. And so on
Devices in action Hardware-housed software versus all-software codecs: Hardware codecs, are very easy to interface within the Broadcasting house, offer a lot of functionalities, but are harder to handle for non-technicians and require upgrade attention. Software codecs, have a friendly Graphic User Interface, are easy to use for reporters and non-technicians, offer less functionalities, but can easily be integrated in the portable laptop. Our activities in the last two years: Icehockey World Championship Final in Moscow (2007): the reporters had a very nice quite expensive ISDN
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connection but decided to try using the available IP and found they were able to transmit stereo with 256 kbit/s MPEG Layer II 25 hours for 10 matches. In Quebec (2008) a 384 ??? kbit/s connection was used with no errors at all. Swedish Icehockey National series (October to April, with 3-4 matches 2 times a week at the peak period) audio contribution over IP mainly for Web and 3G distribution (10 000 ('listeners'). Opera transmission - 3 hours, with a 48 kHz, 20-bit PCM signal; it was risky, but we tested a couple of days before so we got a good idea of the traffic shape and there was satellite backup. University of Gothenburg: professors participate live over IP in current affairs programmes. The AoIP equipment was financed by the University. All Swedish Radios foreign correspondents are IP-enabled, so transporting less expensive equipment - where IP available. Alltogether, 30 external companies contribute to SR Radio programmes. For the past 6 months some have been delivering music live streams over IP at a fairly good rate (384 kbit/s).
Final questions and answers Can the Internet be used and recommended in general for audio contribution? Of course not yet! It is not secure yet! Only as a last option - if you don't have anything else, try it! Recommendations: keep the packets size very large large (800 bytes or more) and see that you have a 50 % bit rate overhead. What can wireless networks be used for? For file transfer mainly, because the delay and the jitter are too much out of control to be used for live transmissions. Which codecs are the best? Software codecs have a future, but inside the Broadcasting house hardware codecs will remain, because they are easily interfaced with the rest of the infrastructure.
2.4 Demos
Figure 3: Audio over IP network over satellite with DVB-RCS (future Euroradio!)
APT APT
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This session gives an overview of the work and first results of the EBU 'Video Contribution over IP Networks' (N/VCIP) project group and two practical use cases for video contribution over IP networks.
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Profile 1 Uncompressed video (SD, HD) over IP Profile 2 Compressed high bit rate profile: Transport Stream over IP according to Recommendation SMPTE 22 2022 This profile will be chosen when Profile 1 is too expensive or bandwidth not available ~ 7 - 30 Mbit/s for SD (576i25) ~ 30 - 300 Mbit/s for HD (1080i25, 720p50) ~ 30 - 500 Mbit/s for HD (1080p50) depending on the compression technology FEC level B (row and column) To be defined No FEC specified yet - ISMA standard has to be enhanced with FEC Very efficient lower bit rate profile Profile 3 MXF over IP Profile 4 Low bit rate: ISMA 2.0 (Internet Streaming Media Alliance) Between profile 1 and 2 (uncompressed or compressed) ~ 3 - 15 Mbit/s for SD (576i25) for News
~ 300 Mbit/s for SD (576i25) ~ 1,6+ Gbit/s for HD (1080I25, 720p50) ~ 3,2+ Gbit/s (3.5 w. FEC) for HD (1080p50) FEC - still to be cross checked with existing specifications
Positive points
Supports TS with constant bit rates In the future other compression formats beside MPEG-2 and MPEG-4 AVC / H.264 may be transported in MPEG-2 TS No complete standard available at the moment Transmission networks expensive? Capacity available? Video Services Forum (VSF) proposal is under evaluation SMPTE???
Suitable for compressed and uncompressed video No standard available yet Standard has to be created!
Pending questions
Useability for HD to be determined This profile has raised a lot of questions from the manufacturers
Some topics are still to be covered: Signalling: is the usage of SIP recommandeable? Network profiles for video contribution, in relation with the newly created (April 2008) N/IPM project group in charge of: o Reviewing any relevant work defining network QoS classification for broadcast contribution feeds o Specifying and conducting tests of IP network components o Gathering information on IP measurement methods and tools o Specifying classification for IP networks suitable for use by European public service broadcasters o Developing network measurement methods suitable for verifying the various classes of services levels defined for IP broadcast networks Scrambling (encryption)
SMPTE 2022-1-2007 Forward Error Correction for Real-Time Video/Audio Transport Over IP Networks SMPTE 2022-2-2007 Unidirectional Transport of Constant Bit Rate MPEG-2 Transport Streams on IP Networks 23 http://www.t-vips.com/
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17 24 JPEG2000 MXF storage format defined (SMPTE 422M ) JPEG2000 is already in use in production (for example, Thomson Grass Valley HD/SD Infinity camcorder25 and a
number of video servers) JPEG2000 offers 3 operational modes for TV broadcasting: Normal mode (irreversible) with floating point filter, optimised quantisation and codestream with bit rate limitation. The typical compression ratio: is 10-20 %. Lossless mode with bit rate limitation. This mode applies integer filter, no quantisation and codestream limitation with bit rate limitation. The typical compression ratio is 40-60%. Lossless mode. This mode applies integer filter, no quantisation and all coefficients included in the codestream. Bit rate will be unrestrained. The JPEG2000 bit rate characteristics are, after compression: 20-25 Mbit/s from SDI 270 Mbit/s, 60 120 Mbit/s from HD-SDI 1.485 Gbit/s, 200 250 Mbit/s from HD 3G , Digital Cinema uses 250 Mbit/s. The lossless mode allows a bit rate reduction of 30 to 60 %. JPEG2000 is by nature VBR - the number of bits produced by the compression may be less than the bit rate configured. This may be used to save bit rate on transmission. JPEG2000 combined with MXF / FEC / RTP / UDP/ IP provides a good solution for transport over IP [18] MXF does provide a frame-based (progressive or interlaced scanned) wrapping of JPEG2000 picture, plus sound, VANC and HANC data, and can handle synchronisation of video & audio [19]-[21]. ]. In the T-VIPS implementation Preamble and Post-amble are skipped in order to reduce overhead (Typical overhead: 6-8 % incl. Ethernet and IP headers). 26 The Forward Error Correction (FEC) [22] corresponds to the Pro-MPEG Forum Code of Practice #4 or to the 27 SMPTE 2022 matrix. For Variable Bit Rate (VBR), operation the FEC matrix being of fixed size, stuffing packets are inserted [23]. The IP encapsulation supports a bit-rate range from 1 to 1000 Mbit/s,VBR, FEC, and preserves low latency. This contribution service has already been used for: Euro 2008: 2 HD feeds at 250 Mbit/s and 3 SD feeds between Vienna (Austria) and Scandinavian broadcasters in Stockholm, Copenhagen and Bergen. The high bit rates were used as an STM-4 connection was available. Transmission of ice hockey competitions from 12 arenas: 1 HD and 2 SD at 100 Mbit/s, with integration into Content Management Systems Primary Distribution of the the Norwegian DVB-T since September 2007. The network was installed and is operated by the telecom operator Telenor. It comprises 3 multiplexes with MPEG-4 coded SD video. JPEG2000 were preferred as an alternative to MPEG-2 in order to use a higher compression on MPEG-4 and provide the same resulting quality to the viewers.
IP/VPN for real-time contribution (with MPEG2 compression): NGN28 range network, IP-based network (Layer 3)
and native multicast (no tunneling). Used for live and rushes transmission (from/to spots where files exchange
24 25 26 27 28 SMPTE 422M-2006 Material Exchange Format Mapping JPEG2000 Codestreams into the MXF Generic Container
http://www.thomsongrassvalley.com/products/infinity/camcorder/ Peter Elmer and Henry Sariowan 'Interoperability for Professional Video Streaming over IP Networks' SMPTE 2022-1-2007 Forward Error Correction for Real-Time Video/Audio Transport Over IP Networks http://www.itu.int/ITU-T/ngn/
EBU Networks 2008 Seminar / 23 - 24 June 2008 Reproduction prohibited without written permission of the EBU Technical Department & EBU International Training
www.broadcastpapers.com/whitepapers/paper_loader.cfm?pid=221
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dont exist). Both services (called Data and Real-time) are delivered on a mutualized infrastructure (optical fiber) for cost efficiency. The separation of services is logical (VLAN) and in the shaping of the bandwith. The Real-time Network has following characteristics: By choice, there is no QoS! There are 'only' real-time streams on this network! The bandwith is 100 % guaranteed by the provider (+ 10% of over-provisionning). Our NMS manages equipment generating CBR. So we can indirectely manage bandwith and prevent congestion. All streams are in multicast, making it easier to manage for the transmitters and for multi-receivers. Therefore only decoders have to subscribe to multicast streams (very similarly to Set-Top Box in IPTV). Encoding equipment 29 The MPEG-2 encoders / decoders 'ViBE' are products from Thomson Grass Valley , with Ethernet Interfaces Typically, the MPEG-2 MP@ML encoding profile requires about 8 Mbit/s (IP level). The normal delay (about 900 ms, end-to-end) is used for rushes transmission. The low delay (about 450 ms) is used for live. There is no FEC implementation at the moment. Example of the network management at the level of a Regional Center: the network bandwith is of 24 Mbit/s and is managed with 8 Mbit/s one-way, making it possible to have 3 input/output links. The NMS will authorize 3 live receptions, but will first refuse the fourth. All links being managed by a 'Scheduler', the fourth link will be proposed after, or the operator must stop one of the 3 links. Among the evolution foreseen: For the monitoring of the Network, Media Delivery Index (RFC 444530) probes are in evaluation. Implementation of FEC (probably 1D), depending on the results of the network load increase. Implementation of Video servers, in order to be able to record MPEG/IP streams directly through Ethernet decoders interfaces. Development of an interface between our NMS and our Production Media Asset Management for a better performance of our workflow.
HD over Networks
This session is about current status and new key technologies for High Definition Television Production in a networked environment. The high bandwidth demand of uncompressed HD signals often necessitates some kind of signal compression in order to enable storage on hard disks and transport over networks. This session brings some more insight into what is lying ahead of us as well as some experience that can be gained from current real world HD-TV production and emission.
http://www.thomsongrassvalley.com/products_disttrans/ http://tools.ietf.org/html/rfc4445
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19 Via fiber, offering an unlimited bandwidth, but being not efficient for multi-point contribution.
HD content suggests a higher viewing experience for end users compared to SD, and therefore HD contribution should then meet following challenges: Higher data rate to handle efficiently in bandwidth limited networks Lower latencies or at least the same as in the SD world Higher bit depth (at the moment up to 10-bit) Multiple image formats (SD, 720p/50, 1080i/25 and potentially 1080p) Frame rate conversion for overseas transmissions Robustness to cascades Etc. The Table 3 (Annex) details the characteristics, advantages and disadvantages of the choice of codecs. The selection of the codec should be made based on... Product disponibility Experience / knowledge of the technology Comparisons with other systems Recommendations on adequate settings... Suitability for specific applications and/or networks: SNG, live (sports, documentary...), off-line exchange versus via satellite / fiber / copper Costs concerning IPR issues, scalability for futureproofness, backward compatibility with older compression systems... And how can the EBU Technical Department help you make this choice clearer ? The ongoing work concerns: EBU Technical department: study of HDTV format conversion over contribution link (with BBC and IRT) Evaluation of contribution codecs (H.264/AVC, Dirac ...) in the I/HDCC group - Evaluation of the SVC potential for HDTV broadcast applications in the D/SVC group. WBU ISOG with EBU: interoperability test for MPEG-4 AVC / H.264 compression system BBC: evolution of the Dirac Pro codec (SMPTE VC-2)
ftp://vqeg.its.bldrdoc.gov/HDTV/SVT_MultiFormat/SVT_MultiFormat_v10.pdf
ITU-R BT.1122-1 (10/95) User requirements for emission and secondary distribution systems for SDTV, HDTV and hierarchical coding schemes http://www.itu.int/rec/R-REC-BT.1122/en 35 See also the EBU Networks2005 seminar report 1.1.2
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contribution need to be developed for 4:2:2, 10-bit with both I-frame based mode and GOP-mode! Concerning international (HD) contribution SVT has received the 2006 summer football contribution from Germany, skiing and athletics from Sapporo/Osaka, Song Contests etc... All bit rates have been too low! You can see degradation on a 1st generation Omneon server recording (MPEG-2 GOP at 100 Mbit/s is too low). A contribution at 45 Mbit/s (hopefully greater than 60 Mbit/s) before that server isnt an excellent cascading situation. Use MPEG-4 AVC/H.264 for contribution (4:2:2, 10-bit) but not to reduce the too low MPEG-2 bit rates of today by 20-30 %, but to gain robustness! Push-up the AVC/H.264 GOP-based bit rates to more than 70-100 Mbit/s! Format conversions... may happen before or after contribution. SVT prefers to perform any format conversion before contribution, as we did in Sapporo (1080i29.97 to 720p50), we have the bandwidth and can do better de-interlacing thzan at the consumer side. Big bulky frame rate-converters 36 (59.94 to 50) with good performance should be cheaper and card-based, please! Card-based de-interlacers (50Hz-1080i to 50Hz-720p) should offer better performance, please! Down-converters to SD (re-interlace) should have separate H and V sharpness settings, please! to reduce interline twitter when performing the always needed sharpening. Re-insertion of VANC (e.g. timecode) and Dolby E should work without problems, please! 1080p50 (and 59.94) contribution? The S/N ratio in 1080p-camera CCD sensors is problematic, (but not in native 720p CCD sensors). Although 1080p50 production may become marketed during 2009, CMOS sensors may give us good (that means better than CCD) S/N ratio first in 2011. But although the real take-off may not start until 2012, there is a realistic growing need from 2009. So, please... offer us MPEG-4 AVC/H.264 codecs Level 4.2 (SVC is not needed for contribution) codecs, with 4:2:2, 10-bit, both I-frame based high bit rate and GOP.
For example, from Teranex Very low for HD contribution, only on request from the client
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sequence, interoperability. Finally the market will decide. Since MPEG-4 AVC/H.264 equipment range for contribution is not fully available, so MPEG-2 will survive for some timeAnd anyway how much broadcasters are ready to pay? File transfer is an alternative with lower costs (and increased picture quality?).
The final two sessions will outline some recent developments in using new network technologies to support the broadcast environment. The first session looks at the impact of networks on news-gathering kicking off with a use case looking at the application of COFDM for ENG application, with particular attention to the monitoring of the quality of the signals. This will be followed by some real examples of some of the innovative techniques being used by journalists to deliver material to supporting their story-telling.
http://www.linkres.co.uk/ Belgian coast network is not yet operational but will be soon
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For example, with the the 'Toko', manufactured by Toko Japan (http://www.toko.co.jp/top/en/index.html ), a device that digitally compressed video, stored it and forwarded it through a telephone or a satellite phone. 41 See EBU Networks 2007 seminar report 5.3
42 43 44 45 46
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connection - whenever they need to. Why does a VJ need a network connection? Because the basic rule for news is: faster = hotter. The main advantage of VJs is the spontaneous acquisition in difficult areas and locations. What is the payload? It's more ready files than streaming. For audio, the typical bit rate is less than 200 kbit/s (max about 1.5 Mbit/s) - bidirectional Internet streaming can work for it. For video (incl. audio) the bit rates are multiple times of audio < 4 Mbit/s... 12 Mbit/s . 25 Mbit/s - video streaming over open Internet is difficult (no way!). IRT is testing Multiple Description Coding (MCD) and Scalable Video Coding (SVC) solutions. There are numerous ways for a VJ to get connected. The common network interfaces of the VJ equipment are: Ethernet ( 1 Gbit/s), WiFi (nominal up to 56 Mbit/s), mobile (phone) connections (UMTS, HSPA), WiMAX. Table 4 (Annex) lists the overwhelming crowd of networks worldwide and actual bit rates measured on some relevant networks. VPN security is important for VJs. It is useful for the protection of the reporter identity, for very critical material especially in 'monitored' networks (China & Co.), for full LAN-integration (with all the administrative overhead), and for securing applications with low security levels (like FTP). But: Transfers can be secure enough if native mechanisms are used: HTTPs, sFTP, file encryption and signing, dedicated input server (in DMZ). The security overhead can have massive negative impact on the transfer-throughput. Therefore individual performance-tests are essential. For example [14], a 3.4 Mbit/s bit rate through a no VPN circuit, will become 47 48 3.3 Mbit/s with CryptoGuard VPN software (with AES encryption) and decrease to 1.9 Mbit/s with OpenVPN , an open source SSL VPN solution. The VPN throughput is important for daily VJ-business, even very low bit rates are challenging the available connections. Here is an example of calculation: A 5-minute video material coded at 4 Mbit/s will deliver 1.2 Gbit in 5 minutes (30 MByte/minute) 25 Mbit/s will deliver 7,5 Gbit in 5 minutes (187 MByte/minute) The Table 4 (Annex) indicates the corresponding transfer duration of this 5-minute material on relevant networks. Finally: how to connect? Every region and continent has its own focus: e.g. Europe develops more UMTS, Asia more WiMAX. But up to now there is no always fitting connection-type. Fixed line is still the best choice - and worldwide available, with DSL - WiFi, or cable, or Ethernet connection. Most promising for now and especially in the future: WiMAX, LTE (4G) UMTS-HSUPA, especially in Europe Fixed line (xDSL including local WiFi)
In earlier sessions of this seminar, you will have learned about the latest technologies which permit the construction of networks designed to allow contribution transmissions for television production. This final session will provide examples of the impact of new network technology on television production and broadcast.
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http://www.compumatica.de/cms/data/index.php?id=21&L=0 http://openvpn.net/
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Conventional network design does not apply here because of the specific constraints. For example, real-time video collaborative editing is very demanding and requires such networks as Unit Media network with Fibre Channel (initially 1Gbit/s but grew to be 2Gbit/s and currently 4Gbit/s supporting) or Gigabit Ethernet (for 'lower' resolutions such as DV25 or IMX30) or 10GE coming soon for uncompressed HD [9]-[11]. The associated Unity ISIS storage system allows up to 160 GbE and 20 10GE ports [12] and enables the use of higher resolutions IMX50/DV50 and even DNxHD over Gigabit Ethernet with 10GE coming soon for uncompressed HD [12]-[13]. Gigabit Ethernet end-to-end uses UDP instead of TCP. Just like VoIP this type of video protocol should let the application decide if it needs to re-request the data. A Fast Ethernet connection (100 Mbit/s) results in dissatisfied users, especially if they move between machines with Gigabit Ethernet interfaces. Editing application can co-exist with Video over IP corporate solutions if these are designed correctly. Should we transfer jumbo frames or fragmented packets? Video data is in large blocks on the disks, so it should be ideal! It is more efficient to send larger segments of data, especially with UDP, but it only sustainable in a controlled environment (e.g. server room). Fragmented datagrams can pass over non-jumbo frames enabled networks but need on-board memory in network interface cards (NIC). The aggregation of multiple micro-servers requires flexible, expandable buffering [17]. Big chassis-based Gig-E switches have different cards with different abilities - every interface card matters! Different chassis-based switch models from same manufacturer have different abilities - biggest is not always the best! NIC (Network Interface Card) - The buffers (or descriptors) required for this application would have been considered server class just a couple of years ago, but now this class of adapter can be found on high and medium grade platforms. On board memory is needed for fragmented packets - 32 KB on low cost NIC implementations is 49 insufficient. If a TCP-based data flow is used there will be lots of CPU requirement unless ToE -enabled NIC is used. A separate production network is best for the core systems and high bandwidth clients. Corporate network based clients are possible for less demanding application and resolutions, if the corporate infrastructure can cope and has the right products in the correct place. Separate the Network into zones from the core highest bandwidth zone 1 (with direct storage connection) to the lowest bandwidth (customer network) zone [21]-[23]. In a video production environment QoS generally relates to available bandwidth, latency & jitter. Vendors provide different QoS tools that you should use on edge and backbone routers to support QoS. Some QoS mechanisms are aimed at VoIP networks and low bandwidth circuits, but others apply equally to LAN. Example of QoS mechanisms for Video over IP are: LLQ (low latency queuing), PQ (priority queuing), WFQ (weighted fair queuing), CQ (custom queuing), PQ-WFQ, CBWFQ (class-based weighted fair queuing). A good old fibre channel is very deterministic and at 5ms latency the effect begins to show. Design the Gigabit Ethernet system with sufficient capacity to ensure the necessary QoS. In a corporate network ensure sufficient QoS exists - the corporate network may need to be reconfigured to correctly recognise and prioritize video traffic. Firewalls present many challenges. They are not good at bursty, high bandwidth, fragmented real-time flows. In a production environment we deal with REAL TIME VIDEO, not a streaming VC-1 file! Again, contain the network design into zones and locate externally facing clients in the higher number zone, or even in the corporate network. Buffers, buffers, buffers! Switches with dynamically shared buffers are a better choice - some manufacturers provide 1U or 2U condensed versions of popular mid-range chassis-based switches. Switches with statically assigned buffers limit the design scope - this limit affects some large chassis-based switches from some MAJOR manufacturers, while smaller chassis-based models have shared buffers and are an excellent choice. Buffers in the network cards are also critical. In conclusion, understand every link in the chain (some design examples are presented [29]-[31]) and where that link exists! Every switch, every speed change, every aggregation point MATTERS, and that includes the NIC in the
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50
TCP offload Engine -a technology used in network interface cards to offload processing of the entire TCP/IP stack to the network controller. It is primarily used with high-speed network interfaces, such as Gigabit Ethernet and 10- Gigabit Ethernet, where processing overhead of the network stack becomes significant. 50 http://www.techweb.com/encyclopedia/defineterm.jhtml?term=QoS&x=&y=
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client!
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sites. The IP Data Service ensured the transition: o From a point-to-point E1-based service, with ISDN2 backup, unicast, and no QoS; o To a managed IP solution on a 10M/100M main link, with 2M/10M backup link, multicast, QoS, scalable, integrated with Raman Core sites and between Siemens and C&W Management systems. The Broadcast Audio Service, with a transition: o From a Musicline2000 product, unmanaged point-to-point audio, 15 kHz mono, J.41, over E1 - limited scale; o To a C&W Audio-over-IP solution using the APT Oslo platform, a managed solution, multicast, scalable, with enhanced APT-x low latency coding, 22 kHz stereo audio, supporting 5.1 configuration (if required by BBC), and potential for future integration with BBC-wide scheduling systems. The Telephony Service with a transition: o From fractional E1 voice links with legacy TDM PBXs o To a Siemens Managed IP Telephony solution, with the central HiPath 8000 system, IP handsets, the VoIP traffic using the new IP WAN infrastructure.
6.3 DVB-H small gap fillers: home repeaters improving indoor coverage
Davide MILANESIO, Centro Ricerche e Innovazione Tecnologica, RAI, Italy DVB-H is a system specified for bringing broadcast services to battery-powered handheld receivers. It is an extension of the DVB-T system [3], transmitted in the UHF band, allowing multi-channel, robust with respect to disturbances and for reception indoor, pedestrian and at high speed (train), designed to reduce the battery consumption. The video is encoded in MPEG-4 AVC/H.264: either in CIF format (352x288), carrying 10 - 11 channels (with QPSK modulation and FEC), at a bit rate between 350 - 400 kbit/s. or in QCIF format (176x144), carrying 15 - 30 channels (with QPSK modulation and FEC), at a bit rate between 128 - 256 kbit/s. The H.264 video data is transported over RTP/UDP/IP with an additional Multi-Protocol Encapsulation FEC protection (MPE-FEC) and interleaving. The IP streams are then encapsulated in a DVB MPEG-2 Transport Stream. 51 DVB/H supports the datacast of files using the FLUTE protocol . DVB-H is already operational worldwide: in Italy (since 2006), Finland (since 2007), Austria-Switzerland (June 2008), Albania, Asia (India, Malaysia, Philippines, Vietnam) and Africa (Kenya, Namibia, Nigeria). But these networks are mainly planned for outdoor coverage not for indoor! Therefore traditional DVB-T network planning is not sufficient for DVB-H, because indoor DVB-H reception requires higher electromagnetic field strength, since the receiving antenna is integrated in the terminal and not on the roof! [6] To improve indoor coverage, the main transmitters could be completed by a number of low power urban transmitters. But electromagnetic radiation limits have to be respected and there is a risk of interference on traditional TV services in the existing MATV distribution systems [7]. DVB-H 'small gap fillers' are another way to improve indoor coverage using low-power on-channel home repeaters. These consumer-grade devices can be autonomously installed by final users in their private homes, without the help of a professional installer. It is connected to the existing in-building cable distribution system [8], Its coverage area is this of a standard apartment, i.e. about 100 m2 enabling the interested users to be immediately reached by DVB-H services. Standardisation Since these devices are radiating in the UHF band, without a specific regulation, a licence would be needed. Therefore a new standard is necessary, also to avoid that low quality (illegal) devices appear on the market, potentially causing dangerous interference on existing services (e.g. analogue or digital TV). The DVB-H Small Gap Fillers Task Force, with 21 companies involved (Broadcasters, network operators, regulation authorities, manufacturers), has prepared a Technical Specification. It has been approved by the DVB Steering Board, in June 2008, for publication as a DVB Blue Book. It will then be submitted to ETSI/CEPT for publication as European Norm (requiring voting by National Standards Organisations).
51
http://www.ietf.org/rfc/rfc3926.txt
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The DVB-H Small Gap Filler includes two sections [10]: Signal processing section (for on-channel filtering and amplification) and a built-in DVB-H receiver for output signal quality monitoring. This last stage integrates an automatic power control mechanism [12], based on the measured quality, in order to avoid interferences on existing services, even in case of failure of the device electronics or mistakes of the user. The frequency response [11] shows a high attenuation of the adjacent channels. Out-of-band emissions are according to existing regulations. Validation. The Technical Specifications have been validated in laboratory trials on real hardware prototypes [13] 52 and in the framework of the European Project 'CELTIC B21C' . Coverage tests were conducted in the Rai-CRIT laboratories and in a real flat and proved an adequate coverage in standard apartments (e.g. 100 m2) [14]. The disturbance on adjacent TV channels was tested using 2 reference scenarios [15], with positive results [16].
Scenario TV / STB connected to another plug of the in-building cable distribution network TV / STB in another room, connected to an indoor amplified antenna 1 wall separation, 3 m distance Results Video SNR degradation within 1 dB, not noticeable on picture Video SNR degradation within 2 dB if using 2 SAW filters, not noticeable on picture. Degradation within 5 dB with more relaxed masks (out of standard) OK also in pessimistic condition: Adjacent analogue TV channel received by the repeater with level 30 dB higher than DVB-H
A CATV network could allow for a possible future extension of the DVB-H Small Gap Filler concept. The DVB-H multiplex could be transported on the CATV network on behalf of the broadcaster [17]. The indoor DVB-H coverage will be improved in areas where TV aerials are not very popular but where a CATV network is available. CATV would be used only as a carrier, no DVB-C signals being involved. This would allow to reduce the DVB-H network deployment costs. But this involves an additional requirement, the possibility of frequency conversion. Since the CATV network might not cover the full UHF band. Therefore, the coherence of the output frequency has to be guaranteed, i.e. the output frequency has to be cross-checked with the Network Information Table of the incoming stream. Moreover the network operator should guarantee the synchronisation with the traditional DVB-H transmitters as in a standard SFN [18]. Currently, frequency conversion is not included in the Specification.
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Celtic (Cooperation for a sustained European Leadership in Telecommunications) - Broadcast for the 21st Century
http://www.celtic-initiative.org/Projects/B21C/abstract.asp
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2-dimensional Enhanced 3rd generation of wireless communication systems (HSDPA*) 3 generation of wireless communication systems (UMTS*, WCDMA*) 3rd Generation partnership project 4 generation of wireless communication systems (LTE*) also known as Audio/Video, Audiovisual Advanced Audio Coding AAC* Enhanced Low Delay AAC* Low Delay (MPEG-4 Audio) Audio Coding technology (Dolby) Acknowledge-ment message Access Control List Asymmetric Digital Subscriber Line Audio Engineering Society Application Layer Adaptive Multi-Rate WideBand (G.722.2) Audio over IP (broadband audio) Accelerated Private Network Asynchronous Serial Interface (DVB*) Asynchronous Transfer Mode Advanced Video Coding (MPEG-4) AVC* - Intra Bidirectional predicted picture (MPEG) Bit Error Ratio Broadband Global Area Network (Inmarsat) - cf. EBU Networks2007 seminar report 5.3 Border Gateway Protocol (Internet) Bandwidth Cable & Wireless http://www.cw.com/new/ Content-based Adaptive Binary Aritmethic Coding (MPEG-4 AVC) Cable TV Context-based Adaptive Variable Length Coding (MPEG-4 AVC) Constant Bit Rate Class-Based Weighted Fair Queuing Charge Coupled Device Code-Division Multiple Access Confrence Europenne des administrations des Postes et Tlcommunications - European Conference of Postal and Telecommunications Administrations 'Confer', consider, compare Common Intermediate Format (352*288 pixels)
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CJI CMOS CMS COFDM CoP# CPU CQ CRIT CSRC CWDM D-Cinema D/SVC DAB DCI DCM DCT DiffServ DMB DMZ DNxHD DoS DRM DSCP DSL DTT(B) DV DVB DVB-C DVB-H DVB-RCS DVB-T DVB-T DWDM DWT e.g., eg e.m., EM E/S E1 EDGE EF EN END ENG ERCS ERP ETSI FC FDM FEC Cognacq-Jay Image (TDF* subsidiary) Complementary Metal-Oxide Semiconductor Content Management System Coded OFDM* Code of Practice number (Pro-MPEG Forum) Central Processing Unit Custom Queuing Centro Ricerche e Innovazione Tecnologica (RAI) Contribution Source (in RTP*) Coarse Wavelength Division Multiplex(ing) Digital Cinema Scalable Video Coding (EBU Project Group) Digital Audio Broadcasting Digital Cinema Initiative Dynamic Channel Path Management Discrete Cosine Transform Differentiated Services Digital Multimedia Broadcasting DeMilitarised Zone High Definition encoding (Avid) http://www.avid.com/resources/whitepapers/DNxHDWP3.pdf?featureID=882&marketID= Denial-of-service attack Digital radio Mondial Differentiated Services Code Point Digital Subscriber Line Digital Terrestrial Television (Broadcasting) Digital Video cassette recording and compression format Digital Video Broadcasting DVB - Cable DVB - Handheld DVB with Return Channel via Satellite Digital Video Broadcasting - Terrestrial DVB - Terrestrial Dense Wavelength Division Multiplex(ing) Discrete Wavelet Transform exempli gratia, for example Electro-magnetic Earth Station European PDH system level 1 (2.048 Mbit/s) Enhanced Data rates for GSM Evolution Expedited Forwarding European Norm/Standard (ETSI*) 6.3.6 ??? Electronic News Gathering English Region Cluster Sites (BBC) Effective Radiated Power European Telecommunications Standards Institute Fibre Channel Frequency Division Multiplexing Forward Error Correction
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FiNE FLUTE FM FRR FTP G.711 G.722 Gb/s, Gbps GbE, GE GOP GPRS GPS GSM GUI H H/W, HW HA HANC HBA HD(TV) HDRR HD-SDI HE HE-AAC HF HFC HL HQ HSDPA HSPA HSUPA HTTP HTTPs I i.e., ie I/HDCC IBC ICE-net ICT IDCT IEC IEEE IETF IGMP IGP IMS IMX (MPEG-) iNews IOS-XR Fiber Network Eurovision (EBU) http://www.netinsight.net/pdf/040823_Casestudy_EBU_2.pdf File delivery over Unidirectional Transport (RFC 3926) Frequency Modulation Fast Reroute File Transfer Protocol Pulse code modulation (PCM*) of voice frequencies (ITU-T) 7 kHz audio-coding within 64 kbit/s Gigabit per second, Gbit/s 1-Gigabit Ethernet Group Of Pictures (MPEG) General Packet Radio Service Global Positioning System Global System for Mobile Communication Graphical User Interface Horizontal Hardware High Availibility ANCillary data in the Horizontal video blanking interval Host Bus Adapter High-Definition (Television) Haut Dbit Rseau Rgional (TDF* subsidiary) High Definition SDI (1,5 Gbit/s) Head-End (Cable TV) High Efficiency AAC* (MPEG-4 Audio) High Frequency Hybrid Fiber/Coaxial High Level (MPEG-2) Headquarters High-Speed Downlink Packet Access High-Speed Packet Access High-Speed Uplink Packet Access HyperText Transfer Protocol HTTP* using a version of the SSL* or TLS* protocols Intra coded picture (MPEG) id est, that is to say High Definition Contribution Codec (EBU Project Group) International Broadcasting Convention Nordisk Mobiltelefon Irreversible Color Transform Inverse DCT* International Electrotechnical Commission Institute of Electrical and Electronics Engineers Internet Engineering Task Force Internet Group Management Protocol Interior Gateway Protocol IP Multimedia Subsystem Digital Video Tape Recorder recording and compression (MPEG-2 422P@ML) format (Sony) Newsroom Computer System (Avid) Self-healing and self-defending operating system (Cisco)
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IP IPoDWDM IPR IPTV IRD IRT ISDN ISIS ISMA ISO ISOG (WBU*-) ISP IT ITIL ITU JP2K, J2K KLV Lambda LAN LDP LLQ LOF LOS LPCM LSP LSR LTE MAC MAN MATV Mb/s, Mbps MDC MF-TDMA MDI MIB ML MoFRR MOSPF MP MP2 MPE MPEG MPLS MS MTR MXF n/a N/ACIP N/IPM Internet Protocol IP over DWDM* Intellectual Property Rights Internet Protocol Television, Television over IP Integrated Receiver-Decoder (-> STB*) Institut fr Rundfunktechnik (Germany) Integrated Services Digital Network Infinitely Scalable Intelligent Storage (Avid) Internet Streaming Media Alliance International Organization for Standardization International Satellite Operations Group http://www.nabanet.com/wbuArea/members/ISOG.html Internet Service Provider Information Technology (informatique) Information Technology Infrastructure Library International Telecommunication Union JPEG2000 Key-Length-Value coding (MXF*) Wavelength of light ( WDM*) Local Area Network Label Distribution Protocol Low Latency Queuing for VoIP* Loss of Frame Line-of-sight Linear PCM* Label Switched Path Label Switch Router Long Term Evolution (4G mobile system) Medium Access Control layer Metropolitan Area Network Master Antenna Television Megabit per second Multiple Description Coding Multi-Frequency Time Division Multiplex Access Media Delivery Index (RFC*4445) Management Information Base (SNMP*) Main Level (MPEG-2) Multicast-only Fast Reroute Multicast extension to OSPF Main Profile (MPEG-2) MPEG-1 Audio Layer II Multi-Protocol Encapsulation (DVB-H*) Motion Picture Experts Group Multi-Protocol Label Switching Microsoft Multi-Topology Routing Material eXchange Format Not applicable / not available Audio Contribution over IP (EBU Project Group) IP Measurements (EBU Project Group)
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N/VCIP NAT NC NGN NHK NIC NIT NLOS NMS MRC NOC OB OFDM ORT OSI OSPF P PBX PCM PCM PDH PE PHY PIM PJSIP PoE POP PPP PQ PSTN QC QCIF QoS QPSK RCS RF RFC RGB RSTP RSVP RT RTBF RTCP RTP RTSP Rx S/N, SNR S/W, SW SAN SAP Video Contribution over IP (EBU Project Group) Network Address Translation Network Connection Next Generation Networks Nippon Hoso Kyokai (Japan) Network Interface Card Network Information Table (DVB - SI) Non line-of-sight Network Management System Maximal-Ratio Combining Network Operations Center Outside Broadcasting Orthogonal Frequency Division Multiplex(ing) Operational Reliability Testing Open Systems Interconnection Open Shortest Path First Predicted picture (MPEG) Private Branch eXchange Pulse Coded Modulation Pulse Code Modulation (audio) Plesiochronous Digital Hierarchy Provider Edge Physical Layer (OSI* model) Protocol Independent Multicast VoIP* GPL free software Power over Ethernet Point Of Presence Point-Point Protocol Priority Queuing Public Switched Telephone Network Quality Control Quarter Common Intermediate Format (176*144 pixels) Quality of Service Quadrature Phase Shift Keying Return Channel per Satellite (DVB) Radio Frequency Request For Comments (IETF standard) Red-Green-Blue (colour model) Rapid Spanning Tree Protocol Resource Reservation Protocol Real-Time Radio-Tlvision Belge Francophone Real-Time Control Protocol (Internet) Real-time Transport Protocol (RFC*3550) Real-Time Streaming Protocol (Internet) Receiver Signal-to-Noise ratio Software Storage Area Network Session Announcement Protocol (RFC*2974)
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SAW (filter) SCSI SD(TV) SDH SDI SDLC SDP SDSL SFN SFTP SGF SI SIP SIS SLA SMPTE SMTP SNG SOG SONET SP SPH SR SR SRT SSL SSM SSO STB STB STM-1 STM-4 SVC T-VIPS tbd TCP TDD TDF TDM TE TLS TMS TNG ToE (card) TOS TR TS Tx UA Surface Acoustic Wave Small Computer System Interface Standard Definition (Television) Synchronous Digital Hierarchy Serial Digital Interface (270 Mbit/s) Symmetric Digital Subscription Line Session Description Protocol (RFC4566) Symmetric Digital Subscriber Line Single Frequency Network (DVB-T) SSH (Secure Shell) FTP Small Gap Filler System Information (DVB) Session Initiation Protocol (RFC*3261) Siemens IT Solutions and Services Service Level Agreement Society of Motion Picture and Television Engineers Simple Mail Transfer Protocol Satellite News Gathering Summer Olympic Games Synchronous Optical Network (SDH* in U.S.A.) Service/System Provider Service Points Hauts (TDF*) Service Router Swedish Radio Service Readiness Test Secure Socket Layer Source Specific Multicast http://www.ietf.org/ids.by.wg/ssm.html Stateful Switchover Steering Board Set-top box (-> IRD*) Synchronous Transport Module Level 1 (155 Mbit/s) Synchronous Transport Module Level 4 (622 Mbit/s) Scalable Video Coding Norwegian company To be determined Transmission Control Protocol (Internet) Time Division Duplex(ing) (UMTS*) Tl-Diffusion de France Time Division Multiplex(ing) Traffic Engineering Transport Layer Security Transport Multi-Services (TDF*) Terrestrial News Gathering (VRT) TCP/IP offload Engine. An iSCSI TOE card offloads the Gigabit Ethernet and SCSI packet processing from the CPU. Type-Of-Service Temporal Redundancy Transport Stream (MPEG-2) Transmitter Unit Address
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UAT UDP UHF UID UTRA TDD UMTS UPA(HS) V VALID VANC VBR VC-2 VCEG VJ VLAN VLC VoIP VoIP VP VPLS VPN VQEG VRT vs. VSAT VSF W w. WAN W-CDMA, WCDMA WBU WC WDM WFQ WiFi WiMAX WLAN WOG xDSL YCbCr User Acceptance Test User Datagram Protocol (RFC*768) Ultra High Frequency Universal IDentifier UMTS Terrestrial Radio Access Time Division Duplex Universal Mobile Telecommunications System Uplink Packet Access (High-Speed) Vertical Video and Audio Line-up and Identification (http://www.pro-bel.com/products/C132/ ) ANCillary data in the Vertical video blanking interval Variable Bit Rate SMPTE code for the BBC's Dirac Video Codec Video Coding Experts Group (ISO/IEC-MPEG + ITU-T) Video Journalist Virtual LAN* Variable Length Coding Video over IP Voice over IP(narrowband audio) Virtual Path Virtual Private LAN Services Virtual Private Network Video Quality Experts Group (ITU) Vlaamse Radio- en Televisieomroep, Flemish Radio- and Television Network (Belgium) versus; against, compared to Very Small Aperture Terminal Video Services Forum http://videoservicesforum.net/index.shtml Watt With Wide Area Network Wideband CDMA* World Broadcasting Unions http://www.nabanet.com/wbuArea/members/about.asp World Championship Wavelength Division Multiplexing ( Lambda*) Weighted Fair Queuing Wireless Fidelity Worldwide Interoperability for Microwave Access (IEEE 802.16) Wireless LAN IEEE 802.11(a,b & g) Winter Olympic Games x DSL* (x = Asymetrical or Symetrical uplink/downlink) Digital luminance and colour difference information
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Tableau 1: Towards lossless video transport - Deployment scenarios ( 1.1) [ Cisco] 1) Fast Convergence or Fast Reroute (FRR) [8] 2) Application Layer Forward Error Correction (FEC) [15] 3) Temporal Redundancy (TR) [16] 4) Spatial (Path) Diversity ("live / live") [18] 5) Multicast-only Fast Reroute (MoFRR) [20]
Adds redundancy to the transmitted data to allow the receiver to detect and correct errors (within some bound), without the need to resend any data.
The transmitted stream is broken into blocks, each block is then sent twice, separated in time. If block separation period is greater than the loss of connectivity, at least one packet should be received and video stream play-out will be uninterrupted.
Two streams are sent over diverse paths between the sender and receiver
Deliver two disjoint branches of the same PIM (Protocol Independent Multicast) SSM (Source Specific Multicast) tree to the same Provider Edge The Provider Edge locally switch to the backup branch upon detecting a failure on the primary branch http://www.nanog.org/mtg0802/farinacci.html
Lowest bandwidth requirements in working and failure case Lowest solution cost and complexity The simplest and cheapest design / operational approach for a Service Provider is to have such behaviors optimised by default in the software and hardware implementations and is applicable to all its services. Requires fast converging network to minimize visible impact of loss
Supports hitless recovery from loss due to core network failures if loss can be constrained No requirement for network path diversity - works for all topologies
Supports hitless recovery from loss due to core network failures if loss can be constrained No requirement for network path diversity works for all topologies
Supports hitless recovery from loss due to core network failures if have network stream split and merge functions (e.g. Dynamic Channel path Management) Lower overall bandwidth consumed in failure case compared to FEC Introduces no delay if the paths have equal propagation delays
Hitless: The Provider Edge uses the two branches to repair losses and present lossless data to its IGMP (Internet Group Management Protocol) neighbors A simple approach from a design and deployment and operations perspective MPLS and Multi-Topology Routing are options in topologies that do not support MoFRR
May require network-level techniques to ensure spatial diversity, MoFRR (5), Multi-Topology Routing, Traffic Engineering; required techniques depend upon topology
Higher overall bandwidth consumed in failure case compared to "live / live" (4) Incurs delay - longer outages require larger overhead or larger block sizes (more delay)
Incurs 100% overhead Incurs delay - longer outages require larger block separation period
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(*) According to the number of multicast groups 400 - 800 - 4000, the median / max reconvergence time for all channels following a network failure may amount to 200/290 ms 260/380 ms (no more than 1 frame loss) 510/880 ms [11]. To improve it, prefix prioritisation allows important groups (e.g. premium channels) to converge first, and developments with IP optical integration can further reduce the outage to sub 20ms in many cases (lossless in some cases) by identfying degraded link using optical data and signaling before the traffic starts failing [12]. (**) Fast rerouting: the routing protocol detects the failure and computes an alternate path around the failure [10]. But loss of connectivity is experienced before the video stream connectivity is restored.
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Tableau 2: Practical measurement and practical network performance ( 1.2) [ Siemens] Practical measurement Audio latency and lip-sync With a commercially available Lip-sync measurement system; VALID from Pro-Bel. It uses a test signal with time markers in the picture and sound by which a VALID reader can measure the audio to video timing and display the result directly in millisecond [7] [8]. For latency measurement [6], the generator's video signal is connected direct to the reader while the audio is passed through the system under test. The measurement is therefore the latency of the audio path. This is applicable to both sound only and sound with vision circuits. The simple approach is to measure the round trip latency [10]. Ping is a simple and effective tool but has to be used with care. If QoS is used essential when the route is shared with data traffic - the Ping traffic through another port is unlikely to receive the same QoS marking as the media traffic. However, it can still be used when the connection is being set up and before business traffic is carried. The media stream can be run over the connection before any other traffic is routed. It is possible to measure latency over a single route using test equipment with GPS references at each end [9]. So far these tests have confirmed the round trip latency is simply double the one way latency. This goes wrong if there are unexpected constraints in the network configuration but these are very likely to show up in other ways, usually as severe packet loss. Network jitter Again, Ping is a useful tool if used appropriately [15]-[17]. Networks have a tendency to suffer 'data storms' - a useful catchall phrase for sudden and unpredictable changes in the latency and jitter. If the ping time is stable while the media traffic is running, it is a very good sign. As ping time jitter starts to rise, the likelihood is media traffic is affected as well. I have captured the effects of data storms on ping time and correlated them with damage to the media stream. Sometimes however, network operators are reluctant to accept this simple test as evidence of problems. More sophisticated testers can measure the performance with full bit rate data in both the Expedited Forwarding and the Best Effort QoS levels. Ideally, this type of testing is done as part of the acceptance testing of a network connection by the service provider. Video jitter Some codecs maintain their IP buffers at the mid point by pulling the output video frequency. If there is a significant jump in the IP latency, as may happen when switching to a backup route, the output video can become so far off frequency that downstream equipment cannot follow it. Unfortunately, there is no clear cut specification for SDI clock frequency tolerances in the broadcast standards for manufacturers to follow. A synchroniser is normally needed anyway and they accept the widest frequency range. Practical Network performance It is still routine to find codecs that make only a modest effort to get lip-sync correct. ln all cases dealt with, the IP packets carry Transport Stream and there really is no good reason for getting it wrong. Some have had large errors that are clearly unacceptable. Some have fixed errors that on their own are acceptable but add to the perennial lip-sync problems. A few just get it exactly right.
Network latency
Siemens is currently delivering an audio network specified as better than 50ms one way but it is accepted that the measurement is round trip and therefore better than 100ms. The route distances range up to 400km but it is really the number of routers in the circuit that determine the IP latency. The overall latency is dependent on compression system and settings, other things being equal. ln the case of audio, the lower bit rate, the more time is required to make up an IP packet. Live vision circuits have to be timed so a synchroniser is required. This can be built into the codec or separate but either way it will add to the overall latency.
There is no clear cut definition of jitter in this context. The issue is the variation in IP packet transit time so as to exceed the decoder buffer capacity. Some coding systems can deliver a very steady flow of IP data. Others have a steady long term average but show substantial variation in the short term.
[19] [20]
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Practical measurement This type of codec is deriving the SDI clock frequency from the IP data rate so SOI clock jitter can occur if the design is not sufficiently careful. Decoders with built in synchronising have a reference input and deal with these problems internally but add cost and latency. Glitches A fixed tone signal passed through the circuit is filtered using typically a distortion analyser to remove the fundamental [11]. Any pulses over and above noise and distortion marks a glitch. The pulses from the distortion analysis can trigger a digital storage scope, providing a record of the glitch in the tone and a count of the events [12]. The test is run for an agreed period, 12 to 48 hours. ln practice though, if glitches are really a problem, they will usually start within the first few minutes. The exception is where the test is run in conjunction with less predictable traffic over the same route. The users naturally want zero audible and visible glitches and we have tested audio circuits on this basis. While this is good goal to achieve, it is not practical in a managed service. The circuit has to be taken out of service for a long period just to make the measurement and this is after the event that a customer had cause to complain about. It is more realistic therefore to translate this requirement to the performance of the network and codecs. ln practice, this information can only be measured by the decoder. So coders with good traffic analysis and logging are an advantage. When comparing with pre-existing circuits, e.g. E1/T1, it is interesting to note that the maximum permitted bit error rate for the connection is often quoted as 1 in 10e9. This would imply 2 glitches an hour in a high quality audio circuit but in reality, E1 normally performs several orders of magnitude better than this so user expectations are much higher. The audibility of glitches is a highly contentious subject. The relationship between network events, the technical effect on the audio path and the audibility of the effect in real program material of different types could be the subject of a huge survey. Practical Network performance
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Tableau 3: HD contribution codecs ( 4.1) [ EBU]
MPEG-2 Standard ISO/IEC 13818 MPEG-4 AVC / H.264 ISO/IEC 14496 ISO 19446-1 JPEG2000 Dirac Pro SMPTE VC-2 (Developped by BBC Research) Principle DCT based system Spatial and temporal prediction with variable length statistical coding (VLC) Structure 6 Profiles (prediction tools supported) and 4 levels (image formats and max bit-rate supported) For contribution: Main profile (4:2:2) I,P and B frame prediction @ High level (1920x1080) up to 80 Mbit/s Scalability (provided SVC amendment) Highly scalable Spatially Wavelet transform. Temporally I frames Quality statistical coder Spatially scalable due to the wavelet. Scalability on top of H.264 High coding efficiency Scalability is provided layerwise: Base layer H.264/AVC compatible. Enhancement layer SVC decodable. Pros Several products available. Well known technology Cheap Proven 50% benefit compared to MPEG-2 in primary distribution bitrate range State of the art codec (using context based arithmetic coding) High compression capability with good quality at low bit Already in use by some broadcasters for internal contribution Used by DCI for 2k and 4k image format archival. Scalability might be useful to embbed several spatially different streams in the same feed. Strong error resillience of the codestream useful for transmission in error prone channels enabled by synchronisation Strong sustainability to cascadings Open source and License free no commercial risk. Very low latency (order of 6ms) since it use variable length coding for entropy coding. Support for 10 bit depth 4:2:2 Product available All pros of H.264/AVC Backward compatible with H.264/AVC Primarily considered for distribution but might appear useful in contribution; needs 10 to 30% less bit rate than simulcast. Spatial and temporal compression with a choice between context adaptive arithmetic coding (CABAC) or variable length (CAVLC) 7 Profiles and 5 levels For contribution (at the moment): High profile 4:2:2 10bit Level 4.1 (1080i and 720p) Wavelet based (DWT) compression system coupled to an arithmetic coder. Lossless and lossy compression Mixing the advantages of the wavelet transform and motion compensation. Similar wavelet filters to JPEG2000 Only DCI profiles exist at the moment. Broadcast application profiles under investigation in JPEG. Follows H.264 profiles. MPEG-4 SVC / H.264 Amendment to H.264/AVC standard
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MPEG-2 MPEG-4 AVC / H.264 rate.(suitable for SNGs) Enhanced Error resilience with synchronization mechanism in the codestream JPEG2000 High bit depth range. I frame only no interdependence between frames ; valuable for editing Visually friendly artifacts. Low latency (approx 45ms) Part 1 of the standard IPR free no license fees Proof of better performances than AVC-I for high quality large images. [] (reference software) Cons Not anymore state of the art compression system. Better technologies exist (H.264/AVC; DIRAC etc.) but the gain is still unknown Bandwidth greedy for High quality sequence exchange [see BBC presentation - EBU Production Technology 2007 seminar report 1.2] 80Mbit/s needed for high quality content exchange => too expensive on satellite GoP based system Potential for emphasised GoP Pumping visual effect in video sequences (pulsing loss of resolutions) in cascades. Additional decoding delay due to inter-frame coding DCT based => blocky artifacts not visually friendly Not that many products avalaible with the full contribution profile tools (Hi422 Level 4.1) High coding Latency (over 800ms) GoP-based Additional decoding delay for post production. GoP pumping effect. Gain over MPEG-2 in high bit rates (contribution) not yet defined. Needs very High bit rates to provide very good visual quality (over 100Mbit/s) depending on the image format) under cascaded and shifted generations Not that many products available. No comparison with other systems made till now All cons of H.264/AVC No product available yet Dirac Pro (Hardware & software) Handles formats from 720p/50, 1080i25, 1080p/50, to 4K. (Futureproof.) Lossless and visually lossless compression MPEG-4 SVC / H.264
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Tableau 4: The ovewhelming crowd of networks worldwide! [ IRT] Acccess technique Bit rate (uplink) Performance of relevant networks (measured) Transfer duration of a 5-minute material (4 Mbit/s material)
Fixed line Tel. modem ISDN ADSL SDSL VDSL Cable network Powerline (PLC) Wireless WLAN (802.11b/g) Satellite link GSM GPRS HSCSD EDGE CDMA One UMTS HSUPA CDMA2000 EV-DO Flash OFDM LTE / HSUPA WiMAX (mobile) iBURST 11 / 54 Mbit/s 1 Mbit/s 12 kbit/s 171.2 kbit/s 57.6 kbit/s 473 kbit/s 14.4 kbit/s 128 kbit/s 5.8 Mbit/s 144 kbit/s 5.4 Mbit/s 900 kbit/s 50 Mbit/s 7 Mbit/s 346 kbit/s Vodafone IRT-TESTnet ~ 1.35 Mbit/s ~ 1.15 Mbit/s ~ 14.5 min ~ 17 min. Vodafone ~ 120 kbit/s ~ 180 min. InternetCafe Munich EBU WiFi ~ 870 kbit/s ~ 3.55 Mbit/s ~ 30 min. ~ 5.6 min. 56 kbit/s 128 kbit/s 640 kbit/s 4.6 Mbit/s 50 Mbit/s 40 Mbit/s 200 Mbit/s Arcor Business T-Home/IRT ~ 1.95 Mbit/s ~ 2.94 Mbit/s ~ 10 min. ~ 6.7 min
EBU Networks 2008 Seminar / 23 - 24 June 2008 Reproduction prohibited without written permission of the EBU Technical Department & EBU International Training