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Versions: 00 01

Internet Engineering Task Force                          C. Gunther, Ed.
Internet-Draft                                                    HARMAN
Intended status: Informational                          E. Grossman, Ed.
Expires: October 2, 2015                                           DOLBY
                                                          March 31, 2015


        Deterministic Networking Professional Audio Requirements
                  draft-gunther-detnet-proaudio-req-01

Abstract

   This draft documents the needs in the professional audio and video
   industry to establish multi-hop paths and optional redundant paths
   for characterized flows with deterministic properties.  In this
   context deterministic implies that streams can be established which
   provide guaranteed bandwidth and latency which can be established
   from a Layer 3 (IP) interface.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on October 2, 2015.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Requirements Language . . . . . . . . . . . . . . . . . . . .   3
   3.  Fundamental Stream Requirements . . . . . . . . . . . . . . .   3
     3.1.  Guaranteed Bandwidth  . . . . . . . . . . . . . . . . . .   4
     3.2.  Bounded and Consistent Latency  . . . . . . . . . . . . .   4
       3.2.1.  Optimizations . . . . . . . . . . . . . . . . . . . .   5
   4.  Additional Stream Requirements  . . . . . . . . . . . . . . .   6
     4.1.  Deterministic Time to Establish Streaming . . . . . . . .   6
     4.2.  Use of Unused Reservations by Best-Effort Traffic . . . .   6
     4.3.  Layer 3 Interconnecting Layer 2 Islands . . . . . . . . .   7
     4.4.  Secure Transmission . . . . . . . . . . . . . . . . . . .   7
     4.5.  Redundant Paths . . . . . . . . . . . . . . . . . . . . .   7
     4.6.  Link Aggregation  . . . . . . . . . . . . . . . . . . . .   8
     4.7.  Traffic Segregation . . . . . . . . . . . . . . . . . . .   8
       4.7.1.  Packet Forwarding Rules, VLANs and Subnets  . . . . .   8
       4.7.2.  Multicast Addressing (IPv4 and IPv6)  . . . . . . . .   8
   5.  Integration of Reserved Streams into IT Networks  . . . . . .   9
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .   9
     6.1.  Denial of Service . . . . . . . . . . . . . . . . . . . .   9
     6.2.  Control Protocols . . . . . . . . . . . . . . . . . . . .   9
   7.  A State-of-the-Art Broadcast Installation Hits Technology
       Limits  . . . . . . . . . . . . . . . . . . . . . . . . . . .  10
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  10
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  10
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  10
     10.2.  Informative References . . . . . . . . . . . . . . . . .  11
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  11

1.  Introduction

   The professional audio and video industry includes music and film
   content creation, broadcast, cinema, and live exposition as well as
   public address, media and emergency systems at large venues
   (airports, stadiums, churches, theme parks).  These industries have
   already gone through the transition of audio and video signals from
   analog to digital, however the interconnect systems remain primarily
   point-to-point with a single (or small number of) signals per link,
   interconnected with purpose-built hardware.

   These industries are now attempting to transition to packet based
   infrastructure for distributing audio and video in order to reduce




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   cost, increase routing flexibility, and integrate with existing IT
   infrastructure.

   However, there are several requirements for making a network the
   primary infrastructure for audio and video which are not met by
   todays networks and these are our concern in this draft.

   The principal requirement is that pro audio and video applications
   become able to establish streams that provide guaranteed (bounded)
   bandwidth and latency from the Layer 3 (IP) interface.  Such streams
   can be created today within standards-based layer 2 islands however
   these are not sufficient to enable effective distribution over wider
   areas (for example broadcast events that span wide geographical
   areas).

   Some proprietary systems have been created which enable deterministic
   streams at layer 3 however they are engineered networks in that they
   require careful configuration to operate, often require that the
   system be over designed, and it is implied that all devices on the
   network voluntarily play by the rules of that network.  To enable
   these industries to successfully transition to an interoperable
   multi-vendor packet-based infrastructure requires effective open
   standards, and we believe that establishing relevant IETF standards
   is a crucial factor.

   It would be highly desirable if such streams could be routed over the
   open Internet, however even intermediate solutions with more limited
   scope (such as enterprise networks) can provide a substantial
   improvement over todays networks, and a solution that only provides
   for the enterprise network scenario is an acceptable first step.

   We also present more fine grained requirements of the audio and video
   industries such as safety and security, redundant paths, devices with
   limited computing resources on the network, and that reserved stream
   bandwidth is available for use by other best-effort traffic when that
   stream is not currently in use.

2.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

3.  Fundamental Stream Requirements

   The fundamental stream properties are guaranteed bandwidth and
   deterministic latency as described in this section.  Additional
   stream requirements are described in a subsequent section.



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3.1.  Guaranteed Bandwidth

   Transmitting audio and video streams is unlike common file transfer
   activities because guaranteed delivery cannot be achieved by re-
   trying the transmission; by the time the missing or corrupt packet
   has been identified it is too late to execute a re-try operation and
   stream playback is interrupted, which is unacceptable in for example
   a live concert.  In some contexts large amounts of buffering can be
   used to provide enough delay to allow time for one or more retries,
   however this is not an effective solution when live interaction is
   involved, and is not considered an acceptable general solution for
   pro audio and video.  (Have you ever tried speaking into a microphone
   through a sound system that has an echo coming back at you?  It makes
   it almost impossible to speak clearly).

   Providing a way to reserve a specific amount of bandwidth for a given
   stream is a key requirement.

3.2.  Bounded and Consistent Latency

   Latency in this context means the amount of time that passes between
   when a signal is sent over a stream and when it is received, for
   example the amount of time delay between when you speak into a
   microphone and when your voice emerges from the speaker.  Any delay
   longer than about 10-15 milliseconds is noticeable by most live
   performers, and greater latency makes the system unusable because it
   prevents them from playing in time with the other players (see slide
   6 of [SRP_LATENCY]).

   The 15ms latency bound is made even more challenging because it is
   often the case in network based music production with live electric
   instruments that multiple stages of signal processing are used,
   connected in series (i.e. from one to the other for example from
   guitar through a series of digital effects processors) in which case
   the latencies add, so the latencies of each individual stage must all
   together remain less than 15ms.

   In some situations it is acceptable at the local location for content
   from the live remote site to be delayed to allow for a statistically
   acceptable amount of latency in order to reduce jitter.  However,
   once the content begins playing in the local location any audio
   artifacts caused by the local network are unacceptable, especially in
   those situations where a live local performer is mixed into the feed
   from the remote location.

   In addition to being bounded to within some predictable and
   acceptable amount of time (which may be 15 milliseconds or more or
   less depending on the application) the latency also has to be



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   consistent.  For example when playing a film consisting of a video
   stream and audio stream over a network, those two streams must be
   synchronized so that the voice and the picture match up.  A common
   tolerance for audio/video sync is one NTSC video frame (about 33ms)
   and to maintain the audience perception of correct lip sync the
   latency needs to be consistent within some reasonable tolerance, for
   example 10%.

   A common architecture for synchronizing multiple streams that have
   different paths through the network (and thus potentially different
   latencies) is to enable measurement of the latency of each path, and
   have the data sinks (for example speakers) buffer (delay) all packets
   on all but the slowest path.  Each packet of each stream is assigned
   a presentation time which is based on the longest required delay.
   This implies that all sinks must maintain a common time reference of
   sufficient accuracy, which can be achieved by any of various
   techniques.

   This type of architecture is commonly implemented using a central
   controller that determines path delays and arbitrates buffering
   delays.

3.2.1.  Optimizations

   The controller might also perform optimizations based on the
   individual path delays, for example sinks that are closer to the
   source can inform the controller that they can accept greater latency
   since they will be buffering packets to match presentation times of
   farther away sinks.  The controller might then move a stream
   reservation on a short path to a longer path in order to free up
   bandwidth for other critical streams on that short path.  See slides
   3-5 of [SRP_LATENCY].

   Additional optimization can be achieved in cases where sinks have
   differing latency requirements, for example in a live outdoor concert
   the speaker sinks have stricter latency requirements than the
   recording hardware sinks.  See slide 7 of [SRP_LATENCY].

   Device cost can be reduced in a system with guaranteed reservations
   with a small bounded latency due to the reduced requirements for
   buffering (i.e. memory) on sink devices.  For example, a theme park
   might broadcast a live event across the globe via a layer 3 protocol;
   in such cases the size of the buffers required is proportional to the
   latency bounds and jitter caused by delivery, which depends on the
   worst case segment of the end-to-end network path.  For example on
   todays open internet the latency is typically unacceptable for audio
   and video streaming without many seconds of buffering.  In such
   scenarios a single gateway device at the local network that receives



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   the feed from the remote site would provide the expensive buffering
   required to mask the latency and jitter issues associated with long
   distance delivery.  Sink devices in the local location would have no
   additional buffering requirements, and thus no additional costs,
   beyond those required for delivery of local content.  The sink device
   would be receiving the identical packets as those sent by the source
   and would be unaware that there were any latency or jitter issues
   along the path.

4.  Additional Stream Requirements

   The requirements in this section are more specific yet are common to
   multiple audio and video industry applications.

4.1.  Deterministic Time to Establish Streaming

   Some audio systems installed in public environments (airports,
   hospitals) have unique requirements with regards to health, safety
   and fire concerns.  One such requirement is a maximum of 3 seconds
   for a system to respond to an emergency detection and begin sending
   appropriate warning signals and alarms without human intervention.
   For this requirement to be met, the system must support a bounded and
   acceptable time from a notification signal to specific stream
   establishment.  For further details see [ISO7240-16].

   Similar requirements apply when the system is restarted after a power
   cycle, cable re-connection, or system reconfiguration.

   In many cases such re-establishment of streaming state must be
   achieved by the peer devices themselves, i.e. without a central
   controller (since such a controller may only be present during
   initial network configuration).

   Video systems introduce related requirements, for example when
   transitioning from one camera feed to another.  Such systems
   currently use purpose-built hardware to switch feeds smoothly,
   however there is a current initiative in the broadcast industry to
   switch to a packet-based infrastructure (see [STUDIO_IP] and the ESPN
   DC2 use case described below).

4.2.  Use of Unused Reservations by Best-Effort Traffic

   In cases where stream bandwidth is reserved but not currently used
   (or is under-utilized) that bandwidth must be available to best-
   effort (i.e. non-time-sensitive) traffic.  For example a single
   stream may be nailed up (reserved) for specific media content that
   needs to be presented at different times of the day, ensuring timely
   delivery of that content, yet in between those times the full



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   bandwidth of the network can be utilized for best-effort tasks such
   as file transfers.

   This also addresses a concern of IT network administrators that are
   considering adding reserved bandwidth traffic to their networks that
   users will just reserve a ton of bandwidth and then never un-reserve
   it even though they are not using it, and soon they will have no
   bandwidth left.

4.3.  Layer 3 Interconnecting Layer 2 Islands

   As an intermediate step (short of providing guaranteed bandwidth
   across the open internet) it would be valuable to provide a way to
   connect multiple Layer 2 networks.  For example layer 2 techniques
   could be used to create a LAN for a single broadcast studio, and
   several such studios could be interconnected via layer 3 links.

4.4.  Secure Transmission

   Digital Rights Management (DRM) is very important to the audio and
   video industries.  Any time protected content is introduced into a
   network there are DRM concerns that must be maintained (see
   [CONTENT_PROTECTION]).  Many aspects of DRM are outside the scope of
   network technology, however there are cases when a secure link
   supporting authentication and encryption is required by content
   owners to carry their audio or video content when it is outside their
   own secure environment (for example see [DCI]).

   As an example, two techniques are Digital Transmission Content
   Protection (DTCP) and High-Bandwidth Digital Content Protection
   (HDCP).  HDCP content is not approved for retransmission within any
   other type of DRM, while DTCP may be retransmitted under HDCP.
   Therefore if the source of a stream is outside of the network and it
   uses HDCP protection it is only allowed to be placed on the network
   with that same HDCP protection.

4.5.  Redundant Paths

   On-air and other live media streams must be backed up with redundant
   links that seamlessly act to deliver the content when the primary
   link fails for any reason.  In point-to-point systems this is
   provided by an additional point-to-point link; the analogous
   requirement in a packet-based system is to provide an alternate path
   through the network such that no individual link can bring down the
   system.






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4.6.  Link Aggregation

   For transmitting streams that require more bandwidth than a single
   link in the target network can support, link aggregation is a
   technique for combining (aggregating) the bandwidth available on
   multiple physical links to create a single logical link of the
   required bandwidth.  However, if aggregation is to be used, the
   network controller (or equivalent) must be able to determine the
   maximum latency of any path through the aggregate link (see Bounded
   and Consistent Latency section above).

4.7.  Traffic Segregation

   Sink devices may be low cost devices with limited processing power.
   In order to not overwhelm the CPUs in these devices it is important
   to limit the amount of traffic that these devices must process.

   As an example, consider the use of individual seat speakers in a
   cinema.  These speakers are typically required to be cost reduced
   since the quantities in a single theater can reach hundreds of seats.
   Discovery protocols alone in a one thousand seat theater can generate
   enough broadcast traffic to overwhelm a low powered CPU.  Thus an
   installation like this will benefit greatly from some type of traffic
   segregation that can define groups of seats to reduce traffic within
   each group.  All seats in the theater must still be able to
   communicate with a central controller.

   There are many techniques that can be used to support this
   requirement including (but not limited to) the following examples.

4.7.1.  Packet Forwarding Rules, VLANs and Subnets

   Packet forwarding rules can be used to eliminate some extraneous
   streaming traffic from reaching potentially low powered sink devices,
   however there may be other types of broadcast traffic that should be
   eliminated using other means for example VLANs or IP subnets.

4.7.2.  Multicast Addressing (IPv4 and IPv6)

   Multicast addressing is commonly used to keep bandwidth utilization
   of shared links to a minimum.

   Because of the MAC Address forwarding nature of Layer 2 bridges it is
   important that a multicast MAC address is only associated with one
   stream.  This will prevent reservations from forwarding packets from
   one stream down a path that has no interested sinks simply because
   there is another stream on that same path that shares the same
   multicast MAC address.



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   Since each multicast MAC Address can represent 32 different IPv4
   multicast addresses there must be a process put in place to make sure
   this does not occur.  Requiring use of IPv6 address can achieve this,
   however due to their continued prevalence, solutions that are
   effective for IPv4 installations are also required.

5.  Integration of Reserved Streams into IT Networks

   A commonly cited goal of moving to a packet based media
   infrastructure is that costs can be reduced by using off the shelf,
   commodity network hardware.  In addition, economy of scale can be
   realized by combining media infrastructure with IT infrastructure.
   In keeping with these goals, stream reservation technology should be
   compatible with existing protocols, and not compromise use of the
   network for best effort (non-time-sensitive) traffic.

6.  Security Considerations

   Many industries that are moving from the point-to-point world to the
   digital network world have little understanding of the pitfalls that
   they can create for themselves with improperly implemented network
   infrastructure.  DetNet should consider ways to provide security
   against DoS attacks in solutions directed at these markets.  Some
   considerations are given here as examples of ways that we can help
   new users avoid common pitfalls.

6.1.  Denial of Service

   One security pitfall that this author is aware of involves the use of
   technology that allows a presenter to throw the content from their
   tablet or smart phone onto the A/V system that is then viewed by all
   those in attendance.  The facility introducing this technology was
   quite excited to allow such modern flexibility to those who came to
   speak.  One thing they hadn't realized was that since no security was
   put in place around this technology it left a hole in the system that
   allowed other attendees to "throw" their own content onto the A/V
   system.

6.2.  Control Protocols

   Professional audio systems can include amplifiers that are capable of
   generating hundreds or thousands of watts of audio power which if
   used incorrectly can cause hearing damage to those in the vicinity.
   Apart from the usual care required by the systems operators to
   prevent such incidents, the network traffic that controls these
   devices must be secured (as with any sensitive application traffic).
   In addition, it would be desirable if the configuration protocols
   that are used to create the network paths used by the professional



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   audio traffic could be designed to protect devices that are not meant
   to receive high-amplitude content from having such potentially
   damaging signals routed to them.

7.  A State-of-the-Art Broadcast Installation Hits Technology Limits

   ESPN recently constructed a state-of-the-art 194,000 sq ft, $125
   million broadcast studio called DC2.  The DC2 network is capable of
   handling 46 Tbps of throughput with 60,000 simultaneous signals.
   Inside the facility are 1,100 miles of fiber feeding four audio
   control rooms.  (See details at [ESPN_DC2] ).

   In designing DC2 they replaced as much point-to-point technology as
   they possibly could with packet-based technology.  They constructed
   seven individual studios using layer 2 LANS (using IEEE 802.1 AVB)
   that were entirely effective at routing audio within the LANs, and
   they were very happy with the results, however to interconnect these
   layer 2 LAN islands together they ended up using dedicated links
   because there is no standards-based routing solution available.

   This is the kind of motivation we have to develop these standards
   because customers are ready and able to use them.

8.  Acknowledgements

   The editors would like to acknowledge the help of the following
   individuals and the companies they represent:

   Jeff Koftinoff, Meyer Sound

   Jouni Korhonen, Associate Technical Director, Broadcom

   Pascal Thubert, CTAO, Cisco

   Kieran Tyrrell, Sienda New Media Technologies GmbH

9.  IANA Considerations

   This memo includes no request to IANA.

10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.





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10.2.  Informative References

   [CONTENT_PROTECTION]
              Olsen, D., "1722a Content Protection", 2012,
              <http://grouper.ieee.org/groups/1722/contributions/2012/
              avtp_dolsen_1722a_content_protection.pdf>.

   [DCI]      Digital Cinema Initiatives, LLC, "DCI Specification,
              Version 1.2", 2012, <http://www.dcimovies.com/>.

   [ESPN_DC2]
              Daley, D., "ESPN's DC2 Scales AVB Large", 2014,
              <http://sportsvideo.org/main/blog/2014/06/
              espns-dc2-scales-avb-large>.

   [ISO7240-16]
              ISO, "ISO 7240-16:2007 Fire detection and alarm systems --
              Part 16: Sound system control and indicating equipment",
              2007, <http://www.iso.org/iso/
              catalogue_detail.htm?csnumber=42978>.

   [SRP_LATENCY]
              Gunther, C., "Specifying SRP Latency", 2014,
              <http://www.ieee802.org/1/files/public/docs2014/
              cc-cgunther-acceptable-latency-0314-v01.pdf>.

   [STUDIO_IP]
              Mace, G., "IP Networked Studio Infrastructure for
              Synchronized & Real-Time Multimedia Transmissions", 2007,
              <http://www.ieee802.org/1/files/public/docs2047/
              avb-mace-ip-networked-studio-infrastructure-0107.pdf>.

Authors' Addresses

   Craig Gunther (editor)
   Harman International
   10653 South River Front Parkway
   South Jordan, UT  84095
   USA

   Phone: +1 801 568-7675
   Email: craig.gunther@harman.com
   URI:   http://www.harman.com








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   Ethan Grossman (editor)
   Dolby Laboratories, Inc.
   100 Potrero Ave
   San Francisco, CA  94103
   USA

   Phone: +1 415 645 4726
   Email: ethan.grossman@dolby.com
   URI:   http://www.dolby.com










































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