Internet DRAFT - draft-gunther-detnet-proaudio-req
draft-gunther-detnet-proaudio-req
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
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
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
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
Gunther & Grossman Expires October 2, 2015 [Page 1]
Internet-Draft DetNet Pro Audio requirements March 2015
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
Gunther & Grossman Expires October 2, 2015 [Page 2]
Internet-Draft DetNet Pro Audio requirements March 2015
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.
Gunther & Grossman Expires October 2, 2015 [Page 3]
Internet-Draft DetNet Pro Audio requirements March 2015
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
Gunther & Grossman Expires October 2, 2015 [Page 4]
Internet-Draft DetNet Pro Audio requirements March 2015
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
Gunther & Grossman Expires October 2, 2015 [Page 5]
Internet-Draft DetNet Pro Audio requirements March 2015
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
Gunther & Grossman Expires October 2, 2015 [Page 6]
Internet-Draft DetNet Pro Audio requirements March 2015
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.
Gunther & Grossman Expires October 2, 2015 [Page 7]
Internet-Draft DetNet Pro Audio requirements March 2015
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.
Gunther & Grossman Expires October 2, 2015 [Page 8]
Internet-Draft DetNet Pro Audio requirements March 2015
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
Gunther & Grossman Expires October 2, 2015 [Page 9]
Internet-Draft DetNet Pro Audio requirements March 2015
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.
Gunther & Grossman Expires October 2, 2015 [Page 10]
Internet-Draft DetNet Pro Audio requirements March 2015
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
Gunther & Grossman Expires October 2, 2015 [Page 11]
Internet-Draft DetNet Pro Audio requirements March 2015
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
Gunther & Grossman Expires October 2, 2015 [Page 12]
