AES67

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AES67
Manufacturer Info
ManufacturerAudio Engineering Society
Development dateSeptember 2013; 6 years ago (September 2013)[1]
Network Compatibility
SwitchableYes
RoutableYes
Ethernet data ratesFast Ethernet, Gigabit Ethernet, 5GBASE-T, 10 Gigabit Ethernet
Audio Specifications
Minimum latency125 μs to 4 ms
Maximum channels per link120
Maximum sampling rate48, 44.1, or 96 kHz[1]
Maximum bit depth16 or 24 bits[1]

AES67 is a technical standard for audio over IP and audio over Ethernet (AoE) interoperability. The standard was developed by the Audio Engineering Society and first published in September 2013. It is a layer 3 protocol suite based on existing standards and is designed to allow interoperability between various IP-based audio networking systems such as RAVENNA, Livewire, Q-LAN and Dante.

AES67 promises interoperability between previously competing networked audio systems[2] and long-term network interoperation between systems.[3] It also provides interoperability with layer 2 technologies, like Audio Video Bridging (AVB).[4][5][6]Since its publication, AES67 has been implemented independently by several manufacturers and adopted by many others.

Overview[edit]

AES67 defines requirements for synchronizing clocks, setting QoS priorities for media traffic, and initiating media streams with standard IETF Internet protocols. AES67 also defines audio sample format and sample rate, supported number of channels, as well as IP data packet size and latency/buffering requirements.

The standard deliberately does not define device discovery, connection management and media configuration protocols.

Synchronization[edit]

AES67 uses IEEE 1588-2008 Precision Time Protocol (PTPv2) for clock synchronisation. For standard networking equipment, AES67 defines configuration parameters for a "PTP profile for media applications", based on IEEE 1588 delay request-response sync and (optionally) peer-to-peer sync (IEEE 1588 Annexes J.3 and J4); event messages are encaplusated in IPv4 packets over UDP transport (IEEE 1588 Annex D). Some of the default parameters are adjusted, specifically logSyncInterval and logMinDelayReqInterval are reduced to improve accuracy and startup time. Clock Grade 2 as defined in AES11 Digital Audio Reference Signal (DARS) is signalled with clockClass.

Network equipment conforming to IEEE 1588-2008 uses default PTP profiles; for video streams, SMPTE 2059-2 PTP profile can be used.

In AVB/TSN networks, synchronization is achieved with IEEE 802.1AS profile for Time-Sensitive Applications.

The media clock is based on synchronized network time with an IEEE 1588 epoch (1 January 1970 00:00:00 TAI). Clock rates are fixed at audio sampling frequencies of 44,1 kHz, 48 kHz and 96 kHz (i.e. thousand samples per second). RTP transport works with a fixed time offset to network clock.

Transport[edit]

Media data are transported in IPv4 packets, though packet fragmentation / reassembly is not endorsed.

Real-time Transport Protocol with RTP Profile for Audio and Video (L24 and L16 formats) is used over UDP transport. RTP payload is limited to 1460 bytes, to prevent fragmentation with default Ethernet MTU of 1500 bytes (after subtracting IP/UDP/RTP overhead of 20+8+12=40 Bytes). [7] Contributing source (CSRC) identifiers and TLS encryption are not supported.

Time synchronization, media stream delivery, and discovery protocols may use IP multicasting with IGMPv2 (optionally IGMPv3) negotiation. Each media stream is assigned an unique muticast address (in the range from 239.0.0.0 to 239.255.255.255); only one device can send to this address (many-to-many connections are not supported).

To monitor keepalive status and allocate bandwidth, devices may use RTCP report interval, SIP session timers and OPTIONS ping, or ICMP Echo request (ping).

AES67 uses DiffServ to set QoS traffic priorities in the Differentiated Services Code Point (DSCP) field of the IP packet. Three classes should be suported at a minimum:

QoS classes and DiffServ associations
Class name Traffic type Default DiffServ class (DSCP decimal value)
Clock IEEE 1588-2008 time events * EF (46)
Media RTP / RTCP media streams AF41 (34)
Best effort IEEE 1588-2008 signaling, discovery and connection management DF (0)
  • Announce, Sync, Follow_Up, Delay_Req, Delay_Resp, Pdelay_Req, Pdelay_Resp, Pdelay_Resp_Follow_Up

250 μs maximum delay may be require for time-critical applications to prevent drops of audio. To prioritize critical media streams in a large network, applications may use additional values in the Assured Forwarding class 4 with low-drop probability (AF41), typically implemented as a weighted round robin queue. Clock traffic is assigned to the Expedited Forwarding (EF) class, which typically implements strict priority per-hop behavior (PHB). All other traffic is handled on a best effort basis with Default Forwarding.

RTP Clock Source Signalling procedure is used to specify PTP domaind and grandmaster ID for each media stream.

Audio encoding[edit]

Sample formats include 16-bit and 24-bit Linear PCM with 48 kHz sampling frequency, and optional 24-bit 96 kHz and 16-bit 44.1 kHz. Other RTP audio video formats may be supported. Multiple sample frequences are optional. Devices may enforce a global sample frequency setting.

Media packets are scheduled according to 'packet time' - transmission duration of a standard Ethernet packet. Packet time is negotiated by the stream source for each streaming session. Short packet times provide low latency and high transmission rate, but introduce high overhead and require high-performance equipment and links. Long packet times increase latencies and require more buffering. A range from 125 μs to 4 ms is defined, though it is recommended that devices shall adapt to packet time changes and/or determine packet time by analyzing RTP timestamps.

Packet time determines RTP payload size according to a supported sample rate. 1 ms is required for all devices. Devices should support a minimum of 1 to 8 channels per stream. [7]

Recommended packet times
Packet time Samples per packet Notes
48 / 44.1 kHz 96 kHz
125 μs 6 12 Compatible with AVB Class A
250 μs 12 24 High-performance low-latency operation. Compatible with AVB Class B, interoperable with AVB Class A
33313 μs 16 32 Efficient low-latency operation
1 ms 48 96 Required packet time for all devices
4 ms 192 384 Wide area networks, networks with limited QoS capabilities, or interoperability with EBU 3326
  • MTU size restrictions limit a 96 kHz audio stream using 4-ms packet time to a single channel.
Maximum channels per stream
Audio format Packet time
125 μs 250 μs 33313 μs 1 ms 4 ms
16-bit 48 kHz 120 60 45 15 3
24-bit 48 kHz 80 40 30 10 2
24-bit, 96 kHz 40 20 15 5 1

Latency[edit]

Network latency (link offset) is the time difference between the moment an audio stream enters the source (ingress time), marked by RTP timesta in the media packet, and the moment it leaves the destination (egress time). Latency depends on packet time, propagation and queuing delays, packet processing overhead, and buffering in the destination device; thus minimum latency is the shortest packet time and network forwarding time, which can be less than 1 μs on a point-to-point Gigabit Ethernet link with minimum packet size, but in real-world networks could be twice the packet time.

Small buffers decrease latency but may result drops of audio when media data does not arrive on time. Unexpected changes to network conditions and jitter from packet encoding and processing may may require longer buffering and therefore higher latency. Destinations are required to use a buffer of 3 times the packet time, though at least 20 times the packet time (or 20 ms if smaller) is recommended. Sources are required to maintain transmission with jitter of less than 17 packet times (or 17 ms if shorter), though 1 packet time (or 1 ms if shorter) is recommended.

Interoperability with AVB[edit]

AES67 may transport media streams as IEEE 802.1BA AVB time-sensitive traffic Classes A and B on supported networks, with guaranteed latency of 2 ms and 50 ms respectively. Reservation of bandwidth with the Stream Reservation Protocol (SRP) specifies the amount of traffic generated through a measurement interval of 125 μs and 250 μs respectively. Multicast IP addresses have to be used, though only with a single source, as AVB networks only support Ethernet multicast destination addressing in the range from 01:00:5e:00:00:00 to 01:00:5e:7f:ff:ff.

An SRP talker advertise message shall be mapped as follows:

Talker advertise message
StreamID A 64-bit globally-unique ID (48-bit Ethernet MAC address of the source and 16-bit unique source stream ID).
Stream destination address Ethernet multicast destination address.
VLAN ID 12-bit IEEE 802.1Q VLAN tag. Default VLAN identifier for AVB streams is 2.
MaxFrameSize The maximum size of the media stream packets, including the IP header but excluding Ethernet overhead.
MaxIntervalFrames Maximum number of frames a source may transmit in one measurement interval. Since allowed packet times are greater than (or equal to) AVB measurement intervals, this is always 1.
Data Frame Priority 3 for Class A, 2 for Class B.
Rank 1 for normal traffic, 0 for emergency traffic.

Under both IEEE 1588-2008 and IEEE 802.1AS, a PTP clock can be designated as an ordinary clock (OC), boundary clock (BC) or transparent clock (TC), though 802.1AS transparent clocks also have some boundary clock capabilities. A device may implement one or more of these capabilities. OC may have as few as one port (network connection), while TC and BC must have two or more ports. BC and OC ports can work as a master (grandmaster) or a slave. An IEEE 1588 profile is associated with each port. TC can belong to multiple clock domains and profiles. These provisions make it possible to synchronize IEEE 802.1AS clocks to IEEE 1588-2008 clocks used by AES67.

Development history[edit]

The standard was developed by the Audio Engineering Society beginning at the end of 2010.[8] The standard was initially published September 2013.[9][10][11][12] A second printing including a patent statement from Audinate was published in March 2014. An update including clarifications and error corrections was issued in September 2015.[1]

The Media Networking Alliance was formed in October 2014 to promote adoption of AES67.[13]

In October 2014 a plugfest was held to test interoperability achieved with AES67.[14][15] A second plugfest was conducted in November 2015[16] and third in February 2017.[17]

In May 2016, the AES published a report describing synchronization interoperability between AES67 and SMPTE 2059-2.[18]

In June 2016, AES67 audio transport enhanced by AVB/TSN clock synchronisation and bandwidth reservation was demoed at InfoComm 2016.[19]

In September 2017, SMPTE published ST 2110, a standard for professional video over IP.[20] ST 2110-30 uses AES67 as the transport for audio accompanying the video.[21]

In December 2017 the Media Networking Alliance merged with the Alliance for IP Media Solutions (AIMS) combining efforts to promote standards-based network transport for audio and video.[22]

In April 2018 AES67-2018 was published. The principal change in this revision is addition of a protocol implementation conformance statement (PICS).[23]

Adoption[edit]

The standard has been implemented by Lawo,[24] Axia,[25] AMX (in SVSI devices), Wheatstone,[26][27] Extron Electronics, Riedel,[28] Coveloz Technologies[permanent dead link],[29] ALC NetworX,[30] Audinate,[31][32][33][34][35][36] Archwave,[37] Digigram,[38] Sonifex,[39] Yamaha,[40] QSC,[41] Neutrik, Attero Tech,[42] Merging Technologies,[43][44][45][46] Gallery SIENNA,[47] and is supported by RAVENNA-enabled devices under its AES67 Operational Profile.[48]

Shipping products[edit]

Over time this table will grow to become a resource for integration and compatibility between devices. The discovery methods supported by each device are critical for integration since the AES67 specification does not stipulate how this should be done, but instead provides a variety of options or suggestions. Also, AES67 specifies Multicast or Unicast but many AES67 devices only support Multicast.

Vendor Product Description OS Platform AES67 Model Send Receive Multicast Unicast Notes
Merging Technologies Virtual Audio Device[49] Ravenna/AES67 drivers macOS[50], Linux[51], Windows Ravenna AES67 SAP, mDNS/RTSP SAP, mDNS/RTSP Y Y Free
ALC Networks Virtual Sound Card[52] Ravenna/AES67 WDM driver Windows Ravenna AES67 Y Free
ALC Networks RAV2SAP[53] AES67 Discovery Tools Windows Ravenna AES67 SAP mDNS/RTSP Y Free
Sienna AES67 to NDI Gateway[54] AES67 to NDI Gateway macOS, Linux, Windows Native AES67 SAP SAP Y N
Sienna NDI to AES67[55] NDI to AES67 Sender macOS, Linux Native AES67 SAP SAP Y N
Lawo VRX4[56] Audio Mixer Windows Ravenna AES67 Y
Hasseb AoE[57] Analog and optical AES67 Interface Native AES67 mDNS/RTSP mDNS/RTSP Y Y
QSC DSP, Amplifiers[58] various Q-SYS AES67 SAP SAP Y
AXIA Various[59] various Livewire+ AES67 Y Y
Yamaha Mixers[60] various Dante AES67 SAP SAP Y N
Attero Tech Endpoints [61] Endpoints Attero AES67 SAP SAP Y N
SoundTube Entertainment Various[62] Various Dante AES67

References[edit]

  1. ^ a b c d "AES67-2013: AES standard for audio applications of networks - High-performance streaming audio-over-IP interoperability". Audio Engineering Society. 2013-09-11. Retrieved 2014-02-11.
  2. ^ Steve Harvey (2014-06-27). "NAB Show Product Review: Audio". TV Technology. Archived from the original on 2016-03-03. Retrieved 2014-06-29. Cite journal requires |journal= (help)
  3. ^ Dave Davies (2014-07-22). "Mark Yonge on the new dawn for networking". Installation. Archived from the original on 2014-07-28. Retrieved 2014-07-23. Cite journal requires |journal= (help)
  4. ^ AES67-2018 – Annex C (Informative) – AVB network transport
  5. ^ AES67-2018 – Annex D (Informative) – Interfacing to IEEE 802.1AS clock domains
  6. ^ https://www.smpte.org/sites/default/files/users/user27446/AES67%20for%20Audio%20Production-Background%20Applications%20and%20Challenges.pdf
  7. ^ a b AES67-101: The Basics of AES67. Anthony P. Kuzub
  8. ^ AES-X192 initiation, Audio Engineering Society, 2010-12-01
  9. ^ Dan Daley (October 2013). "AES Throws New Audio Networking Standard Into the Ring". Retrieved 2014-02-11.
  10. ^ Dan Daley (2013-09-16). "AES Announces AES67-2013 Networked Audio-Over-IP Interoperability Standard". Retrieved 2014-02-11. Cite journal requires |journal= (help)
  11. ^ "AES Announces New Networked Audio-Over-IP Interoperability Standard: AES67-2013". ProSoundWeb. 2013-09-12. Retrieved 2014-02-11.
  12. ^ "AES Announces New Networked Audio-Over-IP Interoperability Standard: AES67-2013". Radio. 2013-09-16. Archived from the original on 2017-02-17. Retrieved 2014-02-11. Cite journal requires |journal= (help)
  13. ^ "AES Show Introduces Media Networking Alliance". Radio World. 2014-10-06. Archived from the original on 2015-09-24. Retrieved 2014-11-11.
  14. ^ Jon Chapple (27 November 2014), "AES Standards Committee, EBU test AES67 at PlugFest", PSN Europe, archived from the original on 4 December 2014, retrieved 29 November 2014
  15. ^ AES-R12-2014: Standards project report - AES67 Interoperability PlugFest - Munich 2014, Audio Engineering Society, 2014-11-24
  16. ^ AES-R15-2015: Standards project report - AES67 Interoperability PlugFest - Washington 2015, Audio Engineering Society, 2016-01-02
  17. ^ AES-R17-2017: Standards project report - AES67 Interoperability PlugFest - London 2017, Audio Engineering Society, 2017-04-28
  18. ^ AES-R16-2016: AES Standards Report - PTP parameters for AES67 and SMPTE ST 2059-2 interoperability, Audio Engineering Society, 2016-05-02
  19. ^ Joao Martins (2016-06-16). "AVB/TSN Momentum and AES67/AVB Harmony at InfoComm 2016". Retrieved 2016-12-08.
  20. ^ "SMPTE Approves ST 2110-30 Standards for Professional Media Over Managed IP Networks". Retrieved 2017-11-30.
  21. ^ Leigh Whitcomb (30 June 2017), "Audio for Television: How AES67 and Uncompressed 2022/2110/TR03 Video Fit Together", SMPTE Motion Imaging Journal, SMPTE, doi:10.5594/JMI.2017.2703479
  22. ^ Michelle Clancy (28 December 2017), AIMS, Media Networking Alliance merge, Rapid TV News
  23. ^ "AES67-2018: AES standard for audio applications of networks - High-performance streaming audio-over-IP interoperability Published". 2018-04-24.
  24. ^ Lawo. "Lawo supports successful AES67 Interoperability Demo during AES Convention in New York". www.lawo.com. Retrieved 2017-10-26.
  25. ^ "Axia Announces First Broadcast Product with AES67 Compliance". Sound & Picture. 2013-11-14. Retrieved 2014-02-11. Cite journal requires |journal= (help)
  26. ^ "IP Audio Takes a Big Step Forward". Radio World. 2014-02-21. Archived from the original on 2015-09-24. Retrieved 2014-06-18.
  27. ^ Steve Harvey (2014-08-11). "Standardizing AoIP is Enabling Interoperability". TV Technology. Archived from the original on 2014-08-13. Retrieved 2014-08-13. Cite journal requires |journal= (help)
  28. ^ "Reidel to make a splash at SATIS 2014". Digital Production. Oct 29, 2014. Retrieved 2014-11-11.
  29. ^ "Coveloz Bach: World's first AES67 endpoint to gain AVnu Certification'". Pro-Audio Central. 6 January 2016. Retrieved 2016-02-06.
  30. ^ "ALC NetworX Shows Ravenna, AES67". Radio World. 2014-01-29. Archived from the original on 2015-09-24. Retrieved 2014-02-11. Cite journal requires |journal= (help)
  31. ^ https://www.audinate.com/faq-catagories/aes67
  32. ^ "AUDINATE TO SUPPORT AES67 STANDARD". InAVate. 2014-02-04. Archived from the original on 2014-02-23. Retrieved 2014-02-11. Cite journal requires |journal= (help)
  33. ^ "Audinate Announces Support for AES67 Standard". Sound Forums Network. 2014-02-04. Archived from the original on 2014-02-22. Retrieved 2014-02-11.
  34. ^ "Audinate announces support for AES67 standard". Pro Audio Central. 2014-02-04. Archived from the original on 2014-02-21. Retrieved 2014-02-11.
  35. ^ "Dante to Add AES67 Transport". ProSound News. 2014-02-05. Archived from the original on 2015-09-27. Retrieved 2014-02-11. Cite journal requires |journal= (help)
  36. ^ Michael Williams (April 8, 2015), Audinate Announces Availability of Firmware Update to Support AES67, rAVe
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  39. ^ "Archived copy". Archived from the original on 2016-02-07. Retrieved 2016-05-17.CS1 maint: archived copy as title (link)
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  41. ^ "QSC Q-SYS Platform Software Release to Support AES67". QSC. December 7, 2016.
  42. ^ Attero Tech Ships AES67 Networked Audio Products, retrieved 2017-12-17
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  49. ^ https://www.merging.com/products/interfaces/for-any-daw
  50. ^ https://www.merging.com/products/aes67_vad_standard
  51. ^ https://www.merging.com/products/alsa_ravenna_aes67_driver
  52. ^ https://www.ravenna-network.com/2013/09/13/free-version-of-ravenna-virtual-sound-card-for-windows-now-available-for-download/
  53. ^ https://www.ravenna-network.com/aes67/rav2sap/
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  55. ^ http://www.sienna-tv.com/ndi/ndiprocessor.html
  56. ^ http://www.r3lay.com/product/vrx4-virtual-radio-mixer-software/
  57. ^ http://hasseb.fi/shop2/index.php?route=product/product&product_id=65
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  59. ^ https://www.telosalliance.com/Axia/Livewire-AoIP-Networking
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  62. ^ "Series". Soundtube Entertainment. Retrieved 2019-04-11.

External links[edit]