Difference between revisions of "AES67"
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Because TTLs are difficult to manage in practice, nowadays IP routing rules is the preferred choice for confining Multicast, based on "scoping", typically used inside the "administratively scoped IPv4 multicast space" (239.0.0.0 to 239.255.255.255) which is the range normally used by AES67 audio packets (RFC2365). In this case the TTL may be forced to 255. | Because TTLs are difficult to manage in practice, nowadays IP routing rules is the preferred choice for confining Multicast, based on "scoping", typically used inside the "administratively scoped IPv4 multicast space" (239.0.0.0 to 239.255.255.255) which is the range normally used by AES67 audio packets (RFC2365). In this case the TTL may be forced to 255. | ||
| − | CAUTION : The TTL always has to be checked against your network governance, especially when Multicast Routing is used. | + | <pre style="color: red">CAUTION : The TTL always has to be checked against your network governance, especially when Multicast Routing is used.</pre> |
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===QOS=== | ===QOS=== | ||
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Revision as of 14:19, 7 July 2021
Contents
Generalities
These docs are useful :
AES67 Practical Guide (from RAVENNA)
File:AES67 Practical Guide.pdf
SDP
The SDP (Session Description Protocol) is a mandatory AES67 data that gives all useful information to connect to an audio/video stream. It is usually automatically exchanged transparently by devices but can be used to manually connect to an audio/video stream.
The SDP file for generated AES67 streams can be downloaded on the SOUND4 device web server at : "http://<ULA IP address>:8554/by-id/<Stream number>"
with <ULA IP address> = IP address of the AES67 Ethernet Port of the SOUND4 device
and <Stream number> = from 1 to 8
eg http://192.168.6.50:8554/by-id/1
The UDP file can be saved with .udp extension and then directly opened in VLC for instance (Media->open file).
The lines of interest are :
- "c=IN IP4 <IP address>/<TTL>" gives the stream's Multicast IP address
- "m=audio <udp port> RTP/AVP <payload type>" for the UDP port and PayloadType
- "a=rtpmap:<payload type> <format>/<samplerate>/<n channels>" for the Sample Format (usually "L24") and Number of Channels (usually 2 ; if the stream has more than 2 only the first two will be used on SOUND4 device)
Transport
IPv4 Multicast addresses
AES67 does not automatically attribute multicast address, it's under user's responsibility.
Multicast address range is : 224.0.0.0/4. The group includes the addresses from 224.0.0.0 to 239.255.255.255 (cf https://en.wikipedia.org/wiki/Multicast_address)
For Audio over IP usage it's recommended to use 239.x.y.z range ("Administratively scoped, private use within an organization").
Actually some usual sub-ranges are :
- 239.192.y.z => Livewire (Livestream and Standard Stereo Streams)
- 239.192.255.1 to 4 are Livewire clock/advertising/GPIO
- 239.193.y.z => Livewire (Back Standard Streams)
- 239.195.y.z => Livewire (Back Live Streams)
- 239.255.255.255 => SAP (advertisement) Dante
Never use 224.x.y.z range which is reserved for Network administration and control.
Windows
Under Windows, there may be difficulties with multicast routing when more than one Network Interfaces is used.
UDP Port
Usually port 5004 is used for RTP (audio) and 5005 for RTCP (control) (cf http://www.networksorcery.com/enp/protocol/ip/ports05000.htm) but you may choose another number:
- RTP port must be even
- RTCP port = RTP port + 1
- don't use port under 1024 which are Unix "system ports"
Note : RTCP is not widely used except by RAVENNA
IP TTL
Time To Live was a way to restrict multicast packets to a subnet. This is important because AoIP bandwidth is huge and audio packets should not flood away and overload a company's network.
TTL=1 is a widely used value but it may disturb some CISCO routers, this is why SOUND4 devices have a default TTL=2 value.
Because TTLs are difficult to manage in practice, nowadays IP routing rules is the preferred choice for confining Multicast, based on "scoping", typically used inside the "administratively scoped IPv4 multicast space" (239.0.0.0 to 239.255.255.255) which is the range normally used by AES67 audio packets (RFC2365). In this case the TTL may be forced to 255.
CAUTION : The TTL always has to be checked against your network governance, especially when Multicast Routing is used.
QOS
Clocking
AES67 uses PTP (Precise Time Protocol, IEEE1588v2-2008) as clock reference for sub-sample synchronization.
You have to configure some parameters, which are normally chosen according to a "PTP Profile" (1588, AES67, ST2110) and must be the same for all the audio devices of your network.
These parameters are critical for good and fast synchronization, especially when a Master Clock is disappearing and another one takes precedence.
Parameters :
- domain [0..255] : choose a common number, 0 is AES67 default, 127 is SMPTE ST2110 default
- priority1 and priority2 [0..255] : (for Master only) used to elect the clock Master. Lower values take precedence. priority1 is used first, then Clock Quality, then priority2. Default is 128,128.
- sync delay (in seconds) : (for Master only) mean delay between SYNC messages. Default is 0.125s
- announce interval (in seconds) : (for Master only) mean delay between ANNOUNCE messages. Default is 2s
- announce timeout (in packets) : number of announce interval that has to pass before timeout (which launch a Best Master election). Default is 3 packets.
- request interval (in seconds) : (for Slave only) mean delay between DELAY_REQ messages. Default is 1s
Interconnecting...
Troubleshooting
See LANAUDIO
