StormAudio - Knowledge Center
AoIP option - Additional information & Troubleshooting
How to deal with unconventional devices
In case of an unconventional situation, it is always possible to set connections manually.
Important Information
All AoIP devices must be on the same network.
All AoIP devices must use the same prefix in its broadcast IP as: 239.IP.xxx.xxx.
All AoIP devices must use the same PTP domain number: 0 by default.
ISP AoIP as a transmitter
Open the WebUI of our ISP AoIP option either using ANEMAN or by clicking on IP in “setting” tab.
Select the “Session sources” tab.
Create a Source by clicking on the connection cable icon (top left), fill in your parameters (IO is the input to be outputted by the option, name is the label of the stream and channel count define the number of channels in the stream).
Download the SDP file describing this source by clicking on the blue link at bottom right of the page. This could be useful to configure other devices.
If the target needs a specific port it is possible to specify the port for the address accordingly (e.g. 239.1.40.12:6520). The port end number must be even (0, 2, 4, 6, 8).
ISP AoIP as a receiver
Open the WebUI of the ISP AoIP option with ANEMAN or by clicking on IP in “setting” Tab.
Select Session sinks tab.
Create a Sink by clicking on the connection cable icon (top left).
Configure the Sink : you can simply name the connection, choose the source and map channels from the source to ISP inputs in the “Channels” section.
You can also click on “Manual” and provide the SDP description for a complete configuration :
Line | Description |
---|---|
v=0 | protocol version |
o=-1 0 IN IP4 169.254.89.126 | device(session creator/owner) IP |
s=Test1 | session name |
t=0 | time the session is active |
a=clock-domain:PTPv2 0 | - |
m=audio 5678RTP/AVP 98 | port:5678 payload type:98 |
c=IN IP4 239.1.40.64 | stream dest IP |
a=rtpmap:98L24/48000/8 | payload type:98 , codec:L24, SR:48000, 8 channels |
a=sync-time:0 | - |
a=framecount:1-48 | samples per frame [1..48] |
a=ptime 1.0 | packet time |
a=mediaclk:direct=0 | - |
a=ts-refclk:ptp=IEEE1588-2008:00-0B-2F-FF-FE-01-38-83:0 | Master PTP clock mac address. It not mandatory to add this line, it could work without. |
a=recvonly | - |
a=maxptime 1.0 | max packet time |
Troubleshooting
If ISP streams do not appear in the Dante transmitters list
Make sure the devices are correctly connected to the same switch.
Make sure the ISP device stream multicast address prefix is the same as in the Dante controller.
No audio
Make sure the switch is Gigabit.
Make sure devices are operating using the same sampling rate. ISP sample rate is available in the WebUI settings tab.
Make sure you have chosen the AES67 Flow when you created the Dante Multicast Stream.
Latency performance issue: Go in Dante Controller > Device view > Latency tab. The latency value displayed should always be green. Otherwise, go in Dante Device Config tab and increase the latency.
PTP Clocks: make sure you only have 1 PTP Master and the other devices are slaves and synced on it. See Dante Controller> Clock Status and ANEMAN PTP tab.
Audio and connections not stable
You can add more delay in the “Session sinks” tab with your connections.
It’s also possible in some situations to use “unicast” connections in ANEMAN between devices when you don’t need multicast. This would avoid potential conflict between devices, allowing better stability.
If you can’t find the mac address of PTP master device you can remove this line (a=ts-refclk:ptp=IEEE1588-2008:00-0B-2F-FF-FE-01-38-83:0) from the sdp. This line is most of the time not mandatory.
=IEEE1588-2008:00-0B-2F-FF-FE-01-38-83:0) from the sdp. This line is most of the time not mandatory.
Dante Source not recognized
If your purchased your AoIP option before 1st October 2022, in some cases the firmware of the option needs to be updated. We will update your AoIP ourselves during a remote session with you. You cannot do the update yourself. It should take about 30 minutes and nothing is needed from you aside from setting a connection through TeamViewer or AnyDesk on a computer connected to the same network as your ISP and AoIP option. Please submit a ticket for this in our Service Desk and we’ll get on with you ASAP: https://support-stormaudio.atlassian.net/servicedesk/customer/portal/5/group/26/create/103
If you’re using a Cisco 350 switch, you also need to check this section to configure it correctly: AoIP option - Output configuration | Cisco SG350 Users
Related pages
StormAudio - Knowledge Center