Dante/AES67-RTP-Stream to SBC: Latency

CVX DEV cvxdev at gmail.com
Tue Feb 18 13:14:25 UTC 2020

Hi all,

I am currently playing around with an AudioOverIP Project and wondered if
you could help me out. I have a LAN, with an Audio Source
(Dante/AES67-RTP-Stream) which I would like to distribute to multiple
receivers (SBC (e.g. RaspberryPi) with an Audio Output (e.g. Headphone

Source (e.g. PC)-->Dante-Audio-USB-Dongle-->AES67/RTP-Multicast-Stream-->LAN-Network-Switch-->RPI
(Gstreamer --> AudioJack)

I currently use the following Gstreamer Pipeline command on the RPi:

gst-launch-1.0 -v udpsrc uri=udp://
! rtpL24depay ! audioconvert ! alsasink device=hw:0,0

It all works fine, but if I watch a video on the PC and listen to the Audio
on the RPI, I have some latency (~200-300ms), therefore my questions:

   1. Do I miss something in my Gstreamer Pipeline to be able to reduce
   2. What is the minimal Latency to be expected with RTP-Streams, is
   *<50ms* achievable?
   3. Would the latency occur due to the network or due to the speed of the
   4. Since my audio-input is not a Gstreamer input, I assume
   rtpjitterbuffer or similar would not help to decrease latency / improve

Here the SDP-extract from the Audio-Source:
o=- 1484410 1484415 IN IP4
s=avio : 2
c=IN IP4
t=0 0
m=audio 5004 RTP/AVP 97
i=2 channels: Left, Right
a=rtpmap:97 L24/48000/2

Thanks in advance for your feedback & help.
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