title: Audio Streaming: RTP-Stream receiving with Gstreamer - Latency
body: <p>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 jack):</p>
  
  <pre><code>PC--&gt;Audio-USB-Dongle--&gt;AES67/RTP-Multicast-Stream--&gt;LAN-Network-Switch--&gt;RPI (Gstreamer --&gt; AudioJack)
  </code></pre>
  
  <p>I currently use Gstreamer for the Pipeline:</p>
  
  <pre><code>gst-launch-1.0 -v udpsrc uri=udp://239.69.xxx.xx:5004 caps="application/x-rtp,channels=(int)2,format=(string)S16LE,media=(string)audio,payload=(int)96,clock-rate=(int)48000,encoding-name=(string)L24" ! rtpL24depay ! audioconvert ! alsasink device=hw:0,0
  </code></pre>
  
  <p>It all works fine, but if I watch a video on the PC and listen to the Audio from the RPI, I have some latency (~200-300ms), therefore my questions:</p>
  
  <ol>
  <li>Do I miss something in my Gstreamer Pipeline to be able to reduce latency?</li>
  <li>What is the minimal Latency to be expected with RTP-Streams, is &lt;50ms achievable?</li>
  <li>Would the latency occur due to the network or due to the speed of the RPi?</li>
  <li>Since my audio-input is not a Gstreamer input, I assume <code>rtpjitterbuffer</code> or similar would not help to decrease latency?</li>
  </ol>