pulsar256 · January 29, 2024 01:19 · Jun 8, 2020 · Jun 8, 2020 · Jun 8, 2020 · Jun 7, 2020
diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -23,7 +23,7 @@ gst-launch-1.0 \
 
 ### Gstreamer Based
 
-### Default / auto audio sink
+#### Default / auto audio sink
 
 ``` bash
 RECEIVER_PORT=5100
@@ -38,7 +38,7 @@ gst-launch-1.0 \
   ! autoaudiosink
 ```
 
-### Jack Audio Sink
+#### Jack Audio Sink
 
 ``` bash
 RECEIVER_PORT=5100

diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -23,6 +23,8 @@ gst-launch-1.0 \
 
 ### Gstreamer Based
 
+### Default / auto audio sink
+
 ``` bash
 RECEIVER_PORT=5100
 
@@ -36,7 +38,7 @@ gst-launch-1.0 \
   ! autoaudiosink
 ```
 
-### Gstreamer Based Jack Target
+### Jack Audio Sink
 
 ``` bash
 RECEIVER_PORT=5100
@@ -52,8 +54,6 @@ gst-launch-1.0 \
   ! jackaudiosink
 ```
 
-
-
 ### VLC Based
 
 VLC requires a SDP File specifying the RTP payload.

diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -29,6 +29,7 @@ RECEIVER_PORT=5100
 gst-launch-1.0 \
   udpsrc port=$RECEIVER_PORT \
   ! application/x-rtp,media=audio,payload=101,encoding-name=OPUS \
+  ! rtpjitterbuffer \
   ! rtpopusdepay \
   ! opusdec \
   ! audioconvert \
@@ -43,6 +44,7 @@ RECEIVER_PORT=5100
 gst-launch-1.0 \
   udpsrc port=$RECEIVER_PORT \
   ! application/x-rtp,media=audio,payload=101,encoding-name=OPUS \
+  ! rtpjitterbuffer \
   ! rtpopusdepay \
   ! opusdec \
   ! audioconvert \
@@ -77,3 +79,5 @@ See https://tools.ietf.org/html/rfc7587 for more details.
 ## Notes
 
 this solution provides a peer to peer path only. If multiple clients shall connect to the same audio source, a RTSP server would be the next logical building block. See https://github.com/GStreamer/gst-rtsp-server for examples how to implement one. `test-sdp` and `test-launch` in the `1.16` branch did not yeld any working servers during my tests.
+
+rtpjitterbuffer for the client might be optional if your network is stable and transport is udp (which it is in the examples above).
diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -35,6 +35,22 @@ gst-launch-1.0 \
   ! autoaudiosink
 ```
 
+### Gstreamer Based Jack Target
+
+``` bash
+RECEIVER_PORT=5100
+
+gst-launch-1.0 \
+  udpsrc port=$RECEIVER_PORT \
+  ! application/x-rtp,media=audio,payload=101,encoding-name=OPUS \
+  ! rtpopusdepay \
+  ! opusdec \
+  ! audioconvert \
+  ! audioresample \
+  ! jackaudiosink
+```
+
+
 
 ### VLC Based
 

diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -27,7 +27,7 @@ gst-launch-1.0 \
 RECEIVER_PORT=5100
 
 gst-launch-1.0 \
-  udpsrc port=$RECEIVER_PORT\
+  udpsrc port=$RECEIVER_PORT \
   ! application/x-rtp,media=audio,payload=101,encoding-name=OPUS \
   ! rtpopusdepay \
   ! opusdec \

diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -16,7 +16,7 @@ gst-launch-1.0 \
   ! audioconvert \
   ! opusenc bitrate-type=vbr bitrate=128000 \
   ! rtpopuspay pt=101 \
-  ! udpsink host=$TARGET_IP port=$TARGET_PORT blocksize=4096
+  ! udpsink host=$TARGET_IP port=$TARGET_PORT
 ```
 
 ## Consumer
@@ -27,7 +27,7 @@ gst-launch-1.0 \
 RECEIVER_PORT=5100
 
 gst-launch-1.0 \
-  udpsrc port=$RECEIVER_PORT blocksize=128 \
+  udpsrc port=$RECEIVER_PORT\
   ! application/x-rtp,media=audio,payload=101,encoding-name=OPUS \
   ! rtpopusdepay \
   ! opusdec \

diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -60,4 +60,4 @@ See https://tools.ietf.org/html/rfc7587 for more details.
 
 ## Notes
 
-this solution provides a peer to peer path only. If multiple clients shall connect to the same audio source, a RTSP server would be the next logical building block. See https://github.com/GStreamer/gst-rtsp-server for examples how to implement one. `test-odp` and `test-launch` in the `1.16` branch did not yeld any working servers during my tests.
+this solution provides a peer to peer path only. If multiple clients shall connect to the same audio source, a RTSP server would be the next logical building block. See https://github.com/GStreamer/gst-rtsp-server for examples how to implement one. `test-sdp` and `test-launch` in the `1.16` branch did not yeld any working servers during my tests.
diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -1,4 +1,4 @@
-# Abstract
+# Opus p2p streaming
 
 Low latency streaming of ogg compressed audio data between two peers using RTP and gstreamer-1.0. Expected latency over the network is below 1 second.
 
@@ -58,4 +58,6 @@ Please notice the hardcoded UDP port in the SDP file set to 5100, depending on y
 
 See https://tools.ietf.org/html/rfc7587 for more details. 
 
+## Notes
 
+this solution provides a peer to peer path only. If multiple clients shall connect to the same audio source, a RTSP server would be the next logical building block. See https://github.com/GStreamer/gst-rtsp-server for examples how to implement one. `test-odp` and `test-launch` in the `1.16` branch did not yeld any working servers during my tests.
diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -1,9 +1,11 @@
 # Abstract
 
-Low latency streaming of ogg compressed audio data between two peers using RTP and gstreamer-1.0
+Low latency streaming of ogg compressed audio data between two peers using RTP and gstreamer-1.0. Expected latency over the network is below 1 second.
 
 ## Producer
 
+The producer will provide a Jack sink which needs to be connected to a audio producer.
+
 ```bash
 TARGET_PORT=5100
 TARGET_IP=PEER_IP

diff --git a/opus-p2p-streaming.md b/opus-p2p-streaming.md
@@ -0,0 +1,59 @@
+# Abstract
+
+Low latency streaming of ogg compressed audio data between two peers using RTP and gstreamer-1.0
+
+## Producer
+
+```bash
+TARGET_PORT=5100
+TARGET_IP=PEER_IP
+
+gst-launch-1.0 \
+  jackaudiosrc connect=1 \
+  ! audio/x-raw, channels=2 \
+  ! audioconvert \
+  ! opusenc bitrate-type=vbr bitrate=128000 \
+  ! rtpopuspay pt=101 \
+  ! udpsink host=$TARGET_IP port=$TARGET_PORT blocksize=4096
+```
+
+## Consumer
+
+### Gstreamer Based
+
+``` bash
+RECEIVER_PORT=5100
+
+gst-launch-1.0 \
+  udpsrc port=$RECEIVER_PORT blocksize=128 \
+  ! application/x-rtp,media=audio,payload=101,encoding-name=OPUS \
+  ! rtpopusdepay \
+  ! opusdec \
+  ! audioconvert \
+  ! autoaudiosink
+```
+
+
+### VLC Based
+
+VLC requires a SDP File specifying the RTP payload.
+
+`stream.sdp`:
+
+```sdp
+m=audio 5100 RTP/AVP 101
+c=IN IP4 127.0.0.1
+a=rtpmap:101 opus/48000/2
+```
+
+run with
+
+```
+vlc ./stream.sdp
+```
+
+Please notice the hardcoded UDP port in the SDP file set to 5100, depending on your network configuration / forwarding rules this might need to change.
+
+See https://tools.ietf.org/html/rfc7587 for more details. 
+
+