<html><head><meta http-equiv="content-type" content="text/html; charset=us-ascii"></head><body style="overflow-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;">Hi Alan<div><br></div><div>GstRTSPServer supports HTTP tunneling by default, but the client is the one that needs to initiate the tunnel request. For example, using GStreamer as the client you can initiate RTSP over HTTP as:</div><div><br></div><div>gst-play-1.0 rtsp<b>h</b>://$IP_ADDRESS:$PORT/$MAPPING</div><div><br></div><div>(Note the <b>h</b> in rtsp<b>h</b>)</div><div><br></div><div>In VLC, for example, you would go to <i>Settings -> Show All -> Input / Codecs -> Demuxers -> RTP / RTSP -> Tunnel RTP and RTSP over HTTP</i></div><div><br></div><div>Michael</div><div><br><div><br><blockquote type="cite"><div>On 7 Aug 2023, at 11:49, Alan Petrus via gstreamer-devel <gstreamer-devel@lists.freedesktop.org> wrote:</div><br class="Apple-interchange-newline"><div><div dir="ltr"><div>Greetings, <br></div><div><br></div><div>Does gst-rtsp-server actually support streaming RTSP-over-HTTP? Is it supposed to be used in combination with a web server like Apache/Nginx? Is there any documentation or sample/pseudo code on how to go about it?<br></div><div><br></div><div>I found some encouraging comments here that's supported: <a href="https://coaxion.net/blog/2013/10/streaming-gstreamer-pipelines-via-http/#comment-74952">https://coaxion.net/blog/2013/10/streaming-gstreamer-pipelines-via-http/#comment-74952</a>. However, when I searched the code base I only found gst_rtsp_server_transfer_connection() which doesn't seem to be called from anywhere in the code base :( <br></div><div><br></div><div>Thanks a bunch.<br></div><div><br></div></div> </div></blockquote></div><br></div></body></html>