From eb4caff16a2bfea37d75d2ed2e86d3f5a0d39af0 Mon Sep 17 00:00:00 2001
From: Mathieu Duponchelle <mathieu@centricular.com>
Date: Thu, 29 May 2025 14:18:57 +0200
Subject: [PATCH] rtspclientsink: post error message on RTCP timeout

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/9124>
---
 .../gst/rtsp-sink/gstrtspclientsink.c         | 25 +++++++++++++++++--
 1 file changed, 23 insertions(+), 2 deletions(-)

diff --git a/subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c b/subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c
index abb9642786..8eccc59853 100644
--- a/subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c
+++ b/subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c
@@ -3988,6 +3988,24 @@ do_send_data_list (GstBufferList * buffer_list, guint8 channel,
   return res == GST_RTSP_OK;
 }
 
+static void
+transport_timed_out_notify_cb (GstRTSPStreamTransport * transport,
+    GParamSpec * unused, GstRTSPClientSink * sink)
+{
+  gboolean timed_out;
+
+  g_object_get (G_OBJECT (transport), "timed-out", &timed_out, NULL);
+
+  GST_DEBUG_OBJECT (sink, "Transport %p timed out notify: %d", transport,
+      timed_out);
+
+  if (timed_out) {
+    GST_ELEMENT_ERROR (sink, RESOURCE, WRITE, (NULL),
+        ("stream transport timed out"));
+  }
+}
+
+
 static GstRTSPResult
 gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async)
 {
@@ -4273,10 +4291,13 @@ gst_rtsp_client_sink_setup_streams (GstRTSPClientSink * sink, gboolean async)
       if (!retry) {
         GST_DEBUG ("Configuring the stream transport for stream %d",
             context->index);
-        if (context->stream_transport == NULL)
+        if (context->stream_transport == NULL) {
           context->stream_transport =
               gst_rtsp_stream_transport_new (stream, transport);
-        else
+
+          g_signal_connect (context->stream_transport, "notify::timed-out",
+              (GCallback) transport_timed_out_notify_cb, sink);
+        } else
           gst_rtsp_stream_transport_set_transport (context->stream_transport,
               transport);