gst/rtsp/gstrtspsrc.c: Don't try to configure RTCP back to the server when the server did not give us a valid port nu...

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
Don't try to configure RTCP back to the server when the server did not
give us a valid port number.

2 files changed, 16 insertions, 0 deletions

diff --git a/ChangeLog b/ChangeLog
index 587c0fb0..7b1cc150 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,11 @@
 2008-08-20  Wim Taymans  <wim.taymans@collabora.co.uk>
 
+	* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
+	Don't try to configure RTCP back to the server when the server did not
+	give us a valid port number.
+
+2008-08-20  Wim Taymans  <wim.taymans@collabora.co.uk>
+
 	* gst/videobox/gstvideobox.c: (gst_video_box_set_property):
 	Use new basetransform method to renegotiate. Fixes #544956.
 
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 46850833..048786b5 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -2057,6 +2057,11 @@ gst_rtspsrc_stream_configure_udp_sink (GstRTSPSrc * src, GstRTSPStream * stream,
   else
     port = transport->server_port.max;
 
+  /* it's possible that the server does not want us to send RTCP in which case
+   * the port is -1 */
+  if (port == -1)
+    goto no_port;
+
   /* first take the source, then the endpoint to figure out where to send
    * the RTCP. */
   destination = transport->source;
@@ -2114,6 +2119,11 @@ no_sink_element:
     GST_DEBUG_OBJECT (src, "no UDP sink element found");
     return FALSE;
   }
+no_port:
+  {
+    GST_DEBUG_OBJECT (src, "no valid port, ignoring RTCP for this stream");
+    return TRUE;
+  }
 }
 
 /* sets up all elements needed for streaming over the specified transport.