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-rw-r--r--gst/rtsp/gstrtspsrc.c10
1 files changed, 10 insertions, 0 deletions
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.