diff options
| -rw-r--r-- | ChangeLog | 6 | ||||
| -rw-r--r-- | gst/rtsp/gstrtspsrc.c | 10 |
2 files changed, 16 insertions, 0 deletions
@@ -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. |