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gst/rtsp/gstrtspsrc.*: Add property to configure NAT traversal method.

Browse source 
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_udp_sinks),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_send_dummy_packets),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add property to configure NAT traversal method.
Ignore EOS from the internal sinks.
Implement sending dummy packets as a (simple) method to open up
some firewalls.
Send PLAY request to the server after we started the udp sources.
Fixes #559545.
This commit is contained in:
Eric Zhang 2008-11-13 16:11:16 +00:00 committed by Wim Taymans
parent bfb73112af
commit be3906c918
3 changed files with 234 additions and 76 deletions
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21
ChangeLog
View file

@ -1,3 +1,24 @@
2008-11-13 Wim Taymans <wim.taymans@collabora.co.uk>
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_udp_sinks),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_send_dummy_packets),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add property to configure NAT traversal method.
Ignore EOS from the internal sinks.
Implement sending dummy packets as a (simple) method to open up
some firewalls.
Send PLAY request to the server after we started the udp sources.
Fixes #559545.
2008-11-13 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: Yotam <sh dot yotam at gmail dot com>

268
gst/rtsp/gstrtspsrc.c
View file

@ -152,6 +152,7 @@ enum
#define DEFAULT_TCP_TIMEOUT 20000000
#define DEFAULT_LATENCY_MS 3000
#define DEFAULT_CONNECTION_SPEED 0
#define DEFAULT_NAT_METHOD GST_RTSP_NAT_DUMMY
enum
{
@ -163,7 +164,9 @@ enum
PROP_TIMEOUT,
PROP_TCP_TIMEOUT,
PROP_LATENCY,
PROP_CONNECTION_SPEED
PROP_CONNECTION_SPEED,
PROP_NAT_METHOD,
PROP_LAST
};
#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
@ -185,6 +188,24 @@ gst_rtsp_lower_trans_get_type (void)
return rtsp_lower_trans_type;
}
#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
static GType
gst_rtsp_nat_method_get_type (void)
{
static GType rtsp_nat_method_type = 0;
static const GEnumValue rtsp_nat_method[] = {
{GST_RTSP_NAT_NONE, "None", "none"},
{GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
{0, NULL, NULL},
};
if (!rtsp_nat_method_type) {
rtsp_nat_method_type =
g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
}
return rtsp_nat_method_type;
}
static void gst_rtspsrc_base_init (gpointer g_class);
static void gst_rtspsrc_finalize (GObject * object);
@ -315,6 +336,12 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
0, G_MAXINT / 1000, DEFAULT_CONNECTION_SPEED,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
g_param_spec_enum ("nat-method", "NAT Method",
"Method to use for traversing firewalls and NAT",
GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
gstelement_class->change_state = gst_rtspsrc_change_state;
gstbin_class->handle_message = gst_rtspsrc_handle_message;
@ -431,6 +458,9 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
case PROP_CONNECTION_SPEED:
rtspsrc->connection_speed = g_value_get_uint (value);
break;
case PROP_NAT_METHOD:
rtspsrc->nat_method = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -476,6 +506,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
case PROP_CONNECTION_SPEED:
g_value_set_uint (value, rtspsrc->connection_speed);
break;
case PROP_NAT_METHOD:
g_value_set_enum (value, rtspsrc->nat_method);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -656,7 +689,6 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
* the RTP-Info header field returned from PLAY. */
control_url = gst_sdp_media_get_attribute_val (media, "control");
GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
@ -703,19 +735,17 @@ gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
g_free (stream->setup_url);
for (i = 0; i < 2; i++) {
GstElement *udpsrc = stream->udpsrc[i];
if (udpsrc) {
if (stream->udpsrc[i]) {
GstPad *pad;
/* unlink the pad */
pad = gst_element_get_static_pad (udpsrc, "src");
pad = gst_element_get_static_pad (stream->udpsrc[i], "src");
if (stream->channelpad[i]) {
gst_pad_unlink (pad, stream->channelpad[i]);
}
gst_element_set_state (udpsrc, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), udpsrc);
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
}
@ -723,12 +753,18 @@ gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
gst_object_unref (stream->channelpad[i]);
stream->channelpad[i] = NULL;
}
if (stream->udpsink[i]) {
gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
gst_object_unref (stream->udpsink[i]);
stream->udpsink[i] = NULL;
}
}
if (stream->udpsink) {
gst_element_set_state (stream->udpsink, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->udpsink);
gst_object_unref (stream->udpsink);
stream->udpsink = NULL;
if (stream->fakesrc) {
gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
gst_object_unref (stream->fakesrc);
stream->fakesrc = NULL;
}
if (stream->srcpad) {
gst_pad_set_active (stream->srcpad, FALSE);
@ -2052,28 +2088,22 @@ gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
/* configure the UDP sink back to the server for status reports */
static gboolean
gst_rtspsrc_stream_configure_udp_sink (GstRTSPSrc * src, GstRTSPStream * stream,
GstRTSPTransport * transport)
gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
GstRTSPStream * stream, GstRTSPTransport * transport)
{
GstPad *pad;
gint port, sockfd = -1;
gint rtp_port, rtcp_port, sockfd = -1;
const gchar *destination;
gchar *uri, *name;
/* no session, we're done */
if (src->session == NULL)
return TRUE;
/* get host and port */
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST)
port = transport->port.max;
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;
if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
rtp_port = transport->port.min;
rtcp_port = transport->port.max;
} else {
rtp_port = transport->server_port.min;
rtcp_port = transport->server_port.max;
}
/* first take the source, then the endpoint to figure out where to send
* the RTCP. */
@ -2081,51 +2111,108 @@ gst_rtspsrc_stream_configure_udp_sink (GstRTSPSrc * src, GstRTSPStream * stream,
if (destination == NULL)
destination = gst_rtsp_connection_get_ip (src->connection);
GST_DEBUG_OBJECT (src, "configure UDP sink for %s:%d", destination, port);
/* try to construct the fakesrc to the RTP port of the server to open up any
* NAT firewalls */
if (rtp_port != -1) {
GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
rtp_port);
uri = g_strdup_printf ("udp://%s:%d", destination, port);
stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
g_free (uri);
if (stream->udpsink == NULL)
goto no_sink_element;
uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
stream->udpsink[0] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
g_free (uri);
if (stream->udpsink[0] == NULL)
goto no_sink_element;
/* no sync needed */
g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
/* no async state changes needed */
g_object_set (G_OBJECT (stream->udpsink), "async", FALSE, NULL);
/* no sync or async state changes needed */
g_object_set (G_OBJECT (stream->udpsink[0]), "sync", FALSE, "async", FALSE,
NULL);
if (stream->udpsrc[1]) {
/* configure socket, we give it the same UDP socket as the udpsrc for RTCP
* because some servers check the port number of where it sends RTCP to identify
* the RTCP packets it receives */
g_object_get (G_OBJECT (stream->udpsrc[1]), "sock", &sockfd, NULL);
GST_DEBUG_OBJECT (src, "UDP src has sock %d", sockfd);
/* configure socket and make sure udpsink does not close it when shutting
* down, it belongs to udpsrc after all. */
g_object_set (G_OBJECT (stream->udpsink), "sockfd", sockfd, NULL);
g_object_set (G_OBJECT (stream->udpsink), "closefd", FALSE, NULL);
if (stream->udpsrc[0]) {
/* configure socket, we give it the same UDP socket as the udpsrc for RTP
* so that NAT firewalls will open a hole for us */
g_object_get (G_OBJECT (stream->udpsrc[0]), "sock", &sockfd, NULL);
GST_DEBUG_OBJECT (src, "RTP UDP src has sock %d", sockfd);
/* configure socket and make sure udpsink does not close it when shutting
* down, it belongs to udpsrc after all. */
g_object_set (G_OBJECT (stream->udpsink[0]), "sockfd", sockfd, NULL);
g_object_set (G_OBJECT (stream->udpsink[0]), "closefd", FALSE, NULL);
}
/* the source for the dummy packets to open up NAT */
stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
if (stream->fakesrc == NULL)
goto no_fakesrc_element;
/* random data in 5 buffers, a size of 200 bytes should be fine */
g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
NULL);
g_object_set (G_OBJECT (stream->fakesrc), "sizetype", 2, "sizemax", 200,
NULL);
/* we don't want to consider this a sink */
GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_IS_SINK);
/* keep everything locked */
gst_element_set_locked_state (stream->udpsink[0], TRUE);
gst_element_set_locked_state (stream->fakesrc, TRUE);
gst_object_ref (stream->udpsink[0]);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
gst_object_ref (stream->fakesrc);
gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
gst_element_link (stream->fakesrc, stream->udpsink[0]);
}
/* it's possible that the server does not want us to send RTCP in which case
* the port is -1 */
if (rtcp_port != -1 && src->session != NULL) {
GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
rtcp_port);
/* we don't want to consider this a sink */
GST_OBJECT_FLAG_UNSET (stream->udpsink, GST_ELEMENT_IS_SINK);
uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
stream->udpsink[1] = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
g_free (uri);
if (stream->udpsink[1] == NULL)
goto no_sink_element;
/* we keep this playing always */
gst_element_set_locked_state (stream->udpsink, TRUE);
gst_element_set_state (stream->udpsink, GST_STATE_PLAYING);
/* no sync needed */
g_object_set (G_OBJECT (stream->udpsink[1]), "sync", FALSE, NULL);
/* no async state changes needed */
g_object_set (G_OBJECT (stream->udpsink[1]), "async", FALSE, NULL);
gst_object_ref (stream->udpsink);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink);
if (stream->udpsrc[1]) {
/* configure socket, we give it the same UDP socket as the udpsrc for RTCP
* because some servers check the port number of where it sends RTCP to identify
* the RTCP packets it receives */
g_object_get (G_OBJECT (stream->udpsrc[1]), "sock", &sockfd, NULL);
GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %d", sockfd);
/* configure socket and make sure udpsink does not close it when shutting
* down, it belongs to udpsrc after all. */
g_object_set (G_OBJECT (stream->udpsink[1]), "sockfd", sockfd, NULL);
g_object_set (G_OBJECT (stream->udpsink[1]), "closefd", FALSE, NULL);
}
stream->rtcppad = gst_element_get_static_pad (stream->udpsink, "sink");
/* we don't want to consider this a sink */
GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_IS_SINK);
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
pad = gst_element_get_request_pad (src->session, name);
g_free (name);
/* we keep this playing always */
gst_element_set_locked_state (stream->udpsink[1], TRUE);
gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
/* and link */
if (pad)
gst_pad_link (pad, stream->rtcppad);
gst_object_ref (stream->udpsink[1]);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
pad = gst_element_get_request_pad (src->session, name);
g_free (name);
/* and link */
if (pad)
gst_pad_link (pad, stream->rtcppad);
}
return TRUE;
@ -2135,10 +2222,10 @@ no_sink_element:
GST_DEBUG_OBJECT (src, "no UDP sink element found");
return FALSE;
}
no_port:
no_fakesrc_element:
{
GST_DEBUG_OBJECT (src, "no valid port, ignoring RTCP for this stream");
return TRUE;
GST_DEBUG_OBJECT (src, "no fakesrc element found");
return FALSE;
}
}
@ -2192,8 +2279,9 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
case GST_RTSP_LOWER_TRANS_UDP:
if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
goto transport_failed;
/* configure udpsink back to the server for RTCP messages. */
if (!gst_rtspsrc_stream_configure_udp_sink (src, stream, transport))
/* configure udpsinks back to the server for RTCP messages and for the
* dummy RTP messages to open NAT. */
if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
goto transport_failed;
break;
default:
@ -2240,6 +2328,31 @@ no_manager:
}
}
/* send a couple of dummy random packets on the receiver RTP port to the server,
* this should make a firewall think we initiated the data transfer and
* hopefully allow packets to go from the sender port to our RTP receiver port */
static gboolean
gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
{
GList *walk;
if (!src->nat_method != GST_RTSP_NAT_DUMMY)
return TRUE;
for (walk = src->streams; walk; walk = g_list_next (walk)) {
GstRTSPStream *stream = (GstRTSPStream *) walk->data;
if (stream->fakesrc && stream->udpsink[0]) {
GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
}
}
return TRUE;
}
/* Adds the source pads of all configured streams to the element.
* This code is performed when we detected dataflow.
*
@ -4698,6 +4811,9 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
rtspsrc = GST_RTSPSRC (bin);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_EOS:
gst_message_unref (message);
break;
case GST_MESSAGE_ELEMENT:
{
const GstStructure *s = gst_message_get_structure (message);
@ -4788,9 +4904,9 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (rtspsrc, "PAUSED->PLAYING: stop connection flush");
gst_rtsp_connection_flush (rtspsrc->connection, FALSE);
/* FIXME, the server might send UDP packets before we activate the UDP
* ports */
gst_rtspsrc_play (rtspsrc, &rtspsrc->segment);
/* send some dummy packets before we chain up to the parent to activate
* the receive in the udp sources */
gst_rtspsrc_send_dummy_packets (rtspsrc);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
@ -4806,13 +4922,17 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
goto done;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
/* chained up to parent so the udp sources are activated and receiving */
gst_rtspsrc_play (rtspsrc, &rtspsrc->segment);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
gst_rtspsrc_pause (rtspsrc);
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtspsrc_close (rtspsrc);
break;

21
gst/rtsp/gstrtspsrc.h
View file

@ -106,10 +106,13 @@ struct _GstRTSPStream {
GstElement *udpsrc[2];
GstPad *blockedpad;
/* our udp sink back to the server */
GstElement *udpsink;
/* our udp sinks back to the server */
GstElement *udpsink[2];
GstPad *rtcppad;
/* fakesrc for sending dummy data */
GstElement *fakesrc;
/* state */
gint pt;
gboolean container;
@ -127,6 +130,19 @@ struct _GstRTSPStream {
guint rr_bandwidth;
};
/**
* GstRTSPNatMethod:
* @GST_RTSP_NAT_NONE: none
* @GST_RTSP_NAT_DUMMY: send dummy packets
*
* Different methods for trying to traverse firewalls.
*/
typedef enum
{
GST_RTSP_NAT_NONE,
GST_RTSP_NAT_DUMMY
} GstRTSPNatMethod;
struct _GstRTSPSrc {
GstBin parent;
@ -169,6 +185,7 @@ struct _GstRTSPSrc {
GTimeVal *ptcp_timeout;
guint latency;
guint connection_speed;
GstRTSPNatMethod nat_method;
/* state */
GstRTSPState state;