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path: root/gst/rtsp-server/rtsp-client.c
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Diffstat (limited to 'gst/rtsp-server/rtsp-client.c')
-rw-r--r--gst/rtsp-server/rtsp-client.c114
1 files changed, 82 insertions, 32 deletions
diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 45d178a..83131f3 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -1518,6 +1518,7 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
gint matched;
gchar *seek_style = NULL;
GstRTSPStatusCode sig_result;
+ GPtrArray *transports;
if (!(session = ctx->session))
goto no_session;
@@ -1556,6 +1557,14 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
+ /* update the pipeline */
+ transports = gst_rtsp_session_media_get_transports (sessmedia);
+ if (!gst_rtsp_media_complete_pipeline (media, transports)) {
+ g_ptr_array_unref (transports);
+ goto pipeline_error;
+ }
+ g_ptr_array_unref (transports);
+
/* in play we first unsuspend, media could be suspended from SDP or PAUSED */
if (!gst_rtsp_media_unsuspend (media))
goto unsuspend_failed;
@@ -1666,6 +1675,13 @@ invalid_state:
ctx);
return FALSE;
}
+pipeline_error:
+ {
+ GST_ERROR ("client %p: failed to configure the pipeline", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ return FALSE;
+ }
unsuspend_failed:
{
GST_ERROR ("client %p: unsuspend failed", client);
@@ -1784,39 +1800,52 @@ default_configure_client_transport (GstRTSPClient * client,
GstRTSPClientPrivate *priv = client->priv;
/* we have a valid transport now, set the destination of the client. */
- if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
- gboolean use_client_settings;
-
- use_client_settings =
- gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
-
- if (ct->destination && use_client_settings) {
- GstRTSPAddress *addr;
-
- addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
- ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
-
- if (addr == NULL)
- goto no_address;
-
- gst_rtsp_address_free (addr);
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST ||
+ ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP) {
+
+ /* allocate UDP ports */
+ GSocketFamily family;
+ gboolean use_client_settings = FALSE;
+
+ family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
+ if ((ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) &&
+ gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS) &&
+ (ct->destination != NULL))
+ use_client_settings = TRUE;
+
+ if (!gst_rtsp_stream_allocate_udp_sockets (ctx->stream, family, ct,
+ use_client_settings))
+ goto error_allocating_ports;
+
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ GstRTSPAddress *addr = NULL;
+
+ if (use_client_settings) {
+ /* the address has been successfully allocated, let's check if it's
+ * the one requested by the client */
+ addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
+ ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
+
+ if (addr == NULL)
+ goto no_address;
+ } else {
+ g_free (ct->destination);
+ addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
+ if (addr == NULL)
+ goto no_address;
+ ct->destination = g_strdup (addr->address);
+ ct->port.min = addr->port;
+ ct->port.max = addr->port + addr->n_ports - 1;
+ ct->ttl = addr->ttl;
+
+ gst_rtsp_address_free (addr);
+ }
} else {
- GstRTSPAddress *addr;
- GSocketFamily family;
-
- family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
-
- addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
- if (addr == NULL)
- goto no_address;
+ GstRTSPUrl *url;
+ url = gst_rtsp_connection_get_url (priv->connection);
g_free (ct->destination);
- ct->destination = g_strdup (addr->address);
- ct->port.min = addr->port;
- ct->port.max = addr->port + addr->n_ports - 1;
- ct->ttl = addr->ttl;
-
- gst_rtsp_address_free (addr);
+ ct->destination = g_strdup (url->host);
}
} else {
GstRTSPUrl *url;
@@ -1863,9 +1892,14 @@ default_configure_client_transport (GstRTSPClient * client,
return TRUE;
/* ERRORS */
+error_allocating_ports:
+ {
+ GST_ERROR_OBJECT (client, "Failed to allocate UDP ports");
+ return FALSE;
+ }
no_address:
{
- GST_ERROR_OBJECT (client, "failed to acquire address for stream");
+ GST_ERROR_OBJECT (client, "Failed to acquire address for stream");
return FALSE;
}
}
@@ -3036,6 +3070,7 @@ handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
gchar *path;
gint matched;
GstRTSPStatusCode sig_result;
+ GPtrArray *transports;
if (!(session = ctx->session))
goto no_session;
@@ -3074,7 +3109,15 @@ handle_record_request (GstRTSPClient * client, GstRTSPContext * ctx)
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
- /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ /* update the pipeline */
+ transports = gst_rtsp_session_media_get_transports (sessmedia);
+ if (!gst_rtsp_media_complete_pipeline (media, transports)) {
+ g_ptr_array_unref (transports);
+ goto pipeline_error;
+ }
+ g_ptr_array_unref (transports);
+
+ /* in record we first unsuspend, media could be suspended from SDP or PAUSED */
if (!gst_rtsp_media_unsuspend (media))
goto unsuspend_failed;
@@ -3140,6 +3183,13 @@ invalid_state:
ctx);
return FALSE;
}
+pipeline_error:
+ {
+ GST_ERROR ("client %p: failed to configure the pipeline", client);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ ctx);
+ return FALSE;
+ }
unsuspend_failed:
{
GST_ERROR ("client %p: unsuspend failed", client);
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-23 10:16:46 +0000