* Boston, MA 02111-1307, USA.
*/
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <errno.h>
+#include <string.h>
+#include <sys/time.h>
+#include <sys/types.h>
+#include <netinet/in.h>
+#include <netdb.h>
+#include <sys/socket.h>
+#include <sys/wait.h>
+#include <fcntl.h>
+#include <arpa/inet.h>
#include <sys/ioctl.h>
#include "rtsp-client.h"
#include "rtsp-sdp.h"
+#include "rtsp-params.h"
-#define DEBUG
-
-static GMutex *tunnels_lock;
+static GMutex tunnels_lock;
static GHashTable *tunnels;
enum
PROP_LAST
};
-static void gst_rtsp_client_get_property (GObject *object, guint propid,
- GValue *value, GParamSpec *pspec);
-static void gst_rtsp_client_set_property (GObject *object, guint propid,
- const GValue *value, GParamSpec *pspec);
+enum
+{
+ SIGNAL_CLOSED,
+ SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
+#define GST_CAT_DEFAULT rtsp_client_debug
+
+static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec);
static void gst_rtsp_client_finalize (GObject * obj);
-static void client_session_finalized (GstRTSPClient *client, GstRTSPSession *session);
+static void client_session_finalized (GstRTSPClient * client,
+ GstRTSPSession * session);
+static void unlink_session_streams (GstRTSPClient * client,
+ GstRTSPSession * session, GstRTSPSessionMedia * media);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
g_param_spec_object ("session-pool", "Session Pool",
"The session pool to use for client session",
- GST_TYPE_RTSP_SESSION_POOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ GST_TYPE_RTSP_SESSION_POOL,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
g_param_spec_object ("media-mapping", "Media Mapping",
"The media mapping to use for client session",
- GST_TYPE_RTSP_MEDIA_MAPPING, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ GST_TYPE_RTSP_MEDIA_MAPPING,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ gst_rtsp_client_signals[SIGNAL_CLOSED] =
+ g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+ G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
+ g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+ tunnels =
+ g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
+ g_mutex_init (&tunnels_lock);
- tunnels = g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
- tunnels_lock = g_mutex_new ();
+ GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
}
static void
{
}
+static void
+client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+ /* unlink all media managed in this session */
+ while (g_list_length (session->medias) > 0) {
+ GstRTSPSessionMedia *media = g_list_first (session->medias)->data;
+
+ gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
+ unlink_session_streams (client, session, media);
+ /* unmanage the media in the session. this will modify session->medias */
+ gst_rtsp_session_release_media (session, media);
+ }
+}
+
+static void
+client_cleanup_sessions (GstRTSPClient * client)
+{
+ GList *sessions;
+
+ /* remove weak-ref from sessions */
+ for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
+ GstRTSPSession *session = (GstRTSPSession *) sessions->data;
+ g_object_weak_unref (G_OBJECT (session),
+ (GWeakNotify) client_session_finalized, client);
+ client_unlink_session (client, session);
+ }
+ g_list_free (client->sessions);
+ client->sessions = NULL;
+}
+
/* A client is finalized when the connection is broken */
static void
gst_rtsp_client_finalize (GObject * obj)
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
- g_message ("finalize client %p", client);
+ GST_INFO ("finalize client %p", client);
- g_list_free (client->sessions);
+ client_cleanup_sessions (client);
gst_rtsp_connection_free (client->connection);
if (client->session_pool)
g_object_unref (client->session_pool);
if (client->media_mapping)
g_object_unref (client->media_mapping);
+ if (client->auth)
+ g_object_unref (client->auth);
if (client->uri)
gst_rtsp_url_free (client->uri);
if (client->media)
g_object_unref (client->media);
+ g_free (client->server_ip);
+
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
static void
-gst_rtsp_client_get_property (GObject *object, guint propid,
- GValue *value, GParamSpec *pspec)
+gst_rtsp_client_get_property (GObject * object, guint propid,
+ GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
}
static void
-gst_rtsp_client_set_property (GObject *object, guint propid,
- const GValue *value, GParamSpec *pspec)
+gst_rtsp_client_set_property (GObject * object, guint propid,
+ const GValue * value, GParamSpec * pspec)
{
GstRTSPClient *client = GST_RTSP_CLIENT (object);
* gst_rtsp_client_new:
*
* Create a new #GstRTSPClient instance.
+ *
+ * Returns: a new #GstRTSPClient
*/
GstRTSPClient *
gst_rtsp_client_new (void)
}
static void
-send_response (GstRTSPClient *client, GstRTSPSession *session, GstRTSPMessage *response)
+send_response (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPMessage * response)
{
- gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER, "GStreamer RTSP server");
+ gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
+ "GStreamer RTSP server");
/* remove any previous header */
gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
gchar *str;
if (session->timeout != 60)
- str = g_strdup_printf ("%s; timeout=%d", session->sessionid, session->timeout);
+ str =
+ g_strdup_printf ("%s; timeout=%d", session->sessionid,
+ session->timeout);
else
str = g_strdup (session->sessionid);
gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, str);
}
-#ifdef DEBUG
- gst_rtsp_message_dump (response);
-#endif
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (response);
+ }
- gst_rtsp_watch_queue_message (client->watch, response);
+ gst_rtsp_watch_send_message (client->watch, response, NULL);
gst_rtsp_message_unset (response);
}
static void
-send_generic_response (GstRTSPClient *client, GstRTSPStatusCode code,
- GstRTSPMessage *request)
+send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
+ GstRTSPClientState * state)
{
- GstRTSPMessage response = { 0 };
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
- gst_rtsp_message_init_response (&response, code,
- gst_rtsp_status_as_text (code), request);
+ send_response (client, NULL, state->response);
+}
- send_response (client, NULL, &response);
+static void
+handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
+ GstRTSPClientState * state)
+{
+ gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
+ gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
+
+ if (auth) {
+ /* and let the authentication manager setup the auth tokens */
+ gst_rtsp_auth_setup_auth (auth, client, 0, state);
+ }
+
+ send_response (client, state->session, state->response);
}
+
static gboolean
-compare_uri (const GstRTSPUrl *uri1, const GstRTSPUrl *uri2)
+compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
{
if (uri1 == NULL || uri2 == NULL)
return FALSE;
* but is cached for when the same client (without breaking the connection) is
* doing a setup for the exact same url. */
static GstRTSPMedia *
-find_media (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
+find_media (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
+ GstRTSPAuth *auth;
- if (!compare_uri (client->uri, uri)) {
+ if (!compare_uri (client->uri, state->uri)) {
/* remove any previously cached values before we try to construct a new
* media for uri */
if (client->uri)
goto no_mapping;
/* find the factory for the uri first */
- if (!(factory = gst_rtsp_media_mapping_find_factory (client->media_mapping, uri)))
+ if (!(factory =
+ gst_rtsp_media_mapping_find_factory (client->media_mapping,
+ state->uri)))
goto no_factory;
+ state->factory = factory;
+
+ /* check if we have access to the factory */
+ if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
+ if (!gst_rtsp_auth_check (auth, client, 0, state))
+ goto not_allowed;
+
+ g_object_unref (auth);
+ }
+
/* prepare the media and add it to the pipeline */
- if (!(media = gst_rtsp_media_factory_construct (factory, uri)))
+ if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
goto no_media;
+ g_object_unref (factory);
+ factory = NULL;
+ state->factory = NULL;
+
+ /* set ipv6 on the media before preparing */
+ media->is_ipv6 = client->is_ipv6;
+ state->media = media;
+
/* prepare the media */
if (!(gst_rtsp_media_prepare (media)))
goto no_prepare;
/* now keep track of the uri and the media */
- client->uri = gst_rtsp_url_copy (uri);
+ client->uri = gst_rtsp_url_copy (state->uri);
client->media = media;
- }
- else {
+ } else {
/* we have seen this uri before, used cached media */
media = client->media;
- g_message ("reusing cached media %p", media);
+ state->media = media;
+ GST_INFO ("reusing cached media %p", media);
}
if (media)
/* ERRORS */
no_mapping:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return NULL;
}
no_factory:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ return NULL;
+ }
+not_allowed:
+ {
+ handle_unauthorized_request (client, auth, state);
+ g_object_unref (factory);
+ g_object_unref (auth);
return NULL;
}
no_media:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (factory);
return NULL;
}
no_prepare:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
- g_object_unref (factory);
return NULL;
}
}
static gboolean
-do_send_data (GstBuffer *buffer, guint8 channel, GstRTSPClient *client)
+do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
GstRTSPMessage message = { 0 };
+ GstMapInfo map_info;
guint8 *data;
- guint size;
+ guint usize;
gst_rtsp_message_init_data (&message, channel);
- data = GST_BUFFER_DATA (buffer);
- size = GST_BUFFER_SIZE (buffer);
- gst_rtsp_message_take_body (&message, data, size);
+ if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
+ return FALSE;
+
+ gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
+
+ /* FIXME, client->watch could have been finalized here, we need to keep an
+ * extra refcount to the watch. */
+ gst_rtsp_watch_send_message (client->watch, &message, NULL);
- gst_rtsp_watch_queue_message (client->watch, &message);
+ gst_rtsp_message_steal_body (&message, &data, &usize);
+ gst_buffer_unmap (buffer, &map_info);
- gst_rtsp_message_steal_body (&message, &data, &size);
gst_rtsp_message_unset (&message);
return TRUE;
}
static void
-link_stream (GstRTSPClient *client, GstRTSPSessionStream *stream)
+link_stream (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPSessionStream * stream)
{
+ GST_DEBUG ("client %p: linking stream %p", client, stream);
gst_rtsp_session_stream_set_callbacks (stream, (GstRTSPSendFunc) do_send_data,
- (GstRTSPSendFunc) do_send_data, client, NULL);
+ (GstRTSPSendFunc) do_send_data, client, NULL);
client->streams = g_list_prepend (client->streams, stream);
+ /* make sure our session can't expire */
+ gst_rtsp_session_prevent_expire (session);
}
static void
-unlink_stream (GstRTSPClient *client, GstRTSPSessionStream *stream)
+unlink_stream (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPSessionStream * stream)
{
- gst_rtsp_session_stream_set_callbacks (stream, NULL,
- NULL, NULL, NULL);
+ GST_DEBUG ("client %p: unlinking stream %p", client, stream);
+ gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL);
client->streams = g_list_remove (client->streams, stream);
+ /* our session can now expire */
+ gst_rtsp_session_allow_expire (session);
}
static void
-unlink_streams (GstRTSPClient *client)
-{
- GList *walk;
-
- for (walk = client->streams; walk; walk = g_list_next (walk)) {
- GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
-
- gst_rtsp_session_stream_set_callbacks (stream, NULL,
- NULL, NULL, NULL);
- }
- g_list_free (client->streams);
- client->streams = NULL;
-}
-
-static void
-unlink_session_streams (GstRTSPClient *client, GstRTSPSessionMedia *media)
+unlink_session_streams (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPSessionMedia * media)
{
guint n_streams, i;
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, unlink the stream from the TCP connection of the client */
- unlink_stream (client, sstream);
+ unlink_stream (client, session, sstream);
}
}
}
+static void
+close_connection (GstRTSPClient * client)
+{
+ const gchar *tunnelid;
+
+ GST_DEBUG ("client %p: closing connection", client);
+
+ if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
+ g_mutex_lock (&tunnels_lock);
+ /* remove from tunnelids */
+ g_hash_table_remove (tunnels, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+ }
+
+ gst_rtsp_connection_close (client->connection);
+ if (client->watchid) {
+ g_source_destroy ((GSource *) client->watch);
+ client->watchid = 0;
+ client->watch = NULL;
+ }
+}
+
static gboolean
-handle_teardown_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
+handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPSession *session;
GstRTSPSessionMedia *media;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
- if (!session)
+ if (!state->session)
goto no_session;
+ session = state->session;
+
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, uri);
+ media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
+ state->sessmedia = media;
+
/* unlink the all TCP callbacks */
- unlink_session_streams (client, media);
+ unlink_session_streams (client, session, media);
/* remove the session from the watched sessions */
- g_object_weak_unref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client);
+ g_object_weak_unref (G_OBJECT (session),
+ (GWeakNotify) client_session_finalized, client);
client->sessions = g_list_remove (client->sessions, session);
gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
}
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
+
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
+ "close");
+
+ send_response (client, session, state->response);
- send_response (client, session, &response);
+ close_connection (client);
- return FALSE;
+ return TRUE;
/* ERRORS */
no_session:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+{
+ GstRTSPResult res;
+ guint8 *data;
+ guint size;
+
+ res = gst_rtsp_message_get_body (state->request, &data, &size);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ if (size == 0) {
+ /* no body, keep-alive request */
+ send_generic_response (client, GST_RTSP_STS_OK, state);
+ } else {
+ /* there is a body, handle the params */
+ res = gst_rtsp_params_get (client, state);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ send_response (client, state->session, state->response);
+ }
+ return TRUE;
+
+ /* ERRORS */
+bad_request:
+ {
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
+ return FALSE;
+ }
+}
+
+static gboolean
+handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
+{
+ GstRTSPResult res;
+ guint8 *data;
+ guint size;
+
+ res = gst_rtsp_message_get_body (state->request, &data, &size);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ if (size == 0) {
+ /* no body, keep-alive request */
+ send_generic_response (client, GST_RTSP_STS_OK, state);
+ } else {
+ /* there is a body, handle the params */
+ res = gst_rtsp_params_set (client, state);
+ if (res != GST_RTSP_OK)
+ goto bad_request;
+
+ send_response (client, state->session, state->response);
+ }
+ return TRUE;
+
+ /* ERRORS */
+bad_request:
+ {
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
}
static gboolean
-handle_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
+handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPSession *session;
GstRTSPSessionMedia *media;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
- if (!session)
+ if (!(session = state->session))
goto no_session;
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, uri);
+ media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
+ state->sessmedia = media;
+
/* the session state must be playing or recording */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != GST_RTSP_STATE_RECORDING)
goto invalid_state;
/* unlink the all TCP callbacks */
- unlink_session_streams (client, media);
+ unlink_session_streams (client, session, media);
/* then pause sending */
gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
- send_response (client, session, &response);
+ send_response (client, session, state->response);
/* the state is now READY */
media->state = GST_RTSP_STATE_READY;
- return FALSE;
+ return TRUE;
/* ERRORS */
no_session:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
- send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ state);
return FALSE;
}
}
static gboolean
-handle_play_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
+handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
{
+ GstRTSPSession *session;
GstRTSPSessionMedia *media;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
GString *rtpinfo;
guint n_streams, i, infocount;
GstRTSPTimeRange *range;
GstRTSPResult res;
- if (!session)
+ if (!(session = state->session))
goto no_session;
/* get a handle to the configuration of the media in the session */
- media = gst_rtsp_session_get_media (session, uri);
+ media = gst_rtsp_session_get_media (session, state->uri);
if (!media)
goto not_found;
+ state->sessmedia = media;
+
/* the session state must be playing or ready */
if (media->state != GST_RTSP_STATE_PLAYING &&
media->state != GST_RTSP_STATE_READY)
goto invalid_state;
/* parse the range header if we have one */
- res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_RANGE, &str, 0);
+ res =
+ gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
if (res == GST_RTSP_OK) {
if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
/* we have a range, seek to the position */
/* get the stream as configured in the session */
sstream = gst_rtsp_session_media_get_stream (media, i);
/* get the transport, if there is no transport configured, skip this stream */
- if (!(tr = sstream->trans.transport))
+ if (!(tr = sstream->trans.transport)) {
+ GST_INFO ("stream %d is not configured", i);
continue;
+ }
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* for TCP, link the stream to the TCP connection of the client */
- link_stream (client, sstream);
+ link_stream (client, session, sstream);
}
stream = sstream->media_stream;
payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
if (g_object_class_find_property (payobjclass, "seqnum") &&
- g_object_class_find_property (payobjclass, "timestamp")) {
+ g_object_class_find_property (payobjclass, "timestamp")) {
GObject *payobj;
payobj = G_OBJECT (stream->payloader);
if (infocount > 0)
g_string_append (rtpinfo, ", ");
- uristr = gst_rtsp_url_get_request_uri (uri);
- g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u", uristr, i, seqnum, timestamp);
+ uristr = gst_rtsp_url_get_request_uri (state->uri);
+ g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
+ uristr, i, seqnum, timestamp);
g_free (uristr);
infocount++;
+ } else {
+ GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
}
}
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
/* add the RTP-Info header */
- str = g_string_free (rtpinfo, FALSE);
- gst_rtsp_message_take_header (&response, GST_RTSP_HDR_RTP_INFO, str);
+ if (infocount > 0) {
+ str = g_string_free (rtpinfo, FALSE);
+ gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
+ } else {
+ g_string_free (rtpinfo, TRUE);
+ }
/* add the range */
- str = gst_rtsp_range_to_string (&media->media->range);
- gst_rtsp_message_take_header (&response, GST_RTSP_HDR_RANGE, str);
+ str = gst_rtsp_media_get_range_string (media->media, TRUE);
+ gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
- send_response (client, session, &response);
+ send_response (client, session, state->response);
/* start playing after sending the request */
gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
media->state = GST_RTSP_STATE_PLAYING;
- return FALSE;
+ return TRUE;
/* ERRORS */
no_session:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
return FALSE;
}
invalid_state:
{
- send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE, request);
+ send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+ state);
return FALSE;
}
}
+static void
+do_keepalive (GstRTSPSession * session)
+{
+ GST_INFO ("keep session %p alive", session);
+ gst_rtsp_session_touch (session);
+}
+
static gboolean
-handle_setup_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
+handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
+ GstRTSPUrl *uri;
gchar *transport;
gchar **transports;
gboolean have_transport;
GstRTSPTransport *ct, *st;
gint i;
GstRTSPLowerTrans supported;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
+ GstRTSPSession *session;
GstRTSPSessionStream *stream;
gchar *trans_str, *pos;
guint streamid;
GstRTSPSessionMedia *media;
- gboolean need_session;
- GstRTSPUrl *url;
+
+ uri = state->uri;
/* the uri contains the stream number we added in the SDP config, which is
* always /stream=%d so we need to strip that off
* parse the stream we need to configure, look for the stream in the abspath
* first and then in the query. */
- if (!(pos = strstr (uri->abspath, "/stream="))) {
- if (!(pos = strstr (uri->query, "/stream=")))
+ if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
+ if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
goto bad_request;
}
goto bad_request;
/* parse the transport */
- res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_TRANSPORT, &transport, 0);
+ res =
+ gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
+ &transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
transports = g_strsplit (transport, ",", 0);
- gst_rtsp_transport_new (&ct);
+ gst_rtsp_transport_new (&ct);
- /* loop through the transports, try to parse */
+ /* init transports */
have_transport = FALSE;
- for (i = 0; transports[i]; i++) {
+ gst_rtsp_transport_init (ct);
+
+ /* our supported transports */
+ supported = GST_RTSP_LOWER_TRANS_UDP |
+ GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
- gst_rtsp_transport_init (ct);
+ /* loop through the transports, try to parse */
+ for (i = 0; transports[i]; i++) {
res = gst_rtsp_transport_parse (transports[i], ct);
- if (res == GST_RTSP_OK) {
- have_transport = TRUE;
- break;
+ if (res != GST_RTSP_OK) {
+ /* no valid transport, search some more */
+ GST_WARNING ("could not parse transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a transport, see if it's RTP/AVP */
+ if (ct->trans != GST_RTSP_TRANS_RTP || ct->profile != GST_RTSP_PROFILE_AVP) {
+ GST_WARNING ("invalid transport %s", transports[i]);
+ goto next;
}
+
+ if (!(ct->lower_transport & supported)) {
+ GST_WARNING ("unsupported transport %s", transports[i]);
+ goto next;
+ }
+
+ /* we have a valid transport */
+ GST_INFO ("found valid transport %s", transports[i]);
+ have_transport = TRUE;
+ break;
+
+ next:
+ gst_rtsp_transport_init (ct);
}
g_strfreev (transports);
/* we have not found anything usable, error out */
- if (!have_transport)
- goto unsupported_transports;
-
- /* we have a valid transport, check if we can handle it */
- if (ct->trans != GST_RTSP_TRANS_RTP)
- goto unsupported_transports;
- if (ct->profile != GST_RTSP_PROFILE_AVP)
- goto unsupported_transports;
-
- supported = GST_RTSP_LOWER_TRANS_UDP |
- GST_RTSP_LOWER_TRANS_UDP_MCAST |
- GST_RTSP_LOWER_TRANS_TCP;
- if (!(ct->lower_transport & supported))
+ if (!have_transport)
goto unsupported_transports;
if (client->session_pool == NULL)
goto no_pool;
- /* we have a valid transport now, set the destination of the client. */
- g_free (ct->destination);
- url = gst_rtsp_connection_get_url (client->connection);
- ct->destination = g_strdup (url->host);
+ session = state->session;
if (session) {
g_object_ref (session);
/* get a handle to the configuration of the media in the session, this can
* return NULL if this is a new url to manage in this session. */
media = gst_rtsp_session_get_media (session, uri);
-
- need_session = FALSE;
- }
- else {
+ } else {
/* create a session if this fails we probably reached our session limit or
* something. */
if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
goto service_unavailable;
+ state->session = session;
+
/* we need a new media configuration in this session */
media = NULL;
-
- need_session = TRUE;
}
/* we have no media, find one and manage it */
GstRTSPMedia *m;
/* get a handle to the configuration of the media in the session */
- if ((m = find_media (client, uri, request))) {
+ if ((m = find_media (client, state))) {
/* manage the media in our session now */
media = gst_rtsp_session_manage_media (session, uri, m);
}
if (media == NULL)
goto not_found;
+ state->sessmedia = media;
+
+ /* we have a valid transport now, set the destination of the client. */
+ g_free (ct->destination);
+ if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+ ct->destination = gst_rtsp_media_get_multicast_group (media->media);
+ } else {
+ GstRTSPUrl *url;
+
+ url = gst_rtsp_connection_get_url (client->connection);
+ ct->destination = g_strdup (url->host);
+
+ if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
+ /* check if the client selected channels for TCP */
+ if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
+ gst_rtsp_session_media_alloc_channels (media, &ct->interleaved);
+ }
+ }
+ }
+
/* get a handle to the stream in the media */
if (!(stream = gst_rtsp_session_media_get_stream (media, streamid)))
goto no_stream;
st = gst_rtsp_session_stream_set_transport (stream, ct);
+ /* configure keepalive for this transport */
+ gst_rtsp_session_stream_set_keepalive (stream,
+ (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
+
/* serialize the server transport */
trans_str = gst_rtsp_transport_as_text (st);
gst_rtsp_transport_free (st);
/* construct the response now */
code = GST_RTSP_STS_OK;
- gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
+ gst_rtsp_message_init_response (state->response, code,
+ gst_rtsp_status_as_text (code), state->request);
- gst_rtsp_message_add_header (&response, GST_RTSP_HDR_TRANSPORT, trans_str);
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
+ trans_str);
g_free (trans_str);
- send_response (client, session, &response);
+ send_response (client, session, state->response);
/* update the state */
switch (media->state) {
/* ERRORS */
bad_request:
{
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
return FALSE;
}
not_found:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
return FALSE;
}
no_stream:
{
- send_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
- g_object_unref (media);
+ send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
g_object_unref (session);
+ gst_rtsp_transport_free (ct);
return FALSE;
}
no_transport:
{
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
return FALSE;
}
unsupported_transports:
{
- send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
- gst_rtsp_transport_free (ct);
+ send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
+ gst_rtsp_transport_free (ct);
return FALSE;
}
no_pool:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
return FALSE;
}
service_unavailable:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
return FALSE;
}
}
+static GstSDPMessage *
+create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
+{
+ GstSDPMessage *sdp;
+ GstSDPInfo info;
+ const gchar *proto;
+ GstRTSPLowerTrans protocols;
+
+ gst_sdp_message_new (&sdp);
+
+ /* some standard things first */
+ gst_sdp_message_set_version (sdp, "0");
+
+ if (client->is_ipv6)
+ proto = "IP6";
+ else
+ proto = "IP4";
+
+ gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
+ client->server_ip);
+
+ gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
+ gst_sdp_message_set_information (sdp, "rtsp-server");
+ gst_sdp_message_add_time (sdp, "0", "0", NULL);
+ gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
+ gst_sdp_message_add_attribute (sdp, "type", "broadcast");
+ gst_sdp_message_add_attribute (sdp, "control", "*");
+
+ info.server_proto = proto;
+ protocols = gst_rtsp_media_get_protocols (media);
+ if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
+ info.server_ip = gst_rtsp_media_get_multicast_group (media);
+ else
+ info.server_ip = g_strdup (client->server_ip);
+
+ /* create an SDP for the media object */
+ if (!gst_rtsp_sdp_from_media (sdp, &info, media))
+ goto no_sdp;
+
+ g_free (info.server_ip);
+
+ return sdp;
+
+ /* ERRORS */
+no_sdp:
+ {
+ g_free (info.server_ip);
+ gst_sdp_message_free (sdp);
+ return NULL;
+ }
+}
+
/* for the describe we must generate an SDP */
static gboolean
-handle_describe_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
+handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
{
- GstRTSPMessage response = { 0 };
GstRTSPResult res;
GstSDPMessage *sdp;
- guint i;
- gchar *str;
+ guint i, str_len;
+ gchar *str, *content_base;
GstRTSPMedia *media;
/* check what kind of format is accepted, we don't really do anything with it
* and always return SDP for now. */
- for (i = 0; i++; ) {
+ for (i = 0; i++;) {
gchar *accept;
- res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_ACCEPT, &accept, i);
+ res =
+ gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
+ &accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
}
/* find the media object for the uri */
- if (!(media = find_media (client, uri, request)))
+ if (!(media = find_media (client, state)))
goto no_media;
- /* create an SDP for the media object */
- if (!(sdp = gst_rtsp_sdp_from_media (media)))
+ /* create an SDP for the media object on this client */
+ if (!(sdp = create_sdp (client, media)))
goto no_sdp;
g_object_unref (media);
- gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
+ gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
- gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp");
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
+ "application/sdp");
/* content base for some clients that might screw up creating the setup uri */
- str = g_strdup_printf ("rtsp://%s:%u%s/", uri->host, uri->port, uri->abspath);
- gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_BASE, str);
- g_free (str);
+ str = gst_rtsp_url_get_request_uri (state->uri);
+ str_len = strlen (str);
+
+ /* check for trailing '/' and append one */
+ if (str[str_len - 1] != '/') {
+ content_base = g_malloc (str_len + 2);
+ memcpy (content_base, str, str_len);
+ content_base[str_len] = '/';
+ content_base[str_len + 1] = '\0';
+ g_free (str);
+ } else {
+ content_base = str;
+ }
+
+ GST_INFO ("adding content-base: %s", content_base);
+
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
+ content_base);
+ g_free (content_base);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
- gst_rtsp_message_take_body (&response, (guint8 *)str, strlen (str));
+ gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
gst_sdp_message_free (sdp);
- send_response (client, session, &response);
+ send_response (client, state->session, state->response);
return TRUE;
}
no_sdp:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
g_object_unref (media);
return FALSE;
}
}
-static void
-handle_options_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPSession *session, GstRTSPMessage *request)
+static gboolean
+handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
{
- GstRTSPMessage response = { 0 };
GstRTSPMethod options;
gchar *str;
options = GST_RTSP_DESCRIBE |
- GST_RTSP_OPTIONS |
- GST_RTSP_PAUSE |
- GST_RTSP_PLAY |
- GST_RTSP_SETUP |
- GST_RTSP_TEARDOWN;
+ GST_RTSP_OPTIONS |
+ GST_RTSP_PAUSE |
+ GST_RTSP_PLAY |
+ GST_RTSP_SETUP |
+ GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
str = gst_rtsp_options_as_text (options);
- gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
- gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
+ gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
+ gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
- gst_rtsp_message_add_header (&response, GST_RTSP_HDR_PUBLIC, str);
+ gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
- send_response (client, session, &response);
+ send_response (client, state->session, state->response);
+
+ return TRUE;
}
/* remove duplicate and trailing '/' */
static void
-santize_uri (GstRTSPUrl *uri)
+sanitize_uri (GstRTSPUrl * uri)
{
gint i, len;
gchar *s, *d;
}
len = d - uri->abspath;
/* don't remove the first slash if that's the only thing left */
- if (len > 1 && *(d-1) == '/')
+ if (len > 1 && *(d - 1) == '/')
d--;
*d = '\0';
}
static void
-client_session_finalized (GstRTSPClient *client, GstRTSPSession *session)
+client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
{
+ GST_INFO ("client %p: session %p finished", client, session);
+
+ /* unlink all media managed in this session */
+ client_unlink_session (client, session);
+
+ /* remove the session */
if (!(client->sessions = g_list_remove (client->sessions, session))) {
- g_message ("all sessions finalized, close the connection");
- g_source_destroy ((GSource*)client->watch);
+ GST_INFO ("client %p: all sessions finalized, close the connection",
+ client);
+ close_connection (client);
}
}
static void
-client_watch_session (GstRTSPClient *client, GstRTSPSession *session)
+client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
{
GList *walk;
return;
}
- g_message ("watching session %p", session);
+ GST_INFO ("watching session %p", session);
- g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized, client);
+ g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
+ client);
client->sessions = g_list_prepend (client->sessions, session);
}
static void
-handle_request (GstRTSPClient *client, GstRTSPMessage *request)
+handle_request (GstRTSPClient * client, GstRTSPMessage * request)
{
GstRTSPMethod method;
const gchar *uristr;
GstRTSPVersion version;
GstRTSPResult res;
GstRTSPSession *session;
+ GstRTSPClientState state = { NULL };
+ GstRTSPMessage response = { 0 };
gchar *sessid;
-#ifdef DEBUG
- gst_rtsp_message_dump (request);
-#endif
+ state.request = request;
+ state.response = &response;
+
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (request);
+ }
- g_message ("client %p: received a request", client);
+ GST_INFO ("client %p: received a request", client);
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
if (version != GST_RTSP_VERSION_1_0) {
/* we can only handle 1.0 requests */
- send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED, request);
+ send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
+ &state);
return;
}
+ state.method = method;
/* we always try to parse the url first */
- if ((res = gst_rtsp_url_parse (uristr, &uri)) != GST_RTSP_OK) {
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
+ if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
return;
}
/* sanitize the uri */
- santize_uri (uri);
+ sanitize_uri (uri);
+ state.uri = uri;
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
* disappears because it times out, we will be notified. If all sessions are
* gone, we will close the connection */
client_watch_session (client, session);
- }
- else
+ } else
session = NULL;
+ state.session = session;
+
+ if (client->auth) {
+ if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
+ goto not_authorized;
+ }
+
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
- handle_options_request (client, uri, session, request);
+ handle_options_request (client, &state);
break;
case GST_RTSP_DESCRIBE:
- handle_describe_request (client, uri, session, request);
+ handle_describe_request (client, &state);
break;
case GST_RTSP_SETUP:
- handle_setup_request (client, uri, session, request);
+ handle_setup_request (client, &state);
break;
case GST_RTSP_PLAY:
- handle_play_request (client, uri, session, request);
+ handle_play_request (client, &state);
break;
case GST_RTSP_PAUSE:
- handle_pause_request (client, uri, session, request);
+ handle_pause_request (client, &state);
break;
case GST_RTSP_TEARDOWN:
- handle_teardown_request (client, uri, session, request);
+ handle_teardown_request (client, &state);
+ break;
+ case GST_RTSP_SET_PARAMETER:
+ handle_set_param_request (client, &state);
break;
- case GST_RTSP_ANNOUNCE:
case GST_RTSP_GET_PARAMETER:
+ handle_get_param_request (client, &state);
+ break;
+ case GST_RTSP_ANNOUNCE:
case GST_RTSP_RECORD:
case GST_RTSP_REDIRECT:
- case GST_RTSP_SET_PARAMETER:
- send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, request);
+ send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
break;
case GST_RTSP_INVALID:
default:
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
break;
}
if (session)
/* ERRORS */
no_pool:
{
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
return;
}
session_not_found:
{
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
+ return;
+ }
+not_authorized:
+ {
+ handle_unauthorized_request (client, client->auth, &state);
return;
}
}
static void
-handle_data (GstRTSPClient *client, GstRTSPMessage *message)
+handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
GstRTSPResult res;
guint8 channel;
GstBuffer *buffer;
gboolean handled;
- /* find the stream for this message */
+ /* find the stream for this message */
res = gst_rtsp_message_parse_data (message, &channel);
if (res != GST_RTSP_OK)
return;
gst_rtsp_message_steal_body (message, &data, &size);
- buffer = gst_buffer_new ();
- GST_BUFFER_DATA (buffer) = data;
- GST_BUFFER_MALLOCDATA (buffer) = data;
- GST_BUFFER_SIZE (buffer) = size;
+ buffer = gst_buffer_new_wrapped (data, size);
handled = FALSE;
for (walk = client->streams; walk; walk = g_list_next (walk)) {
if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
/* dispatch to the stream based on the channel number */
if (tr->interleaved.min == channel) {
- gst_rtsp_media_stream_rtp (mstream, buffer);
- handled = TRUE;
+ gst_rtsp_media_stream_rtp (mstream, buffer);
+ handled = TRUE;
break;
} else if (tr->interleaved.max == channel) {
- gst_rtsp_media_stream_rtcp (mstream, buffer);
- handled = TRUE;
+ gst_rtsp_media_stream_rtcp (mstream, buffer);
+ handled = TRUE;
break;
}
}
* that created the client but can be overridden later.
*/
void
-gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *pool)
+gst_rtsp_client_set_session_pool (GstRTSPClient * client,
+ GstRTSPSessionPool * pool)
{
GstRTSPSessionPool *old;
* Returns: a #GstRTSPSessionPool, unref after usage.
*/
GstRTSPSessionPool *
-gst_rtsp_client_get_session_pool (GstRTSPClient *client)
+gst_rtsp_client_get_session_pool (GstRTSPClient * client)
{
GstRTSPSessionPool *result;
}
/**
+ * gst_rtsp_client_set_server:
+ * @client: a #GstRTSPClient
+ * @server: a #GstRTSPServer
+ *
+ * Set @server as the server that created @client.
+ */
+void
+gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
+{
+ GstRTSPServer *old;
+
+ old = client->server;
+ if (old != server) {
+ if (server)
+ g_object_ref (server);
+ client->server = server;
+ if (old)
+ g_object_unref (old);
+ }
+}
+
+/**
+ * gst_rtsp_client_get_server:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPServer object that @client was created from.
+ *
+ * Returns: a #GstRTSPServer, unref after usage.
+ */
+GstRTSPServer *
+gst_rtsp_client_get_server (GstRTSPClient * client)
+{
+ GstRTSPServer *result;
+
+ if ((result = client->server))
+ g_object_ref (result);
+
+ return result;
+}
+
+/**
* gst_rtsp_client_set_media_mapping:
* @client: a #GstRTSPClient
* @mapping: a #GstRTSPMediaMapping
* created the client but can be overriden later.
*/
void
-gst_rtsp_client_set_media_mapping (GstRTSPClient *client, GstRTSPMediaMapping *mapping)
+gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
+ GstRTSPMediaMapping * mapping)
{
GstRTSPMediaMapping *old;
* Returns: a #GstRTSPMediaMapping, unref after usage.
*/
GstRTSPMediaMapping *
-gst_rtsp_client_get_media_mapping (GstRTSPClient *client)
+gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
{
GstRTSPMediaMapping *result;
return result;
}
+/**
+ * gst_rtsp_client_set_auth:
+ * @client: a #GstRTSPClient
+ * @auth: a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @client.
+ */
+void
+gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
+{
+ GstRTSPAuth *old;
+
+ g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+ old = client->auth;
+
+ if (old != auth) {
+ if (auth)
+ g_object_ref (auth);
+ client->auth = auth;
+ if (old)
+ g_object_unref (old);
+ }
+}
+
+
+/**
+ * gst_rtsp_client_get_auth:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @client.
+ *
+ * Returns: the #GstRTSPAuth of @client. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_client_get_auth (GstRTSPClient * client)
+{
+ GstRTSPAuth *result;
+
+ g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+ if ((result = client->auth))
+ g_object_ref (result);
+
+ return result;
+}
+
static GstRTSPResult
-message_received (GstRTSPWatch *watch, GstRTSPMessage *message, gpointer user_data)
+message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
+ gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
}
static GstRTSPResult
-message_sent (GstRTSPWatch *watch, guint cseq, gpointer user_data)
+message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
{
- GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ /* GstRTSPClient *client; */
+
+ /* client = GST_RTSP_CLIENT (user_data); */
- g_message ("client %p: sent a message with cseq %d", client, cseq);
+ /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
return GST_RTSP_OK;
}
static GstRTSPResult
-closed (GstRTSPWatch *watch, gpointer user_data)
+closed (GstRTSPWatch * watch, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
const gchar *tunnelid;
- g_message ("client %p: connection closed", client);
+ GST_INFO ("client %p: connection closed", client);
if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
- g_mutex_lock (tunnels_lock);
+ g_mutex_lock (&tunnels_lock);
+ /* remove from tunnelids */
g_hash_table_remove (tunnels, tunnelid);
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
}
- /* remove all streams that are streaming over this client connection */
- unlink_streams (client);
-
return GST_RTSP_OK;
}
static GstRTSPResult
-error (GstRTSPWatch *watch, GstRTSPResult result, gpointer user_data)
+error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
{
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
gchar *str;
str = gst_rtsp_strresult (result);
- g_message ("client %p: received an error %s", client, str);
+ GST_INFO ("client %p: received an error %s", client, str);
g_free (str);
return GST_RTSP_OK;
}
-static GstRTSPStatusCode
-tunnel_start (GstRTSPWatch *watch, gpointer user_data)
+static GstRTSPResult
+error_full (GstRTSPWatch * watch, GstRTSPResult result,
+ GstRTSPMessage * message, guint id, gpointer user_data)
{
- GstRTSPClient *client;
- const gchar *tunnelid;
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ gchar *str;
- client = GST_RTSP_CLIENT (user_data);
+ str = gst_rtsp_strresult (result);
+ GST_INFO
+ ("client %p: received an error %s when handling message %p with id %d",
+ client, str, message, id);
+ g_free (str);
- g_message ("client %p: tunnel start", client);
+ return GST_RTSP_OK;
+}
+
+static gboolean
+remember_tunnel (GstRTSPClient * client)
+{
+ const gchar *tunnelid;
/* store client in the pending tunnels */
tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+ if (tunnelid == NULL)
+ goto no_tunnelid;
- g_message ("client %p: inserting %s", client, tunnelid);
+ GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
/* we can't have two clients connecting with the same tunnelid */
- g_mutex_lock (tunnels_lock);
+ g_mutex_lock (&tunnels_lock);
if (g_hash_table_lookup (tunnels, tunnelid))
goto tunnel_existed;
g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
- g_mutex_unlock (tunnels_lock);
+ g_mutex_unlock (&tunnels_lock);
- return GST_RTSP_STS_OK;
+ return TRUE;
/* ERRORS */
+no_tunnelid:
+ {
+ GST_ERROR ("client %p: no tunnelid provided", client);
+ return FALSE;
+ }
tunnel_existed:
{
- g_mutex_unlock (tunnels_lock);
- g_message ("client %p: tunnel session %s existed", client, tunnelid);
+ g_mutex_unlock (&tunnels_lock);
+ GST_ERROR ("client %p: tunnel session %s already existed", client,
+ tunnelid);
+ return FALSE;
+ }
+}
+
+static GstRTSPStatusCode
+tunnel_start (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client;
+
+ client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel start (connection %p)", client,
+ client->connection);
+
+ if (!remember_tunnel (client))
+ goto tunnel_error;
+
+ return GST_RTSP_STS_OK;
+
+ /* ERRORS */
+tunnel_error:
+ {
+ GST_ERROR ("client %p: error starting tunnel", client);
return GST_RTSP_STS_SERVICE_UNAVAILABLE;
}
}
static GstRTSPResult
-tunnel_complete (GstRTSPWatch *watch, gpointer user_data)
+tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client;
+
+ client = GST_RTSP_CLIENT (user_data);
+
+ GST_INFO ("client %p: tunnel lost (connection %p)", client,
+ client->connection);
+
+ /* ignore error, it'll only be a problem when the client does a POST again */
+ remember_tunnel (client);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
{
const gchar *tunnelid;
GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
GstRTSPClient *oclient;
- g_message ("client %p: tunnel complete", client);
+ GST_INFO ("client %p: tunnel complete", client);
/* find previous tunnel */
tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
+ if (tunnelid == NULL)
+ goto no_tunnelid;
- g_mutex_lock (tunnels_lock);
+ g_mutex_lock (&tunnels_lock);
if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
goto no_tunnel;
* remove the ref to it. */
g_object_ref (oclient);
g_hash_table_remove (tunnels, tunnelid);
- g_mutex_unlock (tunnels_lock);
- g_message ("client %p: found tunnel %p", client, oclient);
+ if (oclient->watch == NULL)
+ goto tunnel_closed;
+ g_mutex_unlock (&tunnels_lock);
+
+ GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
+ oclient->connection, client->connection);
/* merge the tunnels into the first client */
gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
g_object_unref (oclient);
/* we don't need this watch anymore */
- g_source_remove (client->watchid);
+ g_source_destroy ((GSource *) client->watch);
+ client->watchid = 0;
+ client->watch = NULL;
return GST_RTSP_OK;
/* ERRORS */
+no_tunnelid:
+ {
+ GST_INFO ("client %p: no tunnelid provided", client);
+ return GST_RTSP_ERROR;
+ }
no_tunnel:
{
- g_mutex_unlock (tunnels_lock);
- g_message ("client %p: tunnel session %s not found", client, tunnelid);
- return GST_RTSP_OK;
+ g_mutex_unlock (&tunnels_lock);
+ GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
+ return GST_RTSP_ERROR;
+ }
+tunnel_closed:
+ {
+ g_mutex_unlock (&tunnels_lock);
+ GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
+ g_object_unref (oclient);
+ return GST_RTSP_ERROR;
}
}
closed,
error,
tunnel_start,
- tunnel_complete
+ tunnel_complete,
+ error_full,
+ tunnel_lost
};
+static void
+client_watch_notify (GstRTSPClient * client)
+{
+ GST_INFO ("client %p: watch destroyed", client);
+ client->watchid = 0;
+ client->watch = NULL;
+ g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
+ g_object_unref (client);
+}
+
/**
* gst_rtsp_client_attach:
* @client: a #GstRTSPClient
- * @channel: a #GIOChannel
+ * @socket: a #GSocket
+ * @cancellable: a #GCancellable
+ * @error: a #GError
*
- * Accept a new connection for @client on the socket in @source.
+ * Accept a new connection for @client on @socket.
*
* This function should be called when the client properties and urls are fully
* configured and the client is ready to start.
* Returns: %TRUE if the client could be accepted.
*/
gboolean
-gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
+gst_rtsp_client_accept (GstRTSPClient * client, GSocket * socket,
+ GCancellable * cancellable, GError ** error)
{
- int sock;
GstRTSPConnection *conn;
GstRTSPResult res;
+ GSocket *read_socket;
+ GSocketAddress *addres;
GSource *source;
GMainContext *context;
GstRTSPUrl *url;
+ struct sockaddr_storage addr;
+ socklen_t addrlen;
+ gchar ip[INET6_ADDRSTRLEN];
/* a new client connected. */
- sock = g_io_channel_unix_get_fd (channel);
+ GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
+ accept_failed);
+
+ read_socket = gst_rtsp_connection_get_read_socket (conn);
+ client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
+
+ if (!(addres = g_socket_get_remote_address (read_socket, error)))
+ goto no_address;
+
+ addrlen = sizeof (addr);
+ if (!g_socket_address_to_native (addres, &addr, addrlen, error))
+ goto native_failed;
+ g_object_unref (addres);
+
+ if (getnameinfo ((struct sockaddr *) &addr, addrlen, ip, sizeof (ip), NULL, 0,
+ NI_NUMERICHOST) != 0)
+ goto getnameinfo_failed;
- GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
+ /* keep the original ip that the client connected to */
+ g_free (client->server_ip);
+ client->server_ip = g_strndup (ip, sizeof (ip));
+
+ GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
+ client->server_ip, client->is_ipv6);
url = gst_rtsp_connection_get_url (conn);
- g_message ("added new client %p ip %s:%d", client,
- url->host, url->port);
+ GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
client->connection = conn;
/* create watch for the connection and attach */
client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
- g_object_ref (client), g_object_unref);
+ g_object_ref (client), (GDestroyNotify) client_watch_notify);
/* find the context to add the watch */
if ((source = g_main_current_source ()))
else
context = NULL;
- g_message ("attaching to context %p", context);
+ GST_INFO ("attaching to context %p", context);
client->watchid = gst_rtsp_watch_attach (client->watch, context);
gst_rtsp_watch_unref (client->watch);
{
gchar *str = gst_rtsp_strresult (res);
- g_error ("Could not accept client on server socket %d: %s",
- sock, str);
+ GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
g_free (str);
return FALSE;
}
+no_address:
+ {
+ GST_ERROR ("could not get remote address %s", (*error)->message);
+ return FALSE;
+ }
+native_failed:
+ {
+ g_object_unref (addres);
+ GST_ERROR ("could not get native address %s", (*error)->message);
+ return FALSE;
+ }
+getnameinfo_failed:
+ {
+ GST_ERROR ("getnameinfo failed: %s", g_strerror (errno));
+ return FALSE;
+ }
}
projects / platform / upstream / gstreamer.git / blobdiff