* 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"
-#undef DEBUG
-
-#define DEFAULT_TIMEOUT 60
+static GMutex tunnels_lock;
+static GHashTable *tunnels;
enum
{
PROP_0,
- PROP_TIMEOUT,
PROP_SESSION_POOL,
PROP_MEDIA_MAPPING,
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 unlink_session_streams (GstRTSPClient * client,
+ GstRTSPSession * session, GstRTSPSessionMedia * media);
+
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
static void
gobject_class->finalize = gst_rtsp_client_finalize;
g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
- g_param_spec_uint ("timeout", "Timeout", "The client timeout",
- 0, G_MAXUINT, DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
- 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);
+
+ GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
- client->timeout = DEFAULT_TIMEOUT;
+}
+
+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 */
{
GstRTSPClient *client = GST_RTSP_CLIENT (obj);
- g_message ("finalize client %p", client);
+ GST_INFO ("finalize client %p", client);
+
+ 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);
switch (propid) {
- case PROP_TIMEOUT:
- g_value_set_uint (value, gst_rtsp_client_get_timeout (client));
- break;
case PROP_SESSION_POOL:
g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
break;
}
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);
switch (propid) {
- case PROP_TIMEOUT:
- gst_rtsp_client_set_timeout (client, g_value_get_uint (value));
- break;
case PROP_SESSION_POOL:
gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
break;
* 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, GstRTSPMessage *response)
+send_response (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPMessage * response)
{
- GTimeVal timeout;
+ 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);
+
+ /* add the new session header for new session ids */
+ if (session) {
+ gchar *str;
- gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER, "GStreamer RTSP server");
+ if (session->timeout != 60)
+ str =
+ g_strdup_printf ("%s; timeout=%d", session->sessionid,
+ session->timeout);
+ else
+ str = g_strdup (session->sessionid);
-#ifdef DEBUG
- gst_rtsp_message_dump (response);
-#endif
+ gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, str);
+ }
- timeout.tv_sec = client->timeout;
- timeout.tv_usec = 0;
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (response);
+ }
- gst_rtsp_connection_send (client->connection, response, &timeout);
+ 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);
+
+ send_response (client, NULL, state->response);
+}
- gst_rtsp_message_init_response (&response, code,
- gst_rtsp_status_as_text (code), request);
+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, &response);
+ 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;
}
}
-/* Get the session or NULL when there was no session */
-static GstRTSPSession *
-ensure_session (GstRTSPClient *client, GstRTSPMessage *request)
+static gboolean
+do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
{
- GstRTSPResult res;
- GstRTSPSession *session;
- gchar *sessid;
+ GstRTSPMessage message = { 0 };
+ GstMapInfo map_info;
+ guint8 *data;
+ guint usize;
- res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
- if (res == GST_RTSP_OK) {
- if (client->session_pool == NULL)
- goto no_pool;
+ gst_rtsp_message_init_data (&message, channel);
- /* we had a session in the request, find it again */
- if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
- goto session_not_found;
+ if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
+ return FALSE;
- client->timeout = gst_rtsp_session_get_timeout (session);
- }
- else
- goto service_unavailable;
+ gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
- return session;
+ /* 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);
- /* ERRORS */
-no_pool:
- {
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
- return NULL;
+ gst_rtsp_message_steal_body (&message, &data, &usize);
+ gst_buffer_unmap (buffer, &map_info);
+
+ gst_rtsp_message_unset (&message);
+
+ return TRUE;
+}
+
+static void
+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);
+ 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, GstRTSPSession * session,
+ GstRTSPSessionStream * stream)
+{
+ 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_session_streams (GstRTSPClient * client, GstRTSPSession * session,
+ GstRTSPSessionMedia * media)
+{
+ guint n_streams, i;
+
+ n_streams = gst_rtsp_media_n_streams (media->media);
+ for (i = 0; i < n_streams; i++) {
+ GstRTSPSessionStream *sstream;
+ GstRTSPTransport *tr;
+
+ /* 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))
+ continue;
+
+ if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+ /* for TCP, unlink the stream from the TCP connection of the client */
+ unlink_stream (client, session, sstream);
+ }
}
-session_not_found:
- {
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
- return NULL;
+}
+
+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);
}
-service_unavailable:
- {
- send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
- return NULL;
+
+ 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, GstRTSPMessage *request)
+handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
{
- GstRTSPSessionMedia *media;
GstRTSPSession *session;
- GstRTSPMessage response = { 0 };
+ GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
- if (!(session = ensure_session (client, request)))
+ 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;
- gst_rtsp_session_media_stop (media);
+ state->sessmedia = media;
+
+ /* unlink the all TCP callbacks */
+ 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);
+ client->sessions = g_list_remove (client->sessions, session);
+
+ gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
/* unmanage the media in the session, returns false if all media session
* are torn down. */
if (!gst_rtsp_session_release_media (session, media)) {
/* remove the session */
gst_rtsp_session_pool_remove (client->session_pool, session);
-
- /* remove the session id from the request, which will also remove it from the
- * response */
- gst_rtsp_message_remove_header (request, GST_RTSP_HDR_SESSION, -1);
}
- g_object_unref (session);
-
/* 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, &response);
+ close_connection (client);
- return FALSE;
+ return TRUE;
/* ERRORS */
no_session:
{
- /* error was sent already */
+ 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_pause_request (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
+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, GstRTSPClientState * state)
{
- GstRTSPSessionMedia *media;
GstRTSPSession *session;
- GstRTSPMessage response = { 0 };
+ GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
- if (!(session = ensure_session (client, request)))
+ 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;
- gst_rtsp_session_media_pause (media);
+ /* unlink the all TCP callbacks */
+ 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, &response);
+ send_response (client, session, state->response);
/* the state is now READY */
media->state = GST_RTSP_STATE_READY;
- g_object_unref (session);
- return FALSE;
+ return TRUE;
/* ERRORS */
no_session:
{
+ 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);
- g_object_unref (session);
+ 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);
- g_object_unref (session);
+ 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, GstRTSPMessage *request)
+handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
{
- GstRTSPSessionMedia *media;
GstRTSPSession *session;
- GstRTSPMessage response = { 0 };
+ GstRTSPSessionMedia *media;
GstRTSPStatusCode code;
GString *rtpinfo;
- guint n_streams, i;
+ guint n_streams, i, infocount;
guint timestamp, seqnum;
gchar *str;
+ GstRTSPTimeRange *range;
+ GstRTSPResult res;
- if (!(session = ensure_session (client, request)))
+ 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 (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 */
+ gst_rtsp_media_seek (media->media, range);
+ gst_rtsp_range_free (range);
+ }
+ }
+
/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
n_streams = gst_rtsp_media_n_streams (media->media);
- for (i = 0; i < n_streams; i++) {
+ for (i = 0, infocount = 0; i < n_streams; i++) {
+ GstRTSPSessionStream *sstream;
GstRTSPMediaStream *stream;
+ GstRTSPTransport *tr;
+ GObjectClass *payobjclass;
gchar *uristr;
- stream = gst_rtsp_media_get_stream (media->media, i);
+ /* 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)) {
+ 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, 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")) {
+ GObject *payobj;
+
+ payobj = G_OBJECT (stream->payloader);
+
+ /* only add RTP-Info for streams with seqnum and timestamp */
+ g_object_get (payobj, "seqnum", &seqnum, "timestamp", ×tamp, NULL);
- g_object_get (G_OBJECT (stream->payloader), "seqnum", &seqnum, NULL);
- g_object_get (G_OBJECT (stream->payloader), "timestamp", ×tamp, NULL);
+ if (infocount > 0)
+ g_string_append (rtpinfo, ", ");
- if (i > 0)
- g_string_append (rtpinfo, ", ");
+ 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);
- 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);
- 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, &response);
+ send_response (client, session, state->response);
/* start playing after sending the request */
- gst_rtsp_session_media_play (media);
+ gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
media->state = GST_RTSP_STATE_PLAYING;
- g_object_unref (session);
- return FALSE;
+ return TRUE;
/* ERRORS */
no_session:
{
- /* error was sent */
+ 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);
- g_object_unref (session);
+ 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);
- g_object_unref (session);
+ 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, GstRTSPMessage *request)
+handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
{
GstRTSPResult res;
- gchar *sessid;
+ GstRTSPUrl *uri;
gchar *transport;
gchar **transports;
gboolean have_transport;
GstRTSPTransport *ct, *st;
- GstRTSPSession *session;
gint i;
GstRTSPLowerTrans supported;
- GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
+ GstRTSPSession *session;
GstRTSPSessionStream *stream;
gchar *trans_str, *pos;
guint streamid;
GstRTSPSessionMedia *media;
- gboolean need_session;
+
+ 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);
- ct->destination = g_strdup (inet_ntoa (client->address.sin_addr));
-
- /* a setup request creates a session for a client, check if the client already
- * sent a session id to us */
- res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
- if (res == GST_RTSP_OK) {
- /* we had a session in the request, find it again */
- if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
- goto session_not_found;
+ 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;
- /* setup the server transport from the client transport */
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);
-
- /* add the new session header for new session ids */
- if (need_session) {
- gchar *str;
-
- if (session->timeout != 60)
- 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);
- }
+ 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, &response);
+ send_response (client, session, state->response);
/* update the state */
switch (media->state) {
media->state = GST_RTSP_STATE_READY;
break;
}
-
g_object_unref (session);
return TRUE;
/* 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);
- return FALSE;
- }
-session_not_found:
- {
- send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
+ 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, 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, &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, 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, &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';
}
-/* this function runs in a client specific thread and handles all rtsp messages
- * with the client */
-static gpointer
-handle_client (GstRTSPClient *client)
+static void
+client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
{
- GstRTSPMessage request = { 0 };
- GstRTSPResult res;
- GstRTSPMethod method;
- const gchar *uristr;
- GstRTSPUrl *uri;
- GstRTSPVersion version;
+ GST_INFO ("client %p: session %p finished", client, session);
- while (TRUE) {
- GTimeVal timeout;
+ /* unlink all media managed in this session */
+ client_unlink_session (client, session);
- timeout.tv_sec = client->timeout;
+ /* remove the session */
+ if (!(client->sessions = g_list_remove (client->sessions, session))) {
+ GST_INFO ("client %p: all sessions finalized, close the connection",
+ client);
+ close_connection (client);
+ }
+}
+
+static void
+client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+ GList *walk;
- /* start by waiting for a message from the client */
- res = gst_rtsp_connection_receive (client->connection, &request, &timeout);
- if (res < 0)
- goto receive_failed;
+ for (walk = client->sessions; walk; walk = g_list_next (walk)) {
+ GstRTSPSession *msession = (GstRTSPSession *) walk->data;
-#ifdef DEBUG
- gst_rtsp_message_dump (&request);
-#endif
+ /* we already know about this session */
+ if (msession == session)
+ return;
+ }
- gst_rtsp_message_parse_request (&request, &method, &uristr, &version);
+ GST_INFO ("watching session %p", session);
- 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);
- continue;
- }
+ g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
+ client);
+ client->sessions = g_list_prepend (client->sessions, session);
+}
- /* 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);
- continue;
- }
+static void
+handle_request (GstRTSPClient * client, GstRTSPMessage * request)
+{
+ GstRTSPMethod method;
+ const gchar *uristr;
+ GstRTSPUrl *uri;
+ GstRTSPVersion version;
+ GstRTSPResult res;
+ GstRTSPSession *session;
+ GstRTSPClientState state = { NULL };
+ GstRTSPMessage response = { 0 };
+ gchar *sessid;
- /* sanitize the uri */
- santize_uri (uri);
+ state.request = request;
+ state.response = &response;
- /* now see what is asked and dispatch to a dedicated handler */
- switch (method) {
- case GST_RTSP_OPTIONS:
- handle_options_request (client, uri, &request);
- break;
- case GST_RTSP_DESCRIBE:
- handle_describe_request (client, uri, &request);
- break;
- case GST_RTSP_SETUP:
- handle_setup_request (client, uri, &request);
- break;
- case GST_RTSP_PLAY:
- handle_play_request (client, uri, &request);
- break;
- case GST_RTSP_PAUSE:
- handle_pause_request (client, uri, &request);
- break;
- case GST_RTSP_TEARDOWN:
- handle_teardown_request (client, uri, &request);
- break;
- case GST_RTSP_ANNOUNCE:
- case GST_RTSP_GET_PARAMETER:
- case GST_RTSP_RECORD:
- case GST_RTSP_REDIRECT:
- case GST_RTSP_SET_PARAMETER:
- send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &request);
- break;
- case GST_RTSP_INVALID:
- default:
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &request);
- break;
- }
- gst_rtsp_url_free (uri);
+ if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+ gst_rtsp_message_dump (request);
}
- g_object_unref (client);
- return NULL;
- /* ERRORS */
-receive_failed:
- {
- gchar *str;
- str = gst_rtsp_strresult (res);
- g_message ("receive failed %d (%s), disconnect client %p", res,
- str, client);
- g_free (str);
- gst_rtsp_message_unset (&request);
- gst_rtsp_connection_close (client->connection);
- g_object_unref (client);
- return NULL;
+ 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,
+ &state);
+ return;
}
-}
+ state.method = method;
-/* called when we need to accept a new request from a client */
-static gboolean
-client_accept (GstRTSPClient *client, GIOChannel *channel)
-{
- /* a new client connected. */
- int server_sock_fd, fd;
- unsigned int address_len;
- GstRTSPConnection *conn;
+ /* we always try to parse the url first */
+ if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
+ return;
+ }
- server_sock_fd = g_io_channel_unix_get_fd (channel);
+ /* sanitize the uri */
+ sanitize_uri (uri);
+ state.uri = uri;
- address_len = sizeof (client->address);
- memset (&client->address, 0, address_len);
+ /* get the session if there is any */
+ res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
+ if (res == GST_RTSP_OK) {
+ if (client->session_pool == NULL)
+ goto no_pool;
- fd = accept (server_sock_fd, (struct sockaddr *) &client->address,
- &address_len);
- if (fd == -1)
- goto accept_failed;
+ /* we had a session in the request, find it again */
+ if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
+ goto session_not_found;
- /* now create the connection object */
- gst_rtsp_connection_create (NULL, &conn);
- conn->fd.fd = fd;
+ /* we add the session to the client list of watched sessions. When a session
+ * 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
+ session = NULL;
- /* FIXME some hackery, we need to have a connection method to accept server
- * connections */
- gst_poll_add_fd (conn->fdset, &conn->fd);
+ state.session = session;
- g_message ("added new client %p ip %s with fd %d", client,
- inet_ntoa (client->address.sin_addr), conn->fd.fd);
+ if (client->auth) {
+ if (!gst_rtsp_auth_check (client->auth, client, 0, &state))
+ goto not_authorized;
+ }
- client->connection = conn;
+ /* now see what is asked and dispatch to a dedicated handler */
+ switch (method) {
+ case GST_RTSP_OPTIONS:
+ handle_options_request (client, &state);
+ break;
+ case GST_RTSP_DESCRIBE:
+ handle_describe_request (client, &state);
+ break;
+ case GST_RTSP_SETUP:
+ handle_setup_request (client, &state);
+ break;
+ case GST_RTSP_PLAY:
+ handle_play_request (client, &state);
+ break;
+ case GST_RTSP_PAUSE:
+ handle_pause_request (client, &state);
+ break;
+ case GST_RTSP_TEARDOWN:
+ handle_teardown_request (client, &state);
+ break;
+ case GST_RTSP_SET_PARAMETER:
+ handle_set_param_request (client, &state);
+ break;
+ case GST_RTSP_GET_PARAMETER:
+ handle_get_param_request (client, &state);
+ break;
+ case GST_RTSP_ANNOUNCE:
+ case GST_RTSP_RECORD:
+ case GST_RTSP_REDIRECT:
+ 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, &state);
+ break;
+ }
+ if (session)
+ g_object_unref (session);
- return TRUE;
+ gst_rtsp_url_free (uri);
+ return;
/* ERRORS */
-accept_failed:
+no_pool:
{
- g_error ("Could not accept client on server socket %d: %s (%d)",
- server_sock_fd, g_strerror (errno), errno);
- return FALSE;
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
+ return;
+ }
+session_not_found:
+ {
+ send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
+ return;
+ }
+not_authorized:
+ {
+ handle_unauthorized_request (client, client->auth, &state);
+ return;
}
}
-/**
- * gst_rtsp_client_set_timeout:
- * @client: a #GstRTSPClient
- * @timeout: a timeout in seconds
- *
- * Set the connection timeout to @timeout seconds for @client.
- */
-void
-gst_rtsp_client_set_timeout (GstRTSPClient *client, guint timeout)
+static void
+handle_data (GstRTSPClient * client, GstRTSPMessage * message)
{
- client->timeout = timeout;
-}
+ GstRTSPResult res;
+ guint8 channel;
+ GList *walk;
+ guint8 *data;
+ guint size;
+ GstBuffer *buffer;
+ gboolean handled;
+
+ /* find the stream for this message */
+ res = gst_rtsp_message_parse_data (message, &channel);
+ if (res != GST_RTSP_OK)
+ return;
-/**
- * gst_rtsp_client_get_timeout:
- * @client: a #GstRTSPClient
- *
- * Get the connection timeout @client.
- *
- * Returns: the connection timeout for @client in seconds.
- */
-guint
-gst_rtsp_client_get_timeout (GstRTSPClient *client)
-{
- return client->timeout;
+ gst_rtsp_message_steal_body (message, &data, &size);
+
+ buffer = gst_buffer_new_wrapped (data, size);
+
+ handled = FALSE;
+ for (walk = client->streams; walk; walk = g_list_next (walk)) {
+ GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
+ GstRTSPMediaStream *mstream;
+ GstRTSPTransport *tr;
+
+ /* get the transport, if there is no transport configured, skip this stream */
+ if (!(tr = stream->trans.transport))
+ continue;
+
+ /* we also need a media stream */
+ if (!(mstream = stream->media_stream))
+ continue;
+
+ /* check for TCP transport */
+ 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;
+ break;
+ } else if (tr->interleaved.max == channel) {
+ gst_rtsp_media_stream_rtcp (mstream, buffer);
+ handled = TRUE;
+ break;
+ }
+ }
+ }
+ if (!handled)
+ gst_buffer_unref (buffer);
}
/**
* 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;
}
/**
+ * 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)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+ switch (message->type) {
+ case GST_RTSP_MESSAGE_REQUEST:
+ handle_request (client, message);
+ break;
+ case GST_RTSP_MESSAGE_RESPONSE:
+ break;
+ case GST_RTSP_MESSAGE_DATA:
+ handle_data (client, message);
+ break;
+ default:
+ break;
+ }
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
+{
+ /* GstRTSPClient *client; */
+
+ /* client = GST_RTSP_CLIENT (user_data); */
+
+ /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+closed (GstRTSPWatch * watch, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ const gchar *tunnelid;
+
+ GST_INFO ("client %p: connection closed", 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);
+ }
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ gchar *str;
+
+ str = gst_rtsp_strresult (result);
+ GST_INFO ("client %p: received an error %s", client, str);
+ g_free (str);
+
+ return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error_full (GstRTSPWatch * watch, GstRTSPResult result,
+ GstRTSPMessage * message, guint id, gpointer user_data)
+{
+ GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+ gchar *str;
+
+ 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);
+
+ 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;
+
+ 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);
+ 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);
+
+ return TRUE;
+
+ /* ERRORS */
+no_tunnelid:
+ {
+ GST_ERROR ("client %p: no tunnelid provided", client);
+ return FALSE;
+ }
+tunnel_existed:
+ {
+ 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_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;
+
+ 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);
+ if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
+ goto no_tunnel;
+
+ /* remove the old client from the table. ref before because removing it will
+ * remove the ref to it. */
+ g_object_ref (oclient);
+ g_hash_table_remove (tunnels, tunnelid);
+
+ 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);
+ gst_rtsp_watch_reset (oclient->watch);
+ g_object_unref (oclient);
+
+ /* we don't need this watch anymore */
+ 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);
+ 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;
+ }
+}
+
+static GstRTSPWatchFuncs watch_funcs = {
+ message_received,
+ message_sent,
+ closed,
+ error,
+ tunnel_start,
+ 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)
{
- GError *error = NULL;
+ 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. */
+ 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 (!client_accept (client, channel))
- goto accept_failed;
+ if (!(addres = g_socket_get_remote_address (read_socket, error)))
+ goto no_address;
- /* client accepted, spawn a thread for the client, we don't need to join the
- * thread */
- g_object_ref (client);
- client->thread = g_thread_create ((GThreadFunc)handle_client, client, FALSE, &error);
- if (client->thread == NULL)
- goto no_thread;
+ 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;
+
+ /* 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);
+ 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), (GDestroyNotify) client_watch_notify);
+
+ /* find the context to add the watch */
+ if ((source = g_main_current_source ()))
+ context = g_source_get_context (source);
+ else
+ context = NULL;
+
+ GST_INFO ("attaching to context %p", context);
+
+ client->watchid = gst_rtsp_watch_attach (client->watch, context);
+ gst_rtsp_watch_unref (client->watch);
return TRUE;
/* ERRORS */
accept_failed:
{
+ gchar *str = gst_rtsp_strresult (res);
+
+ GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
+ g_free (str);
return FALSE;
}
-no_thread:
+no_address:
{
- if (error) {
- g_warning ("could not create thread for client %p: %s", client, error->message);
- g_error_free (error);
- }
- g_object_unref (client);
+ 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