2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
26 #include <sys/types.h>
27 #include <netinet/in.h>
29 #include <sys/socket.h>
32 #include <arpa/inet.h>
33 #include <sys/ioctl.h>
35 #include "rtsp-client.h"
37 #include "rtsp-params.h"
39 static GMutex *tunnels_lock;
40 static GHashTable *tunnels;
56 GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
57 #define GST_CAT_DEFAULT rtsp_client_debug
59 static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
61 static void gst_rtsp_client_get_property (GObject * object, guint propid,
62 GValue * value, GParamSpec * pspec);
63 static void gst_rtsp_client_set_property (GObject * object, guint propid,
64 const GValue * value, GParamSpec * pspec);
65 static void gst_rtsp_client_finalize (GObject * obj);
67 static void client_session_finalized (GstRTSPClient * client,
68 GstRTSPSession * session);
69 static void unlink_session_streams (GstRTSPClient * client,
70 GstRTSPSession * session, GstRTSPSessionMedia * media);
72 G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
75 gst_rtsp_client_class_init (GstRTSPClientClass * klass)
77 GObjectClass *gobject_class;
79 gobject_class = G_OBJECT_CLASS (klass);
81 gobject_class->get_property = gst_rtsp_client_get_property;
82 gobject_class->set_property = gst_rtsp_client_set_property;
83 gobject_class->finalize = gst_rtsp_client_finalize;
85 g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
86 g_param_spec_object ("session-pool", "Session Pool",
87 "The session pool to use for client session",
88 GST_TYPE_RTSP_SESSION_POOL,
89 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
91 g_object_class_install_property (gobject_class, PROP_MEDIA_MAPPING,
92 g_param_spec_object ("media-mapping", "Media Mapping",
93 "The media mapping to use for client session",
94 GST_TYPE_RTSP_MEDIA_MAPPING,
95 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
97 gst_rtsp_client_signals[SIGNAL_CLOSED] =
98 g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
99 G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
100 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
103 g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
104 tunnels_lock = g_mutex_new ();
106 GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
110 gst_rtsp_client_init (GstRTSPClient * client)
115 client_unlink_session (GstRTSPClient * client, GstRTSPSession * session)
119 /* unlink all media managed in this session */
120 for (medias = session->medias; medias; medias = g_list_next (medias)) {
121 GstRTSPSessionMedia *media = medias->data;
123 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
124 unlink_session_streams (client, session, media);
125 /* unmanage the media in the session. */
126 gst_rtsp_session_release_media (session, media);
131 client_cleanup_sessions (GstRTSPClient * client)
135 /* remove weak-ref from sessions */
136 for (sessions = client->sessions; sessions; sessions = g_list_next (sessions)) {
137 GstRTSPSession *session = (GstRTSPSession *) sessions->data;
138 g_object_weak_unref (G_OBJECT (session),
139 (GWeakNotify) client_session_finalized, client);
140 client_unlink_session (client, session);
142 g_list_free (client->sessions);
143 client->sessions = NULL;
146 /* A client is finalized when the connection is broken */
148 gst_rtsp_client_finalize (GObject * obj)
150 GstRTSPClient *client = GST_RTSP_CLIENT (obj);
152 GST_INFO ("finalize client %p", client);
154 client_cleanup_sessions (client);
156 gst_rtsp_connection_free (client->connection);
157 if (client->session_pool)
158 g_object_unref (client->session_pool);
159 if (client->media_mapping)
160 g_object_unref (client->media_mapping);
162 g_object_unref (client->auth);
165 gst_rtsp_url_free (client->uri);
167 g_object_unref (client->media);
169 g_free (client->server_ip);
171 G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
175 gst_rtsp_client_get_property (GObject * object, guint propid,
176 GValue * value, GParamSpec * pspec)
178 GstRTSPClient *client = GST_RTSP_CLIENT (object);
181 case PROP_SESSION_POOL:
182 g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
184 case PROP_MEDIA_MAPPING:
185 g_value_take_object (value, gst_rtsp_client_get_media_mapping (client));
188 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
193 gst_rtsp_client_set_property (GObject * object, guint propid,
194 const GValue * value, GParamSpec * pspec)
196 GstRTSPClient *client = GST_RTSP_CLIENT (object);
199 case PROP_SESSION_POOL:
200 gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
202 case PROP_MEDIA_MAPPING:
203 gst_rtsp_client_set_media_mapping (client, g_value_get_object (value));
206 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
211 * gst_rtsp_client_new:
213 * Create a new #GstRTSPClient instance.
215 * Returns: a new #GstRTSPClient
218 gst_rtsp_client_new (void)
220 GstRTSPClient *result;
222 result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
228 send_response (GstRTSPClient * client, GstRTSPSession * session,
229 GstRTSPMessage * response)
231 gst_rtsp_message_add_header (response, GST_RTSP_HDR_SERVER,
232 "GStreamer RTSP server");
234 /* remove any previous header */
235 gst_rtsp_message_remove_header (response, GST_RTSP_HDR_SESSION, -1);
237 /* add the new session header for new session ids */
241 if (session->timeout != 60)
243 g_strdup_printf ("%s; timeout=%d", session->sessionid,
246 str = g_strdup (session->sessionid);
248 gst_rtsp_message_take_header (response, GST_RTSP_HDR_SESSION, str);
251 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
252 gst_rtsp_message_dump (response);
255 gst_rtsp_watch_send_message (client->watch, response, NULL);
256 gst_rtsp_message_unset (response);
260 send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
261 GstRTSPClientState * state)
263 gst_rtsp_message_init_response (state->response, code,
264 gst_rtsp_status_as_text (code), state->request);
266 send_response (client, NULL, state->response);
270 handle_unauthorized_request (GstRTSPClient * client, GstRTSPAuth * auth,
271 GstRTSPClientState * state)
273 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_UNAUTHORIZED,
274 gst_rtsp_status_as_text (GST_RTSP_STS_UNAUTHORIZED), state->request);
277 /* and let the authentication manager setup the auth tokens */
278 gst_rtsp_auth_setup_auth (auth, client, 0, state);
281 send_response (client, state->session, state->response);
286 compare_uri (const GstRTSPUrl * uri1, const GstRTSPUrl * uri2)
288 if (uri1 == NULL || uri2 == NULL)
291 if (strcmp (uri1->abspath, uri2->abspath))
297 /* this function is called to initially find the media for the DESCRIBE request
298 * but is cached for when the same client (without breaking the connection) is
299 * doing a setup for the exact same url. */
300 static GstRTSPMedia *
301 find_media (GstRTSPClient * client, GstRTSPClientState * state)
303 GstRTSPMediaFactory *factory;
307 if (!compare_uri (client->uri, state->uri)) {
308 /* remove any previously cached values before we try to construct a new
311 gst_rtsp_url_free (client->uri);
314 g_object_unref (client->media);
315 client->media = NULL;
317 if (!client->media_mapping)
320 /* find the factory for the uri first */
322 gst_rtsp_media_mapping_find_factory (client->media_mapping,
326 state->factory = factory;
328 /* check if we have access to the factory */
329 if ((auth = gst_rtsp_media_factory_get_auth (factory))) {
330 if (!gst_rtsp_auth_check (auth, client, 0, state))
333 g_object_unref (auth);
336 /* prepare the media and add it to the pipeline */
337 if (!(media = gst_rtsp_media_factory_construct (factory, state->uri)))
340 g_object_unref (factory);
342 state->factory = NULL;
344 /* set ipv6 on the media before preparing */
345 media->is_ipv6 = client->is_ipv6;
346 state->media = media;
348 /* prepare the media */
349 if (!(gst_rtsp_media_prepare (media)))
352 /* now keep track of the uri and the media */
353 client->uri = gst_rtsp_url_copy (state->uri);
354 client->media = media;
356 /* we have seen this uri before, used cached media */
357 media = client->media;
358 state->media = media;
359 GST_INFO ("reusing cached media %p", media);
363 g_object_ref (media);
370 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
375 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
380 handle_unauthorized_request (client, auth, state);
381 g_object_unref (factory);
382 g_object_unref (auth);
387 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
388 g_object_unref (factory);
393 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
394 g_object_unref (media);
400 do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
402 GstRTSPMessage message = { 0 };
406 gst_rtsp_message_init_data (&message, channel);
408 data = GST_BUFFER_DATA (buffer);
409 size = GST_BUFFER_SIZE (buffer);
410 gst_rtsp_message_take_body (&message, data, size);
412 /* FIXME, client->watch could have been finalized here, we need to keep an
413 * extra refcount to the watch. */
414 gst_rtsp_watch_send_message (client->watch, &message, NULL);
416 gst_rtsp_message_steal_body (&message, &data, &size);
417 gst_rtsp_message_unset (&message);
423 do_send_data_list (GstBufferList * blist, guint8 channel,
424 GstRTSPClient * client)
426 GstBufferListIterator *it;
428 it = gst_buffer_list_iterate (blist);
429 while (gst_buffer_list_iterator_next_group (it)) {
430 GstBuffer *group = gst_buffer_list_iterator_merge_group (it);
435 do_send_data (group, channel, client);
437 gst_buffer_list_iterator_free (it);
443 link_stream (GstRTSPClient * client, GstRTSPSession * session,
444 GstRTSPSessionStream * stream)
446 GST_DEBUG ("client %p: linking stream %p", client, stream);
447 gst_rtsp_session_stream_set_callbacks (stream, (GstRTSPSendFunc) do_send_data,
448 (GstRTSPSendFunc) do_send_data, (GstRTSPSendListFunc) do_send_data_list,
449 (GstRTSPSendListFunc) do_send_data_list, client, NULL);
450 client->streams = g_list_prepend (client->streams, stream);
451 /* make sure our session can't expire */
452 gst_rtsp_session_prevent_expire (session);
456 unlink_stream (GstRTSPClient * client, GstRTSPSession * session,
457 GstRTSPSessionStream * stream)
459 GST_DEBUG ("client %p: unlinking stream %p", client, stream);
460 gst_rtsp_session_stream_set_callbacks (stream, NULL, NULL, NULL, NULL, NULL,
462 client->streams = g_list_remove (client->streams, stream);
463 /* our session can now expire */
464 gst_rtsp_session_allow_expire (session);
468 unlink_session_streams (GstRTSPClient * client, GstRTSPSession * session,
469 GstRTSPSessionMedia * media)
473 n_streams = gst_rtsp_media_n_streams (media->media);
474 for (i = 0; i < n_streams; i++) {
475 GstRTSPSessionStream *sstream;
476 GstRTSPTransport *tr;
478 /* get the stream as configured in the session */
479 sstream = gst_rtsp_session_media_get_stream (media, i);
480 /* get the transport, if there is no transport configured, skip this stream */
481 if (!(tr = sstream->trans.transport))
484 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
485 /* for TCP, unlink the stream from the TCP connection of the client */
486 unlink_stream (client, session, sstream);
492 close_connection (GstRTSPClient * client)
494 const gchar *tunnelid;
496 GST_DEBUG ("client %p: closing connection", client);
498 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
499 g_mutex_lock (tunnels_lock);
500 /* remove from tunnelids */
501 g_hash_table_remove (tunnels, tunnelid);
502 g_mutex_unlock (tunnels_lock);
505 gst_rtsp_connection_close (client->connection);
506 if (client->watchid) {
507 g_source_destroy ((GSource *) client->watch);
509 client->watch = NULL;
514 handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
516 GstRTSPSession *session;
517 GstRTSPSessionMedia *media;
518 GstRTSPStatusCode code;
523 session = state->session;
525 /* get a handle to the configuration of the media in the session */
526 media = gst_rtsp_session_get_media (session, state->uri);
530 state->sessmedia = media;
532 /* unlink the all TCP callbacks */
533 unlink_session_streams (client, session, media);
535 /* remove the session from the watched sessions */
536 g_object_weak_unref (G_OBJECT (session),
537 (GWeakNotify) client_session_finalized, client);
538 client->sessions = g_list_remove (client->sessions, session);
540 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
542 /* unmanage the media in the session, returns false if all media session
544 if (!gst_rtsp_session_release_media (session, media)) {
545 /* remove the session */
546 gst_rtsp_session_pool_remove (client->session_pool, session);
548 /* construct the response now */
549 code = GST_RTSP_STS_OK;
550 gst_rtsp_message_init_response (state->response, code,
551 gst_rtsp_status_as_text (code), state->request);
553 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
556 send_response (client, session, state->response);
558 close_connection (client);
565 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
570 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
576 handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
582 res = gst_rtsp_message_get_body (state->request, &data, &size);
583 if (res != GST_RTSP_OK)
587 /* no body, keep-alive request */
588 send_generic_response (client, GST_RTSP_STS_OK, state);
590 /* there is a body, handle the params */
591 res = gst_rtsp_params_get (client, state);
592 if (res != GST_RTSP_OK)
595 send_response (client, state->session, state->response);
602 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
608 handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
614 res = gst_rtsp_message_get_body (state->request, &data, &size);
615 if (res != GST_RTSP_OK)
619 /* no body, keep-alive request */
620 send_generic_response (client, GST_RTSP_STS_OK, state);
622 /* there is a body, handle the params */
623 res = gst_rtsp_params_set (client, state);
624 if (res != GST_RTSP_OK)
627 send_response (client, state->session, state->response);
634 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
640 handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
642 GstRTSPSession *session;
643 GstRTSPSessionMedia *media;
644 GstRTSPStatusCode code;
646 if (!(session = state->session))
649 /* get a handle to the configuration of the media in the session */
650 media = gst_rtsp_session_get_media (session, state->uri);
654 state->sessmedia = media;
656 /* the session state must be playing or recording */
657 if (media->state != GST_RTSP_STATE_PLAYING &&
658 media->state != GST_RTSP_STATE_RECORDING)
661 /* unlink the all TCP callbacks */
662 unlink_session_streams (client, session, media);
664 /* then pause sending */
665 gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
667 /* construct the response now */
668 code = GST_RTSP_STS_OK;
669 gst_rtsp_message_init_response (state->response, code,
670 gst_rtsp_status_as_text (code), state->request);
672 send_response (client, session, state->response);
674 /* the state is now READY */
675 media->state = GST_RTSP_STATE_READY;
682 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
687 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
692 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
699 handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
701 GstRTSPSession *session;
702 GstRTSPSessionMedia *media;
703 GstRTSPStatusCode code;
705 guint n_streams, i, infocount;
706 guint timestamp, seqnum;
708 GstRTSPTimeRange *range;
711 if (!(session = state->session))
714 /* get a handle to the configuration of the media in the session */
715 media = gst_rtsp_session_get_media (session, state->uri);
719 state->sessmedia = media;
721 /* the session state must be playing or ready */
722 if (media->state != GST_RTSP_STATE_PLAYING &&
723 media->state != GST_RTSP_STATE_READY)
726 /* parse the range header if we have one */
728 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
729 if (res == GST_RTSP_OK) {
730 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
731 /* we have a range, seek to the position */
732 gst_rtsp_media_seek (media->media, range);
733 gst_rtsp_range_free (range);
737 /* grab RTPInfo from the payloaders now */
738 rtpinfo = g_string_new ("");
740 n_streams = gst_rtsp_media_n_streams (media->media);
741 for (i = 0, infocount = 0; i < n_streams; i++) {
742 GstRTSPSessionStream *sstream;
743 GstRTSPMediaStream *stream;
744 GstRTSPTransport *tr;
745 GObjectClass *payobjclass;
748 /* get the stream as configured in the session */
749 sstream = gst_rtsp_session_media_get_stream (media, i);
750 /* get the transport, if there is no transport configured, skip this stream */
751 if (!(tr = sstream->trans.transport)) {
752 GST_INFO ("stream %d is not configured", i);
756 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
757 /* for TCP, link the stream to the TCP connection of the client */
758 link_stream (client, session, sstream);
761 stream = sstream->media_stream;
763 payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
765 if (g_object_class_find_property (payobjclass, "seqnum") &&
766 g_object_class_find_property (payobjclass, "timestamp")) {
769 payobj = G_OBJECT (stream->payloader);
771 /* only add RTP-Info for streams with seqnum and timestamp */
772 g_object_get (payobj, "seqnum", &seqnum, "timestamp", ×tamp, NULL);
775 g_string_append (rtpinfo, ", ");
777 uristr = gst_rtsp_url_get_request_uri (state->uri);
778 g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
779 uristr, i, seqnum, timestamp);
784 GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
788 /* construct the response now */
789 code = GST_RTSP_STS_OK;
790 gst_rtsp_message_init_response (state->response, code,
791 gst_rtsp_status_as_text (code), state->request);
793 /* add the RTP-Info header */
795 str = g_string_free (rtpinfo, FALSE);
796 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
798 g_string_free (rtpinfo, TRUE);
802 str = gst_rtsp_media_get_range_string (media->media, TRUE);
803 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
805 send_response (client, session, state->response);
807 /* start playing after sending the request */
808 gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
810 media->state = GST_RTSP_STATE_PLAYING;
817 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
822 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
827 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
834 do_keepalive (GstRTSPSession * session)
836 GST_INFO ("keep session %p alive", session);
837 gst_rtsp_session_touch (session);
841 handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
847 gboolean have_transport;
848 GstRTSPTransport *ct, *st;
850 GstRTSPLowerTrans supported;
851 GstRTSPStatusCode code;
852 GstRTSPSession *session;
853 GstRTSPSessionStream *stream;
854 gchar *trans_str, *pos;
856 GstRTSPSessionMedia *media;
860 /* the uri contains the stream number we added in the SDP config, which is
861 * always /stream=%d so we need to strip that off
862 * parse the stream we need to configure, look for the stream in the abspath
863 * first and then in the query. */
864 if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
865 if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
869 /* we can mofify the parse uri in place */
872 pos += strlen ("/stream=");
873 if (sscanf (pos, "%u", &streamid) != 1)
876 /* parse the transport */
878 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
880 if (res != GST_RTSP_OK)
883 transports = g_strsplit (transport, ",", 0);
884 gst_rtsp_transport_new (&ct);
886 /* init transports */
887 have_transport = FALSE;
888 gst_rtsp_transport_init (ct);
890 /* our supported transports */
891 supported = GST_RTSP_LOWER_TRANS_UDP |
892 GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
894 /* loop through the transports, try to parse */
895 for (i = 0; transports[i]; i++) {
896 res = gst_rtsp_transport_parse (transports[i], ct);
897 if (res != GST_RTSP_OK) {
898 /* no valid transport, search some more */
899 GST_WARNING ("could not parse transport %s", transports[i]);
903 /* we have a transport, see if it's RTP/AVP */
904 if (ct->trans != GST_RTSP_TRANS_RTP || ct->profile != GST_RTSP_PROFILE_AVP) {
905 GST_WARNING ("invalid transport %s", transports[i]);
909 if (!(ct->lower_transport & supported)) {
910 GST_WARNING ("unsupported transport %s", transports[i]);
914 /* we have a valid transport */
915 GST_INFO ("found valid transport %s", transports[i]);
916 have_transport = TRUE;
920 gst_rtsp_transport_init (ct);
922 g_strfreev (transports);
924 /* we have not found anything usable, error out */
926 goto unsupported_transports;
928 if (client->session_pool == NULL)
931 session = state->session;
934 g_object_ref (session);
935 /* get a handle to the configuration of the media in the session, this can
936 * return NULL if this is a new url to manage in this session. */
937 media = gst_rtsp_session_get_media (session, uri);
939 /* create a session if this fails we probably reached our session limit or
941 if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
942 goto service_unavailable;
944 state->session = session;
946 /* we need a new media configuration in this session */
950 /* we have no media, find one and manage it */
954 /* get a handle to the configuration of the media in the session */
955 if ((m = find_media (client, state))) {
956 /* manage the media in our session now */
957 media = gst_rtsp_session_manage_media (session, uri, m);
961 /* if we stil have no media, error */
965 state->sessmedia = media;
967 /* we have a valid transport now, set the destination of the client. */
968 g_free (ct->destination);
969 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
970 ct->destination = gst_rtsp_media_get_multicast_group (media->media);
974 url = gst_rtsp_connection_get_url (client->connection);
975 ct->destination = g_strdup (url->host);
977 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
978 /* check if the client selected channels for TCP */
979 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
980 gst_rtsp_session_media_alloc_channels (media, &ct->interleaved);
985 /* get a handle to the stream in the media */
986 if (!(stream = gst_rtsp_session_media_get_stream (media, streamid)))
989 st = gst_rtsp_session_stream_set_transport (stream, ct);
991 /* configure keepalive for this transport */
992 gst_rtsp_session_stream_set_keepalive (stream,
993 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
995 /* serialize the server transport */
996 trans_str = gst_rtsp_transport_as_text (st);
997 gst_rtsp_transport_free (st);
999 /* construct the response now */
1000 code = GST_RTSP_STS_OK;
1001 gst_rtsp_message_init_response (state->response, code,
1002 gst_rtsp_status_as_text (code), state->request);
1004 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
1008 send_response (client, session, state->response);
1010 /* update the state */
1011 switch (media->state) {
1012 case GST_RTSP_STATE_PLAYING:
1013 case GST_RTSP_STATE_RECORDING:
1014 case GST_RTSP_STATE_READY:
1015 /* no state change */
1018 media->state = GST_RTSP_STATE_READY;
1021 g_object_unref (session);
1028 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1033 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1034 g_object_unref (session);
1039 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1040 g_object_unref (media);
1041 g_object_unref (session);
1046 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1049 unsupported_transports:
1051 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1052 gst_rtsp_transport_free (ct);
1057 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1060 service_unavailable:
1062 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1067 static GstSDPMessage *
1068 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1073 GstRTSPLowerTrans protocols;
1075 gst_sdp_message_new (&sdp);
1077 /* some standard things first */
1078 gst_sdp_message_set_version (sdp, "0");
1080 if (client->is_ipv6)
1085 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
1088 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
1089 gst_sdp_message_set_information (sdp, "rtsp-server");
1090 gst_sdp_message_add_time (sdp, "0", "0", NULL);
1091 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
1092 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
1093 gst_sdp_message_add_attribute (sdp, "control", "*");
1095 info.server_proto = proto;
1096 protocols = gst_rtsp_media_get_protocols (media);
1097 if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
1098 info.server_ip = gst_rtsp_media_get_multicast_group (media);
1100 info.server_ip = g_strdup (client->server_ip);
1102 /* create an SDP for the media object */
1103 if (!gst_rtsp_sdp_from_media (sdp, &info, media))
1106 g_free (info.server_ip);
1113 g_free (info.server_ip);
1114 gst_sdp_message_free (sdp);
1119 /* for the describe we must generate an SDP */
1121 handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
1126 gchar *str, *content_base;
1127 GstRTSPMedia *media;
1129 /* check what kind of format is accepted, we don't really do anything with it
1130 * and always return SDP for now. */
1135 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
1137 if (res == GST_RTSP_ENOTIMPL)
1140 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
1144 /* find the media object for the uri */
1145 if (!(media = find_media (client, state)))
1148 /* create an SDP for the media object on this client */
1149 if (!(sdp = create_sdp (client, media)))
1152 g_object_unref (media);
1154 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1155 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1157 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
1160 /* content base for some clients that might screw up creating the setup uri */
1161 str = gst_rtsp_url_get_request_uri (state->uri);
1162 str_len = strlen (str);
1164 /* check for trailing '/' and append one */
1165 if (str[str_len - 1] != '/') {
1166 content_base = g_malloc (str_len + 2);
1167 memcpy (content_base, str, str_len);
1168 content_base[str_len] = '/';
1169 content_base[str_len + 1] = '\0';
1175 GST_INFO ("adding content-base: %s", content_base);
1177 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
1179 g_free (content_base);
1181 /* add SDP to the response body */
1182 str = gst_sdp_message_as_text (sdp);
1183 gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
1184 gst_sdp_message_free (sdp);
1186 send_response (client, state->session, state->response);
1193 /* error reply is already sent */
1198 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1199 g_object_unref (media);
1205 handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
1207 GstRTSPMethod options;
1210 options = GST_RTSP_DESCRIBE |
1215 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
1217 str = gst_rtsp_options_as_text (options);
1219 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1220 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1222 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
1225 send_response (client, state->session, state->response);
1230 /* remove duplicate and trailing '/' */
1232 sanitize_uri (GstRTSPUrl * uri)
1236 gboolean have_slash, prev_slash;
1238 s = d = uri->abspath;
1239 len = strlen (uri->abspath);
1243 for (i = 0; i < len; i++) {
1244 have_slash = s[i] == '/';
1246 if (!have_slash || !prev_slash)
1248 prev_slash = have_slash;
1250 len = d - uri->abspath;
1251 /* don't remove the first slash if that's the only thing left */
1252 if (len > 1 && *(d - 1) == '/')
1258 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
1260 GST_INFO ("client %p: session %p finished", client, session);
1262 /* unlink all media managed in this session */
1263 client_unlink_session (client, session);
1265 /* remove the session */
1266 if (!(client->sessions = g_list_remove (client->sessions, session))) {
1267 GST_INFO ("client %p: all sessions finalized, close the connection",
1269 close_connection (client);
1274 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
1278 for (walk = client->sessions; walk; walk = g_list_next (walk)) {
1279 GstRTSPSession *msession = (GstRTSPSession *) walk->data;
1281 /* we already know about this session */
1282 if (msession == session)
1286 GST_INFO ("watching session %p", session);
1288 g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
1290 client->sessions = g_list_prepend (client->sessions, session);
1294 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
1296 GstRTSPMethod method;
1297 const gchar *uristr;
1299 GstRTSPVersion version;
1301 GstRTSPSession *session;
1302 GstRTSPClientState state = { NULL };
1303 GstRTSPMessage response = { 0 };
1306 state.request = request;
1307 state.response = &response;
1309 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
1310 gst_rtsp_message_dump (request);
1313 GST_INFO ("client %p: received a request", client);
1315 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
1317 if (version != GST_RTSP_VERSION_1_0) {
1318 /* we can only handle 1.0 requests */
1319 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
1323 state.method = method;
1325 /* we always try to parse the url first */
1326 if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
1327 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1331 /* sanitize the uri */
1335 /* get the session if there is any */
1336 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
1337 if (res == GST_RTSP_OK) {
1338 if (client->session_pool == NULL)
1341 /* we had a session in the request, find it again */
1342 if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
1343 goto session_not_found;
1345 /* we add the session to the client list of watched sessions. When a session
1346 * disappears because it times out, we will be notified. If all sessions are
1347 * gone, we will close the connection */
1348 client_watch_session (client, session);
1352 state.session = session;
1355 if (!gst_rtsp_auth_check (client->auth, client, &state))
1356 goto not_authorized;
1359 /* now see what is asked and dispatch to a dedicated handler */
1361 case GST_RTSP_OPTIONS:
1362 handle_options_request (client, &state);
1364 case GST_RTSP_DESCRIBE:
1365 handle_describe_request (client, &state);
1367 case GST_RTSP_SETUP:
1368 handle_setup_request (client, &state);
1371 handle_play_request (client, &state);
1373 case GST_RTSP_PAUSE:
1374 handle_pause_request (client, &state);
1376 case GST_RTSP_TEARDOWN:
1377 handle_teardown_request (client, &state);
1379 case GST_RTSP_SET_PARAMETER:
1380 handle_set_param_request (client, &state);
1382 case GST_RTSP_GET_PARAMETER:
1383 handle_get_param_request (client, &state);
1385 case GST_RTSP_ANNOUNCE:
1386 case GST_RTSP_RECORD:
1387 case GST_RTSP_REDIRECT:
1388 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
1390 case GST_RTSP_INVALID:
1392 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1396 g_object_unref (session);
1398 gst_rtsp_url_free (uri);
1404 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
1409 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
1414 handle_unauthorized_request (client, client->auth, &state);
1420 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
1430 /* find the stream for this message */
1431 res = gst_rtsp_message_parse_data (message, &channel);
1432 if (res != GST_RTSP_OK)
1435 gst_rtsp_message_steal_body (message, &data, &size);
1437 buffer = gst_buffer_new ();
1438 GST_BUFFER_DATA (buffer) = data;
1439 GST_BUFFER_MALLOCDATA (buffer) = data;
1440 GST_BUFFER_SIZE (buffer) = size;
1443 for (walk = client->streams; walk; walk = g_list_next (walk)) {
1444 GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
1445 GstRTSPMediaStream *mstream;
1446 GstRTSPTransport *tr;
1448 /* get the transport, if there is no transport configured, skip this stream */
1449 if (!(tr = stream->trans.transport))
1452 /* we also need a media stream */
1453 if (!(mstream = stream->media_stream))
1456 /* check for TCP transport */
1457 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1458 /* dispatch to the stream based on the channel number */
1459 if (tr->interleaved.min == channel) {
1460 gst_rtsp_media_stream_rtp (mstream, buffer);
1463 } else if (tr->interleaved.max == channel) {
1464 gst_rtsp_media_stream_rtcp (mstream, buffer);
1471 gst_buffer_unref (buffer);
1475 * gst_rtsp_client_set_session_pool:
1476 * @client: a #GstRTSPClient
1477 * @pool: a #GstRTSPSessionPool
1479 * Set @pool as the sessionpool for @client which it will use to find
1480 * or allocate sessions. the sessionpool is usually inherited from the server
1481 * that created the client but can be overridden later.
1484 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
1485 GstRTSPSessionPool * pool)
1487 GstRTSPSessionPool *old;
1489 old = client->session_pool;
1492 g_object_ref (pool);
1493 client->session_pool = pool;
1495 g_object_unref (old);
1500 * gst_rtsp_client_get_session_pool:
1501 * @client: a #GstRTSPClient
1503 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
1505 * Returns: a #GstRTSPSessionPool, unref after usage.
1507 GstRTSPSessionPool *
1508 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
1510 GstRTSPSessionPool *result;
1512 if ((result = client->session_pool))
1513 g_object_ref (result);
1519 * gst_rtsp_client_set_server:
1520 * @client: a #GstRTSPClient
1521 * @server: a #GstRTSPServer
1523 * Set @server as the server that created @client.
1526 gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
1530 old = client->server;
1531 if (old != server) {
1533 g_object_ref (server);
1534 client->server = server;
1536 g_object_unref (old);
1541 * gst_rtsp_client_get_server:
1542 * @client: a #GstRTSPClient
1544 * Get the #GstRTSPServer object that @client was created from.
1546 * Returns: a #GstRTSPServer, unref after usage.
1549 gst_rtsp_client_get_server (GstRTSPClient * client)
1551 GstRTSPServer *result;
1553 if ((result = client->server))
1554 g_object_ref (result);
1560 * gst_rtsp_client_set_media_mapping:
1561 * @client: a #GstRTSPClient
1562 * @mapping: a #GstRTSPMediaMapping
1564 * Set @mapping as the media mapping for @client which it will use to map urls
1565 * to media streams. These mapping is usually inherited from the server that
1566 * created the client but can be overriden later.
1569 gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
1570 GstRTSPMediaMapping * mapping)
1572 GstRTSPMediaMapping *old;
1574 old = client->media_mapping;
1576 if (old != mapping) {
1578 g_object_ref (mapping);
1579 client->media_mapping = mapping;
1581 g_object_unref (old);
1586 * gst_rtsp_client_get_media_mapping:
1587 * @client: a #GstRTSPClient
1589 * Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
1591 * Returns: a #GstRTSPMediaMapping, unref after usage.
1593 GstRTSPMediaMapping *
1594 gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
1596 GstRTSPMediaMapping *result;
1598 if ((result = client->media_mapping))
1599 g_object_ref (result);
1605 * gst_rtsp_client_set_auth:
1606 * @client: a #GstRTSPClient
1607 * @auth: a #GstRTSPAuth
1609 * configure @auth to be used as the authentication manager of @client.
1612 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
1616 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1622 g_object_ref (auth);
1623 client->auth = auth;
1625 g_object_unref (old);
1631 * gst_rtsp_client_get_auth:
1632 * @client: a #GstRTSPClient
1634 * Get the #GstRTSPAuth used as the authentication manager of @client.
1636 * Returns: the #GstRTSPAuth of @client. g_object_unref() after
1640 gst_rtsp_client_get_auth (GstRTSPClient * client)
1642 GstRTSPAuth *result;
1644 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
1646 if ((result = client->auth))
1647 g_object_ref (result);
1652 static GstRTSPResult
1653 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
1656 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1658 switch (message->type) {
1659 case GST_RTSP_MESSAGE_REQUEST:
1660 handle_request (client, message);
1662 case GST_RTSP_MESSAGE_RESPONSE:
1664 case GST_RTSP_MESSAGE_DATA:
1665 handle_data (client, message);
1673 static GstRTSPResult
1674 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
1676 /* GstRTSPClient *client; */
1678 /* client = GST_RTSP_CLIENT (user_data); */
1680 /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
1685 static GstRTSPResult
1686 closed (GstRTSPWatch * watch, gpointer user_data)
1688 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1689 const gchar *tunnelid;
1691 GST_INFO ("client %p: connection closed", client);
1693 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
1694 g_mutex_lock (tunnels_lock);
1695 /* remove from tunnelids */
1696 g_hash_table_remove (tunnels, tunnelid);
1697 g_mutex_unlock (tunnels_lock);
1703 static GstRTSPResult
1704 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
1706 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1709 str = gst_rtsp_strresult (result);
1710 GST_INFO ("client %p: received an error %s", client, str);
1716 static GstRTSPResult
1717 error_full (GstRTSPWatch * watch, GstRTSPResult result,
1718 GstRTSPMessage * message, guint id, gpointer user_data)
1720 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1723 str = gst_rtsp_strresult (result);
1725 ("client %p: received an error %s when handling message %p with id %d",
1726 client, str, message, id);
1733 remember_tunnel (GstRTSPClient * client)
1735 const gchar *tunnelid;
1737 /* store client in the pending tunnels */
1738 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
1739 if (tunnelid == NULL)
1742 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
1744 /* we can't have two clients connecting with the same tunnelid */
1745 g_mutex_lock (tunnels_lock);
1746 if (g_hash_table_lookup (tunnels, tunnelid))
1747 goto tunnel_existed;
1749 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
1750 g_mutex_unlock (tunnels_lock);
1757 GST_ERROR ("client %p: no tunnelid provided", client);
1762 g_mutex_unlock (tunnels_lock);
1763 GST_ERROR ("client %p: tunnel session %s already existed", client,
1769 static GstRTSPStatusCode
1770 tunnel_start (GstRTSPWatch * watch, gpointer user_data)
1772 GstRTSPClient *client;
1774 client = GST_RTSP_CLIENT (user_data);
1776 GST_INFO ("client %p: tunnel start (connection %p)", client,
1777 client->connection);
1779 if (!remember_tunnel (client))
1782 return GST_RTSP_STS_OK;
1787 GST_ERROR ("client %p: error starting tunnel", client);
1788 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1792 static GstRTSPResult
1793 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
1795 GstRTSPClient *client;
1797 client = GST_RTSP_CLIENT (user_data);
1799 GST_INFO ("client %p: tunnel lost (connection %p)", client,
1800 client->connection);
1802 /* ignore error, it'll only be a problem when the client does a POST again */
1803 remember_tunnel (client);
1808 static GstRTSPResult
1809 tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
1811 const gchar *tunnelid;
1812 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1813 GstRTSPClient *oclient;
1815 GST_INFO ("client %p: tunnel complete", client);
1817 /* find previous tunnel */
1818 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
1819 if (tunnelid == NULL)
1822 g_mutex_lock (tunnels_lock);
1823 if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
1826 /* remove the old client from the table. ref before because removing it will
1827 * remove the ref to it. */
1828 g_object_ref (oclient);
1829 g_hash_table_remove (tunnels, tunnelid);
1831 if (oclient->watch == NULL)
1833 g_mutex_unlock (tunnels_lock);
1835 GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
1836 oclient->connection, client->connection);
1838 /* merge the tunnels into the first client */
1839 gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
1840 gst_rtsp_watch_reset (oclient->watch);
1841 g_object_unref (oclient);
1843 /* we don't need this watch anymore */
1844 g_source_destroy ((GSource *) client->watch);
1845 client->watchid = 0;
1846 client->watch = NULL;
1853 GST_INFO ("client %p: no tunnelid provided", client);
1854 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1858 g_mutex_unlock (tunnels_lock);
1859 GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
1860 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1864 g_mutex_unlock (tunnels_lock);
1865 GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
1866 g_object_unref (oclient);
1867 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1871 static GstRTSPWatchFuncs watch_funcs = {
1883 client_watch_notify (GstRTSPClient * client)
1885 GST_INFO ("client %p: watch destroyed", client);
1886 client->watchid = 0;
1887 client->watch = NULL;
1888 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
1889 g_object_unref (client);
1893 * gst_rtsp_client_attach:
1894 * @client: a #GstRTSPClient
1895 * @channel: a #GIOChannel
1897 * Accept a new connection for @client on the socket in @channel.
1899 * This function should be called when the client properties and urls are fully
1900 * configured and the client is ready to start.
1902 * Returns: %TRUE if the client could be accepted.
1905 gst_rtsp_client_accept (GstRTSPClient * client, GIOChannel * channel)
1908 GstRTSPConnection *conn;
1911 GMainContext *context;
1913 struct sockaddr_storage addr;
1915 gchar ip[INET6_ADDRSTRLEN];
1917 /* a new client connected. */
1918 sock = g_io_channel_unix_get_fd (channel);
1920 GST_RTSP_CHECK (gst_rtsp_connection_accept (sock, &conn), accept_failed);
1922 fd = gst_rtsp_connection_get_readfd (conn);
1924 addrlen = sizeof (addr);
1925 if (getsockname (fd, (struct sockaddr *) &addr, &addrlen) < 0)
1926 goto getpeername_failed;
1928 client->is_ipv6 = addr.ss_family == AF_INET6;
1930 if (getnameinfo ((struct sockaddr *) &addr, addrlen, ip, sizeof (ip), NULL, 0,
1931 NI_NUMERICHOST) != 0)
1932 goto getnameinfo_failed;
1934 /* keep the original ip that the client connected to */
1935 g_free (client->server_ip);
1936 client->server_ip = g_strndup (ip, sizeof (ip));
1938 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
1939 client->server_ip, client->is_ipv6);
1941 url = gst_rtsp_connection_get_url (conn);
1942 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
1944 client->connection = conn;
1946 /* create watch for the connection and attach */
1947 client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
1948 g_object_ref (client), (GDestroyNotify) client_watch_notify);
1950 /* find the context to add the watch */
1951 if ((source = g_main_current_source ()))
1952 context = g_source_get_context (source);
1956 GST_INFO ("attaching to context %p", context);
1958 client->watchid = gst_rtsp_watch_attach (client->watch, context);
1959 gst_rtsp_watch_unref (client->watch);
1966 gchar *str = gst_rtsp_strresult (res);
1968 GST_ERROR ("Could not accept client on server socket %d: %s", sock, str);
1974 GST_ERROR ("getpeername failed: %s", g_strerror (errno));
1979 GST_ERROR ("getnameinfo failed: %s", g_strerror (errno));