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 };
407 gst_rtsp_message_init_data (&message, channel);
409 if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
412 gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
414 /* FIXME, client->watch could have been finalized here, we need to keep an
415 * extra refcount to the watch. */
416 gst_rtsp_watch_send_message (client->watch, &message, NULL);
418 gst_rtsp_message_steal_body (&message, &data, &usize);
419 gst_buffer_unmap (buffer, &map_info);
421 gst_rtsp_message_unset (&message);
427 do_send_data_list (GstBufferList * blist, guint8 channel,
428 GstRTSPClient * client)
432 len = gst_buffer_list_len (blist);
434 for (i = 0; i < len; i++) {
435 GstBuffer *group = gst_buffer_list_get (blist, i);
437 do_send_data (group, channel, client);
444 link_stream (GstRTSPClient * client, GstRTSPSession * session,
445 GstRTSPSessionStream * stream)
447 GST_DEBUG ("client %p: linking stream %p", client, stream);
448 gst_rtsp_session_stream_set_callbacks (stream, (GstRTSPSendFunc) do_send_data,
449 (GstRTSPSendFunc) do_send_data, 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);
461 client->streams = g_list_remove (client->streams, stream);
462 /* our session can now expire */
463 gst_rtsp_session_allow_expire (session);
467 unlink_session_streams (GstRTSPClient * client, GstRTSPSession * session,
468 GstRTSPSessionMedia * media)
472 n_streams = gst_rtsp_media_n_streams (media->media);
473 for (i = 0; i < n_streams; i++) {
474 GstRTSPSessionStream *sstream;
475 GstRTSPTransport *tr;
477 /* get the stream as configured in the session */
478 sstream = gst_rtsp_session_media_get_stream (media, i);
479 /* get the transport, if there is no transport configured, skip this stream */
480 if (!(tr = sstream->trans.transport))
483 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
484 /* for TCP, unlink the stream from the TCP connection of the client */
485 unlink_stream (client, session, sstream);
491 close_connection (GstRTSPClient * client)
493 const gchar *tunnelid;
495 GST_DEBUG ("client %p: closing connection", client);
497 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
498 g_mutex_lock (tunnels_lock);
499 /* remove from tunnelids */
500 g_hash_table_remove (tunnels, tunnelid);
501 g_mutex_unlock (tunnels_lock);
504 gst_rtsp_connection_close (client->connection);
505 if (client->watchid) {
506 g_source_destroy ((GSource *) client->watch);
508 client->watch = NULL;
513 handle_teardown_request (GstRTSPClient * client, GstRTSPClientState * state)
515 GstRTSPSession *session;
516 GstRTSPSessionMedia *media;
517 GstRTSPStatusCode code;
522 session = state->session;
524 /* get a handle to the configuration of the media in the session */
525 media = gst_rtsp_session_get_media (session, state->uri);
529 state->sessmedia = media;
531 /* unlink the all TCP callbacks */
532 unlink_session_streams (client, session, media);
534 /* remove the session from the watched sessions */
535 g_object_weak_unref (G_OBJECT (session),
536 (GWeakNotify) client_session_finalized, client);
537 client->sessions = g_list_remove (client->sessions, session);
539 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
541 /* unmanage the media in the session, returns false if all media session
543 if (!gst_rtsp_session_release_media (session, media)) {
544 /* remove the session */
545 gst_rtsp_session_pool_remove (client->session_pool, session);
547 /* construct the response now */
548 code = GST_RTSP_STS_OK;
549 gst_rtsp_message_init_response (state->response, code,
550 gst_rtsp_status_as_text (code), state->request);
552 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONNECTION,
555 send_response (client, session, state->response);
557 close_connection (client);
564 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
569 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
575 handle_get_param_request (GstRTSPClient * client, GstRTSPClientState * state)
581 res = gst_rtsp_message_get_body (state->request, &data, &size);
582 if (res != GST_RTSP_OK)
586 /* no body, keep-alive request */
587 send_generic_response (client, GST_RTSP_STS_OK, state);
589 /* there is a body, handle the params */
590 res = gst_rtsp_params_get (client, state);
591 if (res != GST_RTSP_OK)
594 send_response (client, state->session, state->response);
601 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
607 handle_set_param_request (GstRTSPClient * client, GstRTSPClientState * state)
613 res = gst_rtsp_message_get_body (state->request, &data, &size);
614 if (res != GST_RTSP_OK)
618 /* no body, keep-alive request */
619 send_generic_response (client, GST_RTSP_STS_OK, state);
621 /* there is a body, handle the params */
622 res = gst_rtsp_params_set (client, state);
623 if (res != GST_RTSP_OK)
626 send_response (client, state->session, state->response);
633 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
639 handle_pause_request (GstRTSPClient * client, GstRTSPClientState * state)
641 GstRTSPSession *session;
642 GstRTSPSessionMedia *media;
643 GstRTSPStatusCode code;
645 if (!(session = state->session))
648 /* get a handle to the configuration of the media in the session */
649 media = gst_rtsp_session_get_media (session, state->uri);
653 state->sessmedia = media;
655 /* the session state must be playing or recording */
656 if (media->state != GST_RTSP_STATE_PLAYING &&
657 media->state != GST_RTSP_STATE_RECORDING)
660 /* unlink the all TCP callbacks */
661 unlink_session_streams (client, session, media);
663 /* then pause sending */
664 gst_rtsp_session_media_set_state (media, GST_STATE_PAUSED);
666 /* construct the response now */
667 code = GST_RTSP_STS_OK;
668 gst_rtsp_message_init_response (state->response, code,
669 gst_rtsp_status_as_text (code), state->request);
671 send_response (client, session, state->response);
673 /* the state is now READY */
674 media->state = GST_RTSP_STATE_READY;
681 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
686 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
691 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
698 handle_play_request (GstRTSPClient * client, GstRTSPClientState * state)
700 GstRTSPSession *session;
701 GstRTSPSessionMedia *media;
702 GstRTSPStatusCode code;
704 guint n_streams, i, infocount;
705 guint timestamp, seqnum;
707 GstRTSPTimeRange *range;
710 if (!(session = state->session))
713 /* get a handle to the configuration of the media in the session */
714 media = gst_rtsp_session_get_media (session, state->uri);
718 state->sessmedia = media;
720 /* the session state must be playing or ready */
721 if (media->state != GST_RTSP_STATE_PLAYING &&
722 media->state != GST_RTSP_STATE_READY)
725 /* parse the range header if we have one */
727 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_RANGE, &str, 0);
728 if (res == GST_RTSP_OK) {
729 if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
730 /* we have a range, seek to the position */
731 gst_rtsp_media_seek (media->media, range);
732 gst_rtsp_range_free (range);
736 /* grab RTPInfo from the payloaders now */
737 rtpinfo = g_string_new ("");
739 n_streams = gst_rtsp_media_n_streams (media->media);
740 for (i = 0, infocount = 0; i < n_streams; i++) {
741 GstRTSPSessionStream *sstream;
742 GstRTSPMediaStream *stream;
743 GstRTSPTransport *tr;
744 GObjectClass *payobjclass;
747 /* get the stream as configured in the session */
748 sstream = gst_rtsp_session_media_get_stream (media, i);
749 /* get the transport, if there is no transport configured, skip this stream */
750 if (!(tr = sstream->trans.transport)) {
751 GST_INFO ("stream %d is not configured", i);
755 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
756 /* for TCP, link the stream to the TCP connection of the client */
757 link_stream (client, session, sstream);
760 stream = sstream->media_stream;
762 payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
764 if (g_object_class_find_property (payobjclass, "seqnum") &&
765 g_object_class_find_property (payobjclass, "timestamp")) {
768 payobj = G_OBJECT (stream->payloader);
770 /* only add RTP-Info for streams with seqnum and timestamp */
771 g_object_get (payobj, "seqnum", &seqnum, "timestamp", ×tamp, NULL);
774 g_string_append (rtpinfo, ", ");
776 uristr = gst_rtsp_url_get_request_uri (state->uri);
777 g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
778 uristr, i, seqnum, timestamp);
783 GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
787 /* construct the response now */
788 code = GST_RTSP_STS_OK;
789 gst_rtsp_message_init_response (state->response, code,
790 gst_rtsp_status_as_text (code), state->request);
792 /* add the RTP-Info header */
794 str = g_string_free (rtpinfo, FALSE);
795 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RTP_INFO, str);
797 g_string_free (rtpinfo, TRUE);
801 str = gst_rtsp_media_get_range_string (media->media, TRUE);
802 gst_rtsp_message_take_header (state->response, GST_RTSP_HDR_RANGE, str);
804 send_response (client, session, state->response);
806 /* start playing after sending the request */
807 gst_rtsp_session_media_set_state (media, GST_STATE_PLAYING);
809 media->state = GST_RTSP_STATE_PLAYING;
816 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, state);
821 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
826 send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
833 do_keepalive (GstRTSPSession * session)
835 GST_INFO ("keep session %p alive", session);
836 gst_rtsp_session_touch (session);
840 handle_setup_request (GstRTSPClient * client, GstRTSPClientState * state)
846 gboolean have_transport;
847 GstRTSPTransport *ct, *st;
849 GstRTSPLowerTrans supported;
850 GstRTSPStatusCode code;
851 GstRTSPSession *session;
852 GstRTSPSessionStream *stream;
853 gchar *trans_str, *pos;
855 GstRTSPSessionMedia *media;
859 /* the uri contains the stream number we added in the SDP config, which is
860 * always /stream=%d so we need to strip that off
861 * parse the stream we need to configure, look for the stream in the abspath
862 * first and then in the query. */
863 if (uri->abspath == NULL || !(pos = strstr (uri->abspath, "/stream="))) {
864 if (uri->query == NULL || !(pos = strstr (uri->query, "/stream=")))
868 /* we can mofify the parse uri in place */
871 pos += strlen ("/stream=");
872 if (sscanf (pos, "%u", &streamid) != 1)
875 /* parse the transport */
877 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_TRANSPORT,
879 if (res != GST_RTSP_OK)
882 transports = g_strsplit (transport, ",", 0);
883 gst_rtsp_transport_new (&ct);
885 /* init transports */
886 have_transport = FALSE;
887 gst_rtsp_transport_init (ct);
889 /* our supported transports */
890 supported = GST_RTSP_LOWER_TRANS_UDP |
891 GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
893 /* loop through the transports, try to parse */
894 for (i = 0; transports[i]; i++) {
895 res = gst_rtsp_transport_parse (transports[i], ct);
896 if (res != GST_RTSP_OK) {
897 /* no valid transport, search some more */
898 GST_WARNING ("could not parse transport %s", transports[i]);
902 /* we have a transport, see if it's RTP/AVP */
903 if (ct->trans != GST_RTSP_TRANS_RTP || ct->profile != GST_RTSP_PROFILE_AVP) {
904 GST_WARNING ("invalid transport %s", transports[i]);
908 if (!(ct->lower_transport & supported)) {
909 GST_WARNING ("unsupported transport %s", transports[i]);
913 /* we have a valid transport */
914 GST_INFO ("found valid transport %s", transports[i]);
915 have_transport = TRUE;
919 gst_rtsp_transport_init (ct);
921 g_strfreev (transports);
923 /* we have not found anything usable, error out */
925 goto unsupported_transports;
927 if (client->session_pool == NULL)
930 session = state->session;
933 g_object_ref (session);
934 /* get a handle to the configuration of the media in the session, this can
935 * return NULL if this is a new url to manage in this session. */
936 media = gst_rtsp_session_get_media (session, uri);
938 /* create a session if this fails we probably reached our session limit or
940 if (!(session = gst_rtsp_session_pool_create (client->session_pool)))
941 goto service_unavailable;
943 state->session = session;
945 /* we need a new media configuration in this session */
949 /* we have no media, find one and manage it */
953 /* get a handle to the configuration of the media in the session */
954 if ((m = find_media (client, state))) {
955 /* manage the media in our session now */
956 media = gst_rtsp_session_manage_media (session, uri, m);
960 /* if we stil have no media, error */
964 state->sessmedia = media;
966 /* we have a valid transport now, set the destination of the client. */
967 g_free (ct->destination);
968 if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
969 ct->destination = gst_rtsp_media_get_multicast_group (media->media);
973 url = gst_rtsp_connection_get_url (client->connection);
974 ct->destination = g_strdup (url->host);
976 if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
977 /* check if the client selected channels for TCP */
978 if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
979 gst_rtsp_session_media_alloc_channels (media, &ct->interleaved);
984 /* get a handle to the stream in the media */
985 if (!(stream = gst_rtsp_session_media_get_stream (media, streamid)))
988 st = gst_rtsp_session_stream_set_transport (stream, ct);
990 /* configure keepalive for this transport */
991 gst_rtsp_session_stream_set_keepalive (stream,
992 (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
994 /* serialize the server transport */
995 trans_str = gst_rtsp_transport_as_text (st);
996 gst_rtsp_transport_free (st);
998 /* construct the response now */
999 code = GST_RTSP_STS_OK;
1000 gst_rtsp_message_init_response (state->response, code,
1001 gst_rtsp_status_as_text (code), state->request);
1003 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_TRANSPORT,
1007 send_response (client, session, state->response);
1009 /* update the state */
1010 switch (media->state) {
1011 case GST_RTSP_STATE_PLAYING:
1012 case GST_RTSP_STATE_RECORDING:
1013 case GST_RTSP_STATE_READY:
1014 /* no state change */
1017 media->state = GST_RTSP_STATE_READY;
1020 g_object_unref (session);
1027 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, state);
1032 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1033 g_object_unref (session);
1038 send_generic_response (client, GST_RTSP_STS_NOT_FOUND, state);
1039 g_object_unref (media);
1040 g_object_unref (session);
1045 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1048 unsupported_transports:
1050 send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, state);
1051 gst_rtsp_transport_free (ct);
1056 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1059 service_unavailable:
1061 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1066 static GstSDPMessage *
1067 create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
1072 GstRTSPLowerTrans protocols;
1074 gst_sdp_message_new (&sdp);
1076 /* some standard things first */
1077 gst_sdp_message_set_version (sdp, "0");
1079 if (client->is_ipv6)
1084 gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
1087 gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
1088 gst_sdp_message_set_information (sdp, "rtsp-server");
1089 gst_sdp_message_add_time (sdp, "0", "0", NULL);
1090 gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
1091 gst_sdp_message_add_attribute (sdp, "type", "broadcast");
1092 gst_sdp_message_add_attribute (sdp, "control", "*");
1094 info.server_proto = proto;
1095 protocols = gst_rtsp_media_get_protocols (media);
1096 if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST)
1097 info.server_ip = gst_rtsp_media_get_multicast_group (media);
1099 info.server_ip = g_strdup (client->server_ip);
1101 /* create an SDP for the media object */
1102 if (!gst_rtsp_sdp_from_media (sdp, &info, media))
1105 g_free (info.server_ip);
1112 g_free (info.server_ip);
1113 gst_sdp_message_free (sdp);
1118 /* for the describe we must generate an SDP */
1120 handle_describe_request (GstRTSPClient * client, GstRTSPClientState * state)
1125 gchar *str, *content_base;
1126 GstRTSPMedia *media;
1128 /* check what kind of format is accepted, we don't really do anything with it
1129 * and always return SDP for now. */
1134 gst_rtsp_message_get_header (state->request, GST_RTSP_HDR_ACCEPT,
1136 if (res == GST_RTSP_ENOTIMPL)
1139 if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
1143 /* find the media object for the uri */
1144 if (!(media = find_media (client, state)))
1147 /* create an SDP for the media object on this client */
1148 if (!(sdp = create_sdp (client, media)))
1151 g_object_unref (media);
1153 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1154 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1156 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_TYPE,
1159 /* content base for some clients that might screw up creating the setup uri */
1160 str = gst_rtsp_url_get_request_uri (state->uri);
1161 str_len = strlen (str);
1163 /* check for trailing '/' and append one */
1164 if (str[str_len - 1] != '/') {
1165 content_base = g_malloc (str_len + 2);
1166 memcpy (content_base, str, str_len);
1167 content_base[str_len] = '/';
1168 content_base[str_len + 1] = '\0';
1174 GST_INFO ("adding content-base: %s", content_base);
1176 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_CONTENT_BASE,
1178 g_free (content_base);
1180 /* add SDP to the response body */
1181 str = gst_sdp_message_as_text (sdp);
1182 gst_rtsp_message_take_body (state->response, (guint8 *) str, strlen (str));
1183 gst_sdp_message_free (sdp);
1185 send_response (client, state->session, state->response);
1192 /* error reply is already sent */
1197 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, state);
1198 g_object_unref (media);
1204 handle_options_request (GstRTSPClient * client, GstRTSPClientState * state)
1206 GstRTSPMethod options;
1209 options = GST_RTSP_DESCRIBE |
1214 GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
1216 str = gst_rtsp_options_as_text (options);
1218 gst_rtsp_message_init_response (state->response, GST_RTSP_STS_OK,
1219 gst_rtsp_status_as_text (GST_RTSP_STS_OK), state->request);
1221 gst_rtsp_message_add_header (state->response, GST_RTSP_HDR_PUBLIC, str);
1224 send_response (client, state->session, state->response);
1229 /* remove duplicate and trailing '/' */
1231 sanitize_uri (GstRTSPUrl * uri)
1235 gboolean have_slash, prev_slash;
1237 s = d = uri->abspath;
1238 len = strlen (uri->abspath);
1242 for (i = 0; i < len; i++) {
1243 have_slash = s[i] == '/';
1245 if (!have_slash || !prev_slash)
1247 prev_slash = have_slash;
1249 len = d - uri->abspath;
1250 /* don't remove the first slash if that's the only thing left */
1251 if (len > 1 && *(d - 1) == '/')
1257 client_session_finalized (GstRTSPClient * client, GstRTSPSession * session)
1259 GST_INFO ("client %p: session %p finished", client, session);
1261 /* unlink all media managed in this session */
1262 client_unlink_session (client, session);
1264 /* remove the session */
1265 if (!(client->sessions = g_list_remove (client->sessions, session))) {
1266 GST_INFO ("client %p: all sessions finalized, close the connection",
1268 close_connection (client);
1273 client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
1277 for (walk = client->sessions; walk; walk = g_list_next (walk)) {
1278 GstRTSPSession *msession = (GstRTSPSession *) walk->data;
1280 /* we already know about this session */
1281 if (msession == session)
1285 GST_INFO ("watching session %p", session);
1287 g_object_weak_ref (G_OBJECT (session), (GWeakNotify) client_session_finalized,
1289 client->sessions = g_list_prepend (client->sessions, session);
1293 handle_request (GstRTSPClient * client, GstRTSPMessage * request)
1295 GstRTSPMethod method;
1296 const gchar *uristr;
1298 GstRTSPVersion version;
1300 GstRTSPSession *session;
1301 GstRTSPClientState state = { NULL };
1302 GstRTSPMessage response = { 0 };
1305 state.request = request;
1306 state.response = &response;
1308 if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
1309 gst_rtsp_message_dump (request);
1312 GST_INFO ("client %p: received a request", client);
1314 gst_rtsp_message_parse_request (request, &method, &uristr, &version);
1316 if (version != GST_RTSP_VERSION_1_0) {
1317 /* we can only handle 1.0 requests */
1318 send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
1322 state.method = method;
1324 /* we always try to parse the url first */
1325 if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
1326 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1330 /* sanitize the uri */
1334 /* get the session if there is any */
1335 res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
1336 if (res == GST_RTSP_OK) {
1337 if (client->session_pool == NULL)
1340 /* we had a session in the request, find it again */
1341 if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
1342 goto session_not_found;
1344 /* we add the session to the client list of watched sessions. When a session
1345 * disappears because it times out, we will be notified. If all sessions are
1346 * gone, we will close the connection */
1347 client_watch_session (client, session);
1351 state.session = session;
1354 if (!gst_rtsp_auth_check (client->auth, client, &state))
1355 goto not_authorized;
1358 /* now see what is asked and dispatch to a dedicated handler */
1360 case GST_RTSP_OPTIONS:
1361 handle_options_request (client, &state);
1363 case GST_RTSP_DESCRIBE:
1364 handle_describe_request (client, &state);
1366 case GST_RTSP_SETUP:
1367 handle_setup_request (client, &state);
1370 handle_play_request (client, &state);
1372 case GST_RTSP_PAUSE:
1373 handle_pause_request (client, &state);
1375 case GST_RTSP_TEARDOWN:
1376 handle_teardown_request (client, &state);
1378 case GST_RTSP_SET_PARAMETER:
1379 handle_set_param_request (client, &state);
1381 case GST_RTSP_GET_PARAMETER:
1382 handle_get_param_request (client, &state);
1384 case GST_RTSP_ANNOUNCE:
1385 case GST_RTSP_RECORD:
1386 case GST_RTSP_REDIRECT:
1387 send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &state);
1389 case GST_RTSP_INVALID:
1391 send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &state);
1395 g_object_unref (session);
1397 gst_rtsp_url_free (uri);
1403 send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, &state);
1408 send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, &state);
1413 handle_unauthorized_request (client, client->auth, &state);
1419 handle_data (GstRTSPClient * client, GstRTSPMessage * message)
1429 /* find the stream for this message */
1430 res = gst_rtsp_message_parse_data (message, &channel);
1431 if (res != GST_RTSP_OK)
1434 gst_rtsp_message_steal_body (message, &data, &size);
1436 buffer = gst_buffer_new ();
1437 gst_buffer_take_memory (buffer, -1,
1438 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
1441 for (walk = client->streams; walk; walk = g_list_next (walk)) {
1442 GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
1443 GstRTSPMediaStream *mstream;
1444 GstRTSPTransport *tr;
1446 /* get the transport, if there is no transport configured, skip this stream */
1447 if (!(tr = stream->trans.transport))
1450 /* we also need a media stream */
1451 if (!(mstream = stream->media_stream))
1454 /* check for TCP transport */
1455 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
1456 /* dispatch to the stream based on the channel number */
1457 if (tr->interleaved.min == channel) {
1458 gst_rtsp_media_stream_rtp (mstream, buffer);
1461 } else if (tr->interleaved.max == channel) {
1462 gst_rtsp_media_stream_rtcp (mstream, buffer);
1469 gst_buffer_unref (buffer);
1473 * gst_rtsp_client_set_session_pool:
1474 * @client: a #GstRTSPClient
1475 * @pool: a #GstRTSPSessionPool
1477 * Set @pool as the sessionpool for @client which it will use to find
1478 * or allocate sessions. the sessionpool is usually inherited from the server
1479 * that created the client but can be overridden later.
1482 gst_rtsp_client_set_session_pool (GstRTSPClient * client,
1483 GstRTSPSessionPool * pool)
1485 GstRTSPSessionPool *old;
1487 old = client->session_pool;
1490 g_object_ref (pool);
1491 client->session_pool = pool;
1493 g_object_unref (old);
1498 * gst_rtsp_client_get_session_pool:
1499 * @client: a #GstRTSPClient
1501 * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
1503 * Returns: a #GstRTSPSessionPool, unref after usage.
1505 GstRTSPSessionPool *
1506 gst_rtsp_client_get_session_pool (GstRTSPClient * client)
1508 GstRTSPSessionPool *result;
1510 if ((result = client->session_pool))
1511 g_object_ref (result);
1517 * gst_rtsp_client_set_server:
1518 * @client: a #GstRTSPClient
1519 * @server: a #GstRTSPServer
1521 * Set @server as the server that created @client.
1524 gst_rtsp_client_set_server (GstRTSPClient * client, GstRTSPServer * server)
1528 old = client->server;
1529 if (old != server) {
1531 g_object_ref (server);
1532 client->server = server;
1534 g_object_unref (old);
1539 * gst_rtsp_client_get_server:
1540 * @client: a #GstRTSPClient
1542 * Get the #GstRTSPServer object that @client was created from.
1544 * Returns: a #GstRTSPServer, unref after usage.
1547 gst_rtsp_client_get_server (GstRTSPClient * client)
1549 GstRTSPServer *result;
1551 if ((result = client->server))
1552 g_object_ref (result);
1558 * gst_rtsp_client_set_media_mapping:
1559 * @client: a #GstRTSPClient
1560 * @mapping: a #GstRTSPMediaMapping
1562 * Set @mapping as the media mapping for @client which it will use to map urls
1563 * to media streams. These mapping is usually inherited from the server that
1564 * created the client but can be overriden later.
1567 gst_rtsp_client_set_media_mapping (GstRTSPClient * client,
1568 GstRTSPMediaMapping * mapping)
1570 GstRTSPMediaMapping *old;
1572 old = client->media_mapping;
1574 if (old != mapping) {
1576 g_object_ref (mapping);
1577 client->media_mapping = mapping;
1579 g_object_unref (old);
1584 * gst_rtsp_client_get_media_mapping:
1585 * @client: a #GstRTSPClient
1587 * Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
1589 * Returns: a #GstRTSPMediaMapping, unref after usage.
1591 GstRTSPMediaMapping *
1592 gst_rtsp_client_get_media_mapping (GstRTSPClient * client)
1594 GstRTSPMediaMapping *result;
1596 if ((result = client->media_mapping))
1597 g_object_ref (result);
1603 * gst_rtsp_client_set_auth:
1604 * @client: a #GstRTSPClient
1605 * @auth: a #GstRTSPAuth
1607 * configure @auth to be used as the authentication manager of @client.
1610 gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
1614 g_return_if_fail (GST_IS_RTSP_CLIENT (client));
1620 g_object_ref (auth);
1621 client->auth = auth;
1623 g_object_unref (old);
1629 * gst_rtsp_client_get_auth:
1630 * @client: a #GstRTSPClient
1632 * Get the #GstRTSPAuth used as the authentication manager of @client.
1634 * Returns: the #GstRTSPAuth of @client. g_object_unref() after
1638 gst_rtsp_client_get_auth (GstRTSPClient * client)
1640 GstRTSPAuth *result;
1642 g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
1644 if ((result = client->auth))
1645 g_object_ref (result);
1650 static GstRTSPResult
1651 message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
1654 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1656 switch (message->type) {
1657 case GST_RTSP_MESSAGE_REQUEST:
1658 handle_request (client, message);
1660 case GST_RTSP_MESSAGE_RESPONSE:
1662 case GST_RTSP_MESSAGE_DATA:
1663 handle_data (client, message);
1671 static GstRTSPResult
1672 message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
1674 /* GstRTSPClient *client; */
1676 /* client = GST_RTSP_CLIENT (user_data); */
1678 /* GST_INFO ("client %p: sent a message with cseq %d", client, cseq); */
1683 static GstRTSPResult
1684 closed (GstRTSPWatch * watch, gpointer user_data)
1686 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1687 const gchar *tunnelid;
1689 GST_INFO ("client %p: connection closed", client);
1691 if ((tunnelid = gst_rtsp_connection_get_tunnelid (client->connection))) {
1692 g_mutex_lock (tunnels_lock);
1693 /* remove from tunnelids */
1694 g_hash_table_remove (tunnels, tunnelid);
1695 g_mutex_unlock (tunnels_lock);
1701 static GstRTSPResult
1702 error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
1704 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1707 str = gst_rtsp_strresult (result);
1708 GST_INFO ("client %p: received an error %s", client, str);
1714 static GstRTSPResult
1715 error_full (GstRTSPWatch * watch, GstRTSPResult result,
1716 GstRTSPMessage * message, guint id, gpointer user_data)
1718 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1721 str = gst_rtsp_strresult (result);
1723 ("client %p: received an error %s when handling message %p with id %d",
1724 client, str, message, id);
1731 remember_tunnel (GstRTSPClient * client)
1733 const gchar *tunnelid;
1735 /* store client in the pending tunnels */
1736 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
1737 if (tunnelid == NULL)
1740 GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
1742 /* we can't have two clients connecting with the same tunnelid */
1743 g_mutex_lock (tunnels_lock);
1744 if (g_hash_table_lookup (tunnels, tunnelid))
1745 goto tunnel_existed;
1747 g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
1748 g_mutex_unlock (tunnels_lock);
1755 GST_ERROR ("client %p: no tunnelid provided", client);
1760 g_mutex_unlock (tunnels_lock);
1761 GST_ERROR ("client %p: tunnel session %s already existed", client,
1767 static GstRTSPStatusCode
1768 tunnel_start (GstRTSPWatch * watch, gpointer user_data)
1770 GstRTSPClient *client;
1772 client = GST_RTSP_CLIENT (user_data);
1774 GST_INFO ("client %p: tunnel start (connection %p)", client,
1775 client->connection);
1777 if (!remember_tunnel (client))
1780 return GST_RTSP_STS_OK;
1785 GST_ERROR ("client %p: error starting tunnel", client);
1786 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1790 static GstRTSPResult
1791 tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
1793 GstRTSPClient *client;
1795 client = GST_RTSP_CLIENT (user_data);
1797 GST_INFO ("client %p: tunnel lost (connection %p)", client,
1798 client->connection);
1800 /* ignore error, it'll only be a problem when the client does a POST again */
1801 remember_tunnel (client);
1806 static GstRTSPResult
1807 tunnel_complete (GstRTSPWatch * watch, gpointer user_data)
1809 const gchar *tunnelid;
1810 GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
1811 GstRTSPClient *oclient;
1813 GST_INFO ("client %p: tunnel complete", client);
1815 /* find previous tunnel */
1816 tunnelid = gst_rtsp_connection_get_tunnelid (client->connection);
1817 if (tunnelid == NULL)
1820 g_mutex_lock (tunnels_lock);
1821 if (!(oclient = g_hash_table_lookup (tunnels, tunnelid)))
1824 /* remove the old client from the table. ref before because removing it will
1825 * remove the ref to it. */
1826 g_object_ref (oclient);
1827 g_hash_table_remove (tunnels, tunnelid);
1829 if (oclient->watch == NULL)
1831 g_mutex_unlock (tunnels_lock);
1833 GST_INFO ("client %p: found tunnel %p (old %p, new %p)", client, oclient,
1834 oclient->connection, client->connection);
1836 /* merge the tunnels into the first client */
1837 gst_rtsp_connection_do_tunnel (oclient->connection, client->connection);
1838 gst_rtsp_watch_reset (oclient->watch);
1839 g_object_unref (oclient);
1841 /* we don't need this watch anymore */
1842 g_source_destroy ((GSource *) client->watch);
1843 client->watchid = 0;
1844 client->watch = NULL;
1851 GST_INFO ("client %p: no tunnelid provided", client);
1852 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1856 g_mutex_unlock (tunnels_lock);
1857 GST_INFO ("client %p: tunnel session %s not found", client, tunnelid);
1858 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1862 g_mutex_unlock (tunnels_lock);
1863 GST_INFO ("client %p: tunnel session %s was closed", client, tunnelid);
1864 g_object_unref (oclient);
1865 return GST_RTSP_STS_SERVICE_UNAVAILABLE;
1869 static GstRTSPWatchFuncs watch_funcs = {
1881 client_watch_notify (GstRTSPClient * client)
1883 GST_INFO ("client %p: watch destroyed", client);
1884 client->watchid = 0;
1885 client->watch = NULL;
1886 g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
1887 g_object_unref (client);
1891 * gst_rtsp_client_attach:
1892 * @client: a #GstRTSPClient
1893 * @socket: a #GSocket
1894 * @cancellable: a #GCancellable
1897 * Accept a new connection for @client on @socket.
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, GSocket * socket,
1906 GCancellable * cancellable, GError ** error)
1908 GstRTSPConnection *conn;
1910 GSocket *read_socket;
1911 GSocketAddress *addres;
1913 GMainContext *context;
1915 struct sockaddr_storage addr;
1917 gchar ip[INET6_ADDRSTRLEN];
1919 /* a new client connected. */
1920 GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, cancellable),
1923 read_socket = gst_rtsp_connection_get_read_socket (conn);
1924 client->is_ipv6 = g_socket_get_family (socket) == G_SOCKET_FAMILY_IPV6;
1926 if (!(addres = g_socket_get_remote_address (read_socket, error)))
1929 addrlen = sizeof (addr);
1930 if (!g_socket_address_to_native (addres, &addr, addrlen, error))
1933 if (getnameinfo ((struct sockaddr *) &addr, addrlen, ip, sizeof (ip), NULL, 0,
1934 NI_NUMERICHOST) != 0)
1935 goto getnameinfo_failed;
1937 /* keep the original ip that the client connected to */
1938 g_free (client->server_ip);
1939 client->server_ip = g_strndup (ip, sizeof (ip));
1941 GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
1942 client->server_ip, client->is_ipv6);
1944 url = gst_rtsp_connection_get_url (conn);
1945 GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
1947 client->connection = conn;
1949 /* create watch for the connection and attach */
1950 client->watch = gst_rtsp_watch_new (client->connection, &watch_funcs,
1951 g_object_ref (client), (GDestroyNotify) client_watch_notify);
1953 /* find the context to add the watch */
1954 if ((source = g_main_current_source ()))
1955 context = g_source_get_context (source);
1959 GST_INFO ("attaching to context %p", context);
1961 client->watchid = gst_rtsp_watch_attach (client->watch, context);
1962 gst_rtsp_watch_unref (client->watch);
1969 gchar *str = gst_rtsp_strresult (res);
1971 GST_ERROR ("Could not accept client on server socket %p: %s", socket, str);
1977 GST_ERROR ("could not get remote address %s", (*error)->message);
1982 GST_ERROR ("could not get native address %s", (*error)->message);
1987 GST_ERROR ("getnameinfo failed: %s", g_strerror (errno));