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gstreamer/gst/rtsp-server/rtsp-client.c

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/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <sys/ioctl.h>
#include <gst/sdp/gstsdpmessage.h>
#include "rtsp-client.h"
#undef DEBUG
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
static void
gst_rtsp_client_class_init (GstRTSPClientClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
}
/**
* gst_rtsp_client_new:
*
* Create a new #GstRTSPClient instance.
*/
GstRTSPClient *
gst_rtsp_client_new (void)
{
GstRTSPClient *result;
result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
return result;
}
static void
handle_generic_response (GstRTSPClient *client, GstRTSPStatusCode code,
GstRTSPMessage *request)
{
GstRTSPMessage response = { 0 };
gst_rtsp_message_init_response (&response, code,
gst_rtsp_status_as_text (code), request);
gst_rtsp_connection_send (client->connection, &response, NULL);
}
static gboolean
handle_teardown_response (GstRTSPClient *client, const gchar *uri, GstRTSPMessage *request)
{
GstRTSPResult res;
GstRTSPSessionMedia *media;
GstRTSPSession *session;
gchar *sessid;
GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
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->pool, sessid)))
goto session_not_found;
}
else
goto service_unavailable;
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, client->media);
if (!media)
goto not_found;
gst_rtsp_session_media_stop (media);
gst_rtsp_session_pool_remove (client->pool, session);
g_object_unref (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);
/* 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_connection_send (client->connection, &response, NULL);
return FALSE;
/* ERRORS */
session_not_found:
{
handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
return FALSE;
}
service_unavailable:
{
return FALSE;
}
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
}
static gboolean
handle_pause_response (GstRTSPClient *client, const gchar *uri, GstRTSPMessage *request)
{
GstRTSPResult res;
GstRTSPSessionMedia *media;
GstRTSPSession *session;
gchar *sessid;
GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
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->pool, sessid)))
goto session_not_found;
}
else
goto service_unavailable;
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, client->media);
if (!media)
goto not_found;
gst_rtsp_session_media_pause (media);
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_connection_send (client->connection, &response, NULL);
return FALSE;
/* ERRORS */
session_not_found:
{
handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
return FALSE;
}
service_unavailable:
{
return FALSE;
}
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
}
static gboolean
handle_play_response (GstRTSPClient *client, const gchar *uri, GstRTSPMessage *request)
{
GstRTSPResult res;
GstRTSPSessionMedia *media;
GstRTSPSession *session;
gchar *sessid;
GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
GstStateChangeReturn ret;
GString *rtpinfo;
guint n_streams, i;
guint timestamp, seqnum;
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->pool, sessid)))
goto session_not_found;
}
else
goto service_unavailable;
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, client->media);
if (!media)
goto not_found;
/* wait for paused to get the caps */
ret = gst_rtsp_session_media_pause (media);
switch (ret) {
case GST_STATE_CHANGE_NO_PREROLL:
break;
case GST_STATE_CHANGE_SUCCESS:
break;
case GST_STATE_CHANGE_FAILURE:
goto service_unavailable;
case GST_STATE_CHANGE_ASYNC:
ret = gst_element_get_state (media->pipeline, NULL, NULL, -1);
break;
}
/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
n_streams = gst_rtsp_media_n_streams (client->media);
for (i = 0; i < n_streams; i++) {
GstRTSPMediaStream *stream;
stream = gst_rtsp_media_get_stream (client->media, i);
g_object_get (G_OBJECT (stream->payloader), "seqnum", &seqnum, NULL);
g_object_get (G_OBJECT (stream->payloader), "timestamp", &timestamp, NULL);
if (i > 0)
g_string_append (rtpinfo, ", ");
g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u", uri, i, seqnum, timestamp);
}
/* 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 RTP-Info header */
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_RTP_INFO, rtpinfo->str);
g_string_free (rtpinfo, TRUE);
gst_rtsp_connection_send (client->connection, &response, NULL);
/* start playing after sending the request */
gst_rtsp_session_media_play (media);
g_object_unref (session);
return FALSE;
/* ERRORS */
session_not_found:
{
handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
return FALSE;
}
service_unavailable:
{
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
return FALSE;
}
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
}
static gboolean
handle_setup_response (GstRTSPClient *client, const gchar *uri, GstRTSPMessage *request)
{
GstRTSPResult res;
gchar *sessid;
gchar *transport;
gchar **transports;
gboolean have_transport;
GstRTSPTransport *ct, *st;
GstRTSPSession *session;
gint i;
GstRTSPLowerTrans supported;
GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
GstRTSPSessionStream *stream;
gchar *trans_str, *pos;
guint streamid;
GstRTSPSessionMedia *media;
gboolean need_session;
/* find the media associated with the uri */
if (client->media == NULL) {
if ((client->media = gst_rtsp_media_new (uri)) == NULL)
goto not_found;
}
/* parse the transport */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_TRANSPORT, &transport, 0);
if (res != GST_RTSP_OK)
goto unsupported_transports;
transports = g_strsplit (transport, ",", 0);
gst_rtsp_transport_new (&ct);
/* loop through the transports, try to parse */
have_transport = FALSE;
for (i = 0; transports[i]; i++) {
gst_rtsp_transport_init (ct);
res = gst_rtsp_transport_parse (transports[i], ct);
if (res == GST_RTSP_OK) {
have_transport = TRUE;
break;
}
}
g_strfreev (transports);
/* we have not found anything usable, error out */
if (!have_transport) {
gst_rtsp_transport_free (ct);
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))
goto unsupported_transports;
/* 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->pool, sessid)))
goto session_not_found;
need_session = FALSE;
}
else {
/* create a session if this fails we probably reached our session limit or
* something. */
if (!(session = gst_rtsp_session_pool_create (client->pool)))
goto service_unavailable;
need_session = TRUE;
}
/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, client->media);
if (!media)
goto not_found;
/* parse the stream we need to configure */
if (!(pos = strstr (uri, "stream=")))
goto bad_request;
pos += strlen ("stream=");
if (sscanf (pos, "%u", &streamid) != 1)
goto bad_request;
/* get a handle to the stream in the media */
stream = gst_rtsp_session_get_stream (media, streamid);
/* setup the server transport from the client transport */
st = gst_rtsp_session_stream_set_transport (stream, inet_ntoa (client->address.sin_addr), ct);
/* serialize the server transport */
trans_str = gst_rtsp_transport_as_text (st);
/* construct the response now */
code = GST_RTSP_STS_OK;
gst_rtsp_message_init_response (&response, code, gst_rtsp_status_as_text (code), request);
if (need_session)
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_SESSION, session->sessionid);
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_TRANSPORT, trans_str);
g_free (trans_str);
g_object_unref (session);
gst_rtsp_connection_send (client->connection, &response, NULL);
return TRUE;
/* ERRORS */
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
bad_request:
{
handle_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
return FALSE;
}
session_not_found:
{
handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
return FALSE;
}
unsupported_transports:
{
handle_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
return FALSE;
}
service_unavailable:
{
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
return FALSE;
}
}
static gboolean
handle_describe_response (GstRTSPClient *client, const gchar *uri, GstRTSPMessage *request)
{
GstRTSPMessage response = { 0 };
GstSDPMessage *sdp;
guint n_streams, i;
gchar *sdptext;
GstRTSPMedia *media;
GstElement *pipeline = NULL;
/* check what kind of format is accepted */
/* for the describe we must generate an SDP */
if (!(media = gst_rtsp_media_new (uri)))
goto no_media;
/* create a pipeline if we have to */
if (pipeline == NULL) {
pipeline = gst_pipeline_new ("client-pipeline");
}
/* prepare the media into the pipeline */
if (!gst_rtsp_media_prepare (media, GST_BIN (pipeline)))
goto no_media;
/* link fakesink to all stream pads and set the pipeline to PLAYING */
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0; i < n_streams; i++) {
GstRTSPMediaStream *stream;
GstElement *sink;
GstPad *sinkpad;
stream = gst_rtsp_media_get_stream (media, i);
sink = gst_element_factory_make ("fakesink", NULL);
gst_bin_add (GST_BIN (pipeline), sink);
sinkpad = gst_element_get_static_pad (sink, "sink");
gst_pad_link (stream->srcpad, sinkpad);
gst_object_unref (sinkpad);
}
/* now play and wait till we get the pads blocked. At that time the pipeline
* is prerolled and we have the caps on the streams too. */
gst_element_set_state (pipeline, GST_STATE_PLAYING);
/* wait for state change to complete */
gst_element_get_state (pipeline, NULL, NULL, -1);
/* we should now be able to construct the SDP message */
gst_sdp_message_new (&sdp);
/* some standard things first */
gst_sdp_message_set_version (sdp, "0");
gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", "IP4", "127.0.0.1");
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");
for (i = 0; i < n_streams; i++) {
GstRTSPMediaStream *stream;
GstSDPMedia *smedia;
GstStructure *s;
const gchar *caps_str, *caps_enc, *caps_params;
gchar *tmp;
gint caps_pt, caps_rate;
guint n_fields, j;
gboolean first;
GString *fmtp;
stream = gst_rtsp_media_get_stream (media, i);
gst_sdp_media_new (&smedia);
s = gst_caps_get_structure (stream->caps, 0);
/* get media type and payload for the m= line */
caps_str = gst_structure_get_string (s, "media");
gst_sdp_media_set_media (smedia, caps_str);
gst_structure_get_int (s, "payload", &caps_pt);
tmp = g_strdup_printf ("%d", caps_pt);
gst_sdp_media_add_format (smedia, tmp);
g_free (tmp);
gst_sdp_media_set_port_info (smedia, 0, 1);
gst_sdp_media_set_proto (smedia, "RTP/AVP");
/* for the c= line */
gst_sdp_media_add_connection (smedia, "IN", "IP4", "127.0.0.1", 0, 0);
/* get clock-rate, media type and params for the rtpmap attribute */
gst_structure_get_int (s, "clock-rate", &caps_rate);
caps_enc = gst_structure_get_string (s, "encoding-name");
caps_params = gst_structure_get_string (s, "encoding-params");
if (caps_params)
tmp = g_strdup_printf ("%d %s/%d/%s", caps_pt, caps_enc, caps_rate,
caps_params);
else
tmp = g_strdup_printf ("%d %s/%d", caps_pt, caps_enc, caps_rate);
gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
g_free (tmp);
/* the config uri */
tmp = g_strdup_printf ("stream=%d", i);
gst_sdp_media_add_attribute (smedia, "control", tmp);
g_free (tmp);
/* collect all other properties and add them to fmtp */
fmtp = g_string_new ("");
g_string_append_printf (fmtp, "%d ", caps_pt);
first = TRUE;
n_fields = gst_structure_n_fields (s);
for (j = 0; j < n_fields; j++) {
const gchar *fname, *fval;
fname = gst_structure_nth_field_name (s, j);
/* filter out standard properties */
if (!strcmp (fname, "media"))
continue;
if (!strcmp (fname, "payload"))
continue;
if (!strcmp (fname, "clock-rate"))
continue;
if (!strcmp (fname, "encoding-name"))
continue;
if (!strcmp (fname, "encoding-params"))
continue;
if (!strcmp (fname, "ssrc"))
continue;
if (!strcmp (fname, "clock-base"))
continue;
if (!strcmp (fname, "seqnum-base"))
continue;
if ((fval = gst_structure_get_string (s, fname))) {
g_string_append_printf (fmtp, "%s%s=%s", first ? "":";", fname, fval);
first = FALSE;
}
}
if (!first) {
tmp = g_string_free (fmtp, FALSE);
gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
g_free (tmp);
}
else {
g_string_free (fmtp, TRUE);
}
gst_sdp_message_add_media (sdp, smedia);
}
/* go back to NULL */
gst_element_set_state (pipeline, GST_STATE_NULL);
g_object_unref (media);
gst_object_unref (pipeline);
pipeline = NULL;
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
/* add SDP to the response body */
sdptext = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (&response, (guint8 *)sdptext, strlen (sdptext));
gst_sdp_message_free (sdp);
gst_rtsp_connection_send (client->connection, &response, NULL);
return TRUE;
/* ERRORS */
no_media:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
}
static void
handle_options_response (GstRTSPClient *client, const gchar *uri, GstRTSPMessage *request)
{
GstRTSPMessage response = { 0 };
GstRTSPMethod options;
GString *str;
gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
options = GST_RTSP_DESCRIBE |
GST_RTSP_OPTIONS |
// GST_RTSP_PAUSE |
GST_RTSP_PLAY |
GST_RTSP_SETUP |
GST_RTSP_TEARDOWN;
/* always return options.. */
str = g_string_new ("OPTIONS");
if (options & GST_RTSP_DESCRIBE)
g_string_append (str, ", DESCRIBE");
if (options & GST_RTSP_ANNOUNCE)
g_string_append (str, ", ANNOUNCE");
if (options & GST_RTSP_GET_PARAMETER)
g_string_append (str, ", GET_PARAMETER");
if (options & GST_RTSP_PAUSE)
g_string_append (str, ", PAUSE");
if (options & GST_RTSP_PLAY)
g_string_append (str, ", PLAY");
if (options & GST_RTSP_RECORD)
g_string_append (str, ", RECORD");
if (options & GST_RTSP_REDIRECT)
g_string_append (str, ", REDIRECT");
if (options & GST_RTSP_SETUP)
g_string_append (str, ", SETUP");
if (options & GST_RTSP_SET_PARAMETER)
g_string_append (str, ", SET_PARAMETER");
if (options & GST_RTSP_TEARDOWN)
g_string_append (str, ", TEARDOWN");
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_PUBLIC, str->str);
g_string_free (str, TRUE);
gst_rtsp_connection_send (client->connection, &response, NULL);
}
/* this function runs in a client specific thread and handles all rtsp messages
* with the client */
static gpointer
handle_client (GstRTSPClient *client)
{
GstRTSPMessage request = { 0 };
GstRTSPResult res;
GstRTSPMethod method;
const gchar *uri;
GstRTSPVersion version;
while (TRUE) {
/* start by waiting for a message from the client */
res = gst_rtsp_connection_receive (client->connection, &request, NULL);
if (res < 0)
goto receive_failed;
#ifdef DEBUG
gst_rtsp_message_dump (&request);
#endif
gst_rtsp_message_parse_request (&request, &method, &uri, &version);
if (version != GST_RTSP_VERSION_1_0) {
/* we can only handle 1.0 requests */
handle_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED, &request);
continue;
}
/* now see what is asked and dispatch to a dedicated handler */
switch (method) {
case GST_RTSP_OPTIONS:
handle_options_response (client, uri, &request);
break;
case GST_RTSP_DESCRIBE:
handle_describe_response (client, uri, &request);
break;
case GST_RTSP_SETUP:
handle_setup_response (client, uri, &request);
break;
case GST_RTSP_PLAY:
handle_play_response (client, uri, &request);
break;
case GST_RTSP_PAUSE:
handle_pause_response (client, uri, &request);
break;
case GST_RTSP_TEARDOWN:
handle_teardown_response (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:
handle_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, &request);
break;
case GST_RTSP_INVALID:
default:
handle_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &request);
break;
}
}
g_object_unref (client);
return NULL;
/* ERRORS */
receive_failed:
{
g_print ("receive failed, disconnect client %p\n", client);
gst_rtsp_connection_close (client->connection);
g_object_unref (client);
return NULL;
}
}
/* called when we need to accept a new request from a client */
static gboolean
client_accept (GstRTSPClient *client, GIOChannel *source)
{
/* a new client connected. */
int server_sock_fd;
unsigned int address_len;
GstRTSPConnection *conn;
conn = client->connection;
server_sock_fd = g_io_channel_unix_get_fd (source);
address_len = sizeof (client->address);
memset (&client->address, 0, address_len);
conn->fd.fd = accept (server_sock_fd, (struct sockaddr *) &client->address,
&address_len);
if (conn->fd.fd == -1)
goto accept_failed;
g_print ("added new client %p ip %s with fd %d\n", client,
inet_ntoa (client->address.sin_addr), conn->fd.fd);
/* FIXME some hackery, we need to have a connection method to accept server
* connections */
gst_poll_add_fd (conn->fdset, &conn->fd);
return TRUE;
/* ERRORS */
accept_failed:
{
g_error ("Could not accept client on server socket %d: %s (%d)",
server_sock_fd, g_strerror (errno), errno);
return FALSE;
}
}
/**
* gst_rtsp_client_set_session_pool:
* @client: a #GstRTSPClient
* @pool: a #GstRTSPSessionPool
*
* Set @pool as the sessionpool for @client which it will use to find
* or allocate sessions.
*/
void
gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *pool)
{
GstRTSPSessionPool *old;
old = client->pool;
if (pool)
g_object_ref (pool);
client->pool = pool;
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_client_get_session_pool:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
*
* Returns: a #GstRTSPSessionPool, unref after usage.
*/
GstRTSPSessionPool *
gst_rtsp_client_get_session_pool (GstRTSPClient *client)
{
GstRTSPSessionPool *result;
if ((result = client->pool))
g_object_ref (result);
return result;
}
/**
* gst_rtsp_client_attach:
* @client: a #GstRTSPClient
* @context: a #GMainContext
*
* Attaches @client to @context. When the mainloop for @context is run, the
* client will be dispatched.
*
* 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 *source)
{
gst_rtsp_connection_create (NULL, &client->connection);
if (!client_accept (client, source))
goto accept_failed;
/* client accepted, spawn a thread for the client */
g_object_ref (client);
client->thread = g_thread_create ((GThreadFunc)handle_client, client, TRUE, NULL);
return TRUE;
/* ERRORS */
accept_failed:
{
gst_rtsp_connection_close (client->connection);
return FALSE;
}
}
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