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 "rtsp-client.h"
#include "rtsp-sdp.h"
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#undef DEBUG
static void gst_rtsp_client_finalize (GObject * obj);
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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);
gobject_class->finalize = gst_rtsp_client_finalize;
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}
static void
gst_rtsp_client_init (GstRTSPClient * client)
{
}
static void
gst_rtsp_client_finalize (GObject * obj)
{
G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
}
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/**
* 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_response (GstRTSPClient *client, GstRTSPMessage *response)
{
#ifdef DEBUG
gst_rtsp_message_dump (response);
#endif
gst_rtsp_connection_send (client->connection, response, NULL);
}
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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);
handle_response (client, &response);
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}
static GstRTSPMedia *
find_media (GstRTSPClient *client, const GstRTSPUrl *uri, GstRTSPMessage *request)
{
GstRTSPMediaFactory *factory;
GstRTSPMedia *media;
/* find the factory for the uri first */
if (!(factory = gst_rtsp_media_mapping_find_factory (client->media_mapping, uri)))
goto no_factory;
/* prepare the media and add it to the pipeline */
if (!(media = gst_rtsp_media_factory_construct (factory, uri)))
goto no_media;
/* prepare the media */
if (!(gst_rtsp_media_prepare (media)))
goto no_prepare;
return media;
/* ERRORS */
no_factory:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return NULL;
}
no_media:
{
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
g_object_unref (factory);
return NULL;
}
no_prepare:
{
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
g_object_unref (media);
g_object_unref (factory);
return NULL;
}
}
/* Get the session or NULL when there was no session */
static GstRTSPSession *
ensure_session (GstRTSPClient *client, GstRTSPMessage *request)
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{
GstRTSPResult res;
GstRTSPSession *session;
gchar *sessid;
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
if (res == GST_RTSP_OK) {
/* we had a session in the request, find it again */
if (!(session = gst_rtsp_session_pool_find (client->session_pool, sessid)))
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goto session_not_found;
}
else
goto service_unavailable;
return session;
/* ERRORS */
session_not_found:
{
handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
return NULL;
}
service_unavailable:
{
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
return NULL;
}
}
static gboolean
handle_teardown_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
{
GstRTSPSessionMedia *media;
GstRTSPSession *session;
GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
if (!(session = ensure_session (client, request)))
goto no_session;
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/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, uri);
if (!media)
goto not_found;
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gst_rtsp_session_media_stop (media);
gst_rtsp_session_pool_remove (client->session_pool, session);
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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);
handle_response (client, &response);
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return FALSE;
/* ERRORS */
no_session:
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{
/* error was sent already */
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return FALSE;
}
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
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}
static gboolean
handle_pause_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
GstRTSPSessionMedia *media;
GstRTSPSession *session;
GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
if (!(session = ensure_session (client, request)))
goto no_session;
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/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, uri);
if (!media)
goto not_found;
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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);
handle_response (client, &response);
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return FALSE;
/* ERRORS */
no_session:
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{
return FALSE;
}
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
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}
static gboolean
handle_play_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
GstRTSPSessionMedia *media;
GstRTSPSession *session;
GstRTSPMessage response = { 0 };
GstRTSPStatusCode code;
GString *rtpinfo;
guint n_streams, i;
guint timestamp, seqnum;
if (!(session = ensure_session (client, request)))
goto no_session;
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/* get a handle to the configuration of the media in the session */
media = gst_rtsp_session_get_media (session, uri);
if (!media)
goto not_found;
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/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
n_streams = gst_rtsp_media_n_streams (media->media);
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for (i = 0; i < n_streams; i++) {
GstRTSPMediaStream *stream;
gchar *uristr;
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stream = gst_rtsp_media_get_stream (media->media, i);
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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, ", ");
uristr = gst_rtsp_url_get_request_uri (uri);
g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u", uristr, i, seqnum, timestamp);
g_free (uristr);
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}
/* 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);
handle_response (client, &response);
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/* start playing after sending the request */
gst_rtsp_session_media_play (media);
g_object_unref (session);
return FALSE;
/* ERRORS */
no_session:
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{
/* error was sent */
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return FALSE;
}
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
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}
static gboolean
handle_setup_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
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;
/* the uri contains the stream number we added in the SDP config, which is
* always /stream=%d so we need to strip that off
* parse the stream we need to configure, look for the stream in the abspath
* first and then in the query. */
if (!(pos = strstr (uri->abspath, "/stream="))) {
if (!(pos = strstr (uri->query, "/stream=")))
goto bad_request;
}
/* we can mofify the parse uri in place */
*pos = '\0';
pos += strlen ("/stream=");
if (sscanf (pos, "%u", &streamid) != 1)
goto bad_request;
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/* parse the transport */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_TRANSPORT, &transport, 0);
if (res != GST_RTSP_OK)
goto no_transport;
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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);
g_free (ct->destination);
ct->destination = g_strdup (inet_ntoa (client->address.sin_addr));
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/* we have not found anything usable, error out */
if (!have_transport) {
goto unsupported_transports;
}
/* we have a valid transport, check if we can handle it */
if (ct->trans != GST_RTSP_TRANS_RTP)
goto unsupported_transports;
if (ct->profile != GST_RTSP_PROFILE_AVP)
goto unsupported_transports;
supported = GST_RTSP_LOWER_TRANS_UDP |
GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP;
if (!(ct->lower_transport & supported))
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->session_pool, sessid)))
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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->session_pool)))
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goto service_unavailable;
need_session = TRUE;
}
if (need_session) {
GstRTSPMedia *m;
/* get a handle to the configuration of the media in the session */
if ((m = find_media (client, uri, request))) {
media = gst_rtsp_session_manage_media (session, uri, m);
}
}
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/* get a handle to the configuration of the media in the session */
if (!(media = gst_rtsp_session_get_media (session, uri)))
goto not_found;
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/* get a handle to the stream in the media */
if (!(stream = gst_rtsp_session_media_get_stream (media, streamid)))
goto no_stream;
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/* setup the server transport from the client transport */
st = gst_rtsp_session_stream_set_transport (stream, ct);
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/* 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);
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gst_rtsp_message_add_header (&response, GST_RTSP_HDR_TRANSPORT, trans_str);
g_free (trans_str);
g_object_unref (session);
handle_response (client, &response);
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return TRUE;
/* ERRORS */
bad_request:
{
handle_generic_response (client, GST_RTSP_STS_BAD_REQUEST, request);
return FALSE;
}
not_found:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
no_stream:
{
handle_generic_response (client, GST_RTSP_STS_NOT_FOUND, request);
return FALSE;
}
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session_not_found:
{
handle_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, request);
return FALSE;
}
no_transport:
{
handle_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
return FALSE;
}
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unsupported_transports:
{
handle_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, request);
gst_rtsp_transport_free (ct);
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return FALSE;
}
service_unavailable:
{
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
return FALSE;
}
}
/* for the describe we must generate an SDP */
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static gboolean
handle_describe_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
GstRTSPMessage response = { 0 };
GstRTSPResult res;
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GstSDPMessage *sdp;
guint i;
gchar *str;
GstRTSPMedia *media;
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/* check what kind of format is accepted, we don't really do anything with it
* and always return SDP for now. */
for (i = 0; i++; ) {
gchar *accept;
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_ACCEPT, &accept, i);
if (res == GST_RTSP_ENOTIMPL)
break;
if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
break;
}
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/* find the media object for the uri */
if (!(media = find_media (client, uri, request)))
goto no_media;
/* create an SDP for the media object */
if (!(sdp = gst_rtsp_sdp_from_media (media)))
goto no_sdp;
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gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_TYPE, "application/sdp");
str = g_strdup_printf ("rtsp://%s:%u%s/", uri->host, uri->port, uri->abspath);
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_CONTENT_BASE, str);
g_free (str);
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/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
gst_rtsp_message_take_body (&response, (guint8 *)str, strlen (str));
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gst_sdp_message_free (sdp);
handle_response (client, &response);
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return TRUE;
/* ERRORS */
no_media:
{
/* error reply is already sent */
return FALSE;
}
no_sdp:
{
handle_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, request);
g_object_unref (media);
return FALSE;
}
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}
static void
handle_options_response (GstRTSPClient *client, GstRTSPUrl *uri, GstRTSPMessage *request)
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{
GstRTSPMessage response = { 0 };
GstRTSPMethod options;
gchar *str;
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options = GST_RTSP_DESCRIBE |
GST_RTSP_OPTIONS |
// GST_RTSP_PAUSE |
GST_RTSP_PLAY |
GST_RTSP_SETUP |
GST_RTSP_TEARDOWN;
str = gst_rtsp_options_as_text (options);
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gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK,
gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
gst_rtsp_message_add_header (&response, GST_RTSP_HDR_PUBLIC, str);
g_free (str);
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handle_response (client, &response);
}
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/* remove duplicate and trailing '/' */
static void
santize_uri (GstRTSPUrl *uri)
{
gint i, len;
gchar *s, *d;
gboolean have_slash, prev_slash;
s = d = uri->abspath;
len = strlen (uri->abspath);
prev_slash = FALSE;
for (i = 0; i < len; i++) {
have_slash = s[i] == '/';
*d = s[i];
if (!have_slash || !prev_slash)
d++;
prev_slash = have_slash;
}
len = d - uri->abspath;
/* don't remove the first slash if that's the only thing left */
if (len > 1 && *(d-1) == '/')
d--;
*d = '\0';
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}
/* 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 *uristr;
GstRTSPUrl *uri;
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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, &uristr, &version);
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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;
}
/* we always try to parse the url first */
if ((res = gst_rtsp_url_parse (uristr, &uri)) != GST_RTSP_OK) {
handle_generic_response (client, GST_RTSP_STS_BAD_REQUEST, &request);
continue;
}
/* sanitize the uri */
santize_uri (uri);
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/* 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;
}
gst_rtsp_url_free (uri);
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}
g_object_unref (client);
return NULL;
/* ERRORS */
receive_failed:
{
g_message ("receive failed %d (%s), disconnect client %p", res,
gst_rtsp_strresult (res), client);
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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 *channel)
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{
/* a new client connected. */
int server_sock_fd, fd;
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unsigned int address_len;
GstRTSPConnection *conn;
server_sock_fd = g_io_channel_unix_get_fd (channel);
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address_len = sizeof (client->address);
memset (&client->address, 0, address_len);
fd = accept (server_sock_fd, (struct sockaddr *) &client->address,
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&address_len);
if (fd == -1)
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goto accept_failed;
/* now create the connection object */
gst_rtsp_connection_create (NULL, &conn);
conn->fd.fd = fd;
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/* FIXME some hackery, we need to have a connection method to accept server
* connections */
gst_poll_add_fd (conn->fdset, &conn->fd);
g_message ("added new client %p ip %s with fd %d", client,
inet_ntoa (client->address.sin_addr), conn->fd.fd);
client->connection = conn;
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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. the sessionpool is usually inherited from the server
* that created the client but can be overridden later.
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*/
void
gst_rtsp_client_set_session_pool (GstRTSPClient *client, GstRTSPSessionPool *pool)
{
GstRTSPSessionPool *old;
old = client->session_pool;
if (old != pool) {
if (pool)
g_object_ref (pool);
client->session_pool = pool;
if (old)
g_object_unref (old);
}
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}
/**
* 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->session_pool))
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g_object_ref (result);
return result;
}
/**
* gst_rtsp_client_set_media_mapping:
* @client: a #GstRTSPClient
* @mapping: a #GstRTSPMediaMapping
*
* Set @mapping as the media mapping for @client which it will use to map urls
* to media streams. These mapping is usually inherited from the server that
* created the client but can be overriden later.
*/
void
gst_rtsp_client_set_media_mapping (GstRTSPClient *client, GstRTSPMediaMapping *mapping)
{
GstRTSPMediaMapping *old;
old = client->media_mapping;
if (old != mapping) {
if (mapping)
g_object_ref (mapping);
client->media_mapping = mapping;
if (old)
g_object_unref (old);
}
}
/**
* gst_rtsp_client_get_media_mapping:
* @client: a #GstRTSPClient
*
* Get the #GstRTSPMediaMapping object that @client uses to manage its sessions.
*
* Returns: a #GstRTSPMediaMapping, unref after usage.
*/
GstRTSPMediaMapping *
gst_rtsp_client_get_media_mapping (GstRTSPClient *client)
{
GstRTSPMediaMapping *result;
if ((result = client->media_mapping))
g_object_ref (result);
return result;
}
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/**
* gst_rtsp_client_attach:
* @client: a #GstRTSPClient
* @channel: a #GIOChannel
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*
* Accept a new connection for @client on the socket in @source.
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*
* This function should be called when the client properties and urls are fully
* configured and the client is ready to start.
*
* Returns: %TRUE if the client could be accepted.
*/
gboolean
gst_rtsp_client_accept (GstRTSPClient *client, GIOChannel *channel)
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{
if (!client_accept (client, channel))
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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:
{
return FALSE;
}
}