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/*****************************************************************************
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* rtsp.c: RTSP support for RTP stream output module
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*****************************************************************************
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* Copyright (C) 2003-2004 the VideoLAN team
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* Copyright © 2007 Rémi Denis-Courmont
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* Authors: Laurent Aimar <fenrir@via.ecp.fr>
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
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*****************************************************************************/
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/*****************************************************************************
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*****************************************************************************/
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#include <vlc_common.h>
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#include <vlc_httpd.h>
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#include <vlc_network.h>
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typedef struct rtsp_session_t rtsp_session_t;
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const char *track_fmt;
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rtsp_session_t **sessionv;
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static int RtspCallback( httpd_callback_sys_t *p_args,
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httpd_client_t *cl, httpd_message_t *answer,
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const httpd_message_t *query );
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static int RtspCallbackId( httpd_callback_sys_t *p_args,
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httpd_client_t *cl, httpd_message_t *answer,
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const httpd_message_t *query );
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static void RtspClientDel( rtsp_stream_t *rtsp, rtsp_session_t *session );
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rtsp_stream_t *RtspSetup( sout_stream_t *p_stream, const vlc_url_t *url )
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rtsp_stream_t *rtsp = malloc( sizeof( *rtsp ) );
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if( rtsp == NULL || ( url->i_port > 99999 ) )
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rtsp->owner = p_stream;
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rtsp->sessionv = NULL;
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rtsp->psz_path = NULL;
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vlc_mutex_init( &rtsp->lock );
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rtsp->port = (url->i_port > 0) ? url->i_port : 554;
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rtsp->psz_path = strdup( ( url->psz_path != NULL ) ? url->psz_path : "/" );
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if( rtsp->psz_path == NULL )
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assert( strlen( rtsp->psz_path ) > 0 );
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if( rtsp->psz_path[strlen( rtsp->psz_path ) - 1] == '/' )
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rtsp->track_fmt = "%strackID=%u";
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rtsp->track_fmt = "%s/trackID=%u";
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msg_Dbg( p_stream, "RTSP stream: host %s port %d at %s",
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url->psz_host, rtsp->port, rtsp->psz_path );
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rtsp->host = httpd_HostNew( VLC_OBJECT(p_stream), url->psz_host,
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if( rtsp->host == NULL )
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rtsp->url = httpd_UrlNewUnique( rtsp->host, rtsp->psz_path,
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if( rtsp->url == NULL )
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httpd_UrlCatch( rtsp->url, HTTPD_MSG_DESCRIBE, RtspCallback, (void*)rtsp );
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httpd_UrlCatch( rtsp->url, HTTPD_MSG_SETUP, RtspCallback, (void*)rtsp );
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httpd_UrlCatch( rtsp->url, HTTPD_MSG_PLAY, RtspCallback, (void*)rtsp );
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httpd_UrlCatch( rtsp->url, HTTPD_MSG_PAUSE, RtspCallback, (void*)rtsp );
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httpd_UrlCatch( rtsp->url, HTTPD_MSG_GETPARAMETER, RtspCallback,
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httpd_UrlCatch( rtsp->url, HTTPD_MSG_TEARDOWN, RtspCallback, (void*)rtsp );
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void RtspUnsetup( rtsp_stream_t *rtsp )
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while( rtsp->sessionc > 0 )
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RtspClientDel( rtsp, rtsp->sessionv[0] );
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httpd_UrlDelete( rtsp->url );
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httpd_HostDelete( rtsp->host );
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free( rtsp->psz_path );
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vlc_mutex_destroy( &rtsp->lock );
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struct rtsp_stream_id_t
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rtsp_stream_t *stream;
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sout_stream_id_t *sout_id;
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uint16_t loport, hiport;
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typedef struct rtsp_strack_t rtsp_strack_t;
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/* For unicast streaming */
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struct rtsp_session_t
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rtsp_stream_t *stream;
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/* output (id-access) */
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rtsp_strack_t *trackv;
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/* Unicast session track */
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sout_stream_id_t *id;
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rtsp_stream_id_t *RtspAddId( rtsp_stream_t *rtsp, sout_stream_id_t *sid,
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unsigned num, uint32_t ssrc,
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/* Multicast stuff - TODO: cleanup */
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const char *dst, int ttl,
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unsigned loport, unsigned hiport )
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char urlbuf[sizeof( "/trackID=123" ) + strlen( rtsp->psz_path )];
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rtsp_stream_id_t *id = malloc( sizeof( *id ) );
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/* TODO: can we assume that this need not be strdup'd? */
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if( id->dst != NULL )
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/* FIXME: num screws up if any ES has been removed and re-added */
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snprintf( urlbuf, sizeof( urlbuf ), rtsp->track_fmt, rtsp->psz_path,
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msg_Dbg( rtsp->owner, "RTSP: adding %s", urlbuf );
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url = id->url = httpd_UrlNewUnique( rtsp->host, urlbuf, NULL, NULL, NULL );
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httpd_UrlCatch( url, HTTPD_MSG_DESCRIBE, RtspCallbackId, (void *)id );
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httpd_UrlCatch( url, HTTPD_MSG_SETUP, RtspCallbackId, (void *)id );
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httpd_UrlCatch( url, HTTPD_MSG_PLAY, RtspCallbackId, (void *)id );
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httpd_UrlCatch( url, HTTPD_MSG_PAUSE, RtspCallbackId, (void *)id );
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httpd_UrlCatch( url, HTTPD_MSG_GETPARAMETER, RtspCallbackId, (void *)id );
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httpd_UrlCatch( url, HTTPD_MSG_TEARDOWN, RtspCallbackId, (void *)id );
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void RtspDelId( rtsp_stream_t *rtsp, rtsp_stream_id_t *id )
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vlc_mutex_lock( &rtsp->lock );
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for( int i = 0; i < rtsp->sessionc; i++ )
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rtsp_session_t *ses = rtsp->sessionv[i];
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for( int j = 0; j < ses->trackc; j++ )
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if( ses->trackv[j].id == id->sout_id )
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rtsp_strack_t *tr = ses->trackv + j;
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REMOVE_ELEM( ses->trackv, ses->trackc, j );
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vlc_mutex_unlock( &rtsp->lock );
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httpd_UrlDelete( id->url );
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/** rtsp must be locked */
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rtsp_session_t *RtspClientNew( rtsp_stream_t *rtsp )
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rtsp_session_t *s = malloc( sizeof( *s ) );
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vlc_rand_bytes (&s->id, sizeof (s->id));
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TAB_APPEND( rtsp->sessionc, rtsp->sessionv, s );
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/** rtsp must be locked */
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rtsp_session_t *RtspClientGet( rtsp_stream_t *rtsp, const char *name )
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id = strtoull( name, &end, 0x10 );
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/* FIXME: use a hash/dictionary */
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for( i = 0; i < rtsp->sessionc; i++ )
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if( rtsp->sessionv[i]->id == id )
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return rtsp->sessionv[i];
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/** rtsp must be locked */
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void RtspClientDel( rtsp_stream_t *rtsp, rtsp_session_t *session )
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TAB_REMOVE( rtsp->sessionc, rtsp->sessionv, session );
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for( i = 0; i < session->trackc; i++ )
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rtp_del_sink( session->trackv[i].id, session->trackv[i].fd );
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free( session->trackv );
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/** Finds the next transport choice */
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static inline const char *transport_next( const char *str )
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/* Looks for comma */
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str = strchr( str, ',' );
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return NULL; /* No more transport options */
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str++; /* skips comma */
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while( strchr( "\r\n\t ", *str ) )
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return (*str) ? str : NULL;
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/** Finds the next transport parameter */
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static inline const char *parameter_next( const char *str )
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while( strchr( ",;", *str ) == NULL )
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return (*str == ';') ? (str + 1) : NULL;
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/** RTSP requests handler
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* @param id selected track for non-aggregate URLs,
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* NULL for aggregate URLs
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static int RtspHandler( rtsp_stream_t *rtsp, rtsp_stream_id_t *id,
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httpd_message_t *answer,
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const httpd_message_t *query )
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sout_stream_t *p_stream = rtsp->owner;
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const char *psz_session = NULL, *psz;
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char control[sizeof("rtsp://[]:12345") + NI_MAXNUMERICHOST
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+ strlen( rtsp->psz_path )];
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if( answer == NULL || query == NULL || cl == NULL )
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/* Build self-referential control URL */
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char ip[NI_MAXNUMERICHOST], *ptr;
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httpd_ServerIP( cl, ip );
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ptr = strchr( ip, '%' );
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if( strchr( ip, ':' ) != NULL )
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sprintf( control, "rtsp://[%s]:%u%s", ip, rtsp->port,
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sprintf( control, "rtsp://%s:%u%s", ip, rtsp->port,
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answer->i_proto = HTTPD_PROTO_RTSP;
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answer->i_version= 0;
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answer->i_type = HTTPD_MSG_ANSWER;
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answer->p_body = NULL;
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httpd_MsgAdd( answer, "Server", "%s", PACKAGE_STRING );
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/* Date: is always allowed, and sometimes mandatory with RTSP/2.0. */
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if (gmtime_r (&now, &ut) != NULL)
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{ /* RFC1123 format, GMT is mandatory */
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static const char wdays[7][4] = {
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"Sun", "Mon", "Tue", "Wed", "Thu", "Fri", "Sat" };
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static const char mons[12][4] = {
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"Jan", "Feb", "Mar", "Apr", "May", "Jun",
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"Jul", "Aug", "Sep", "Oct", "Nov", "Dec" };
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httpd_MsgAdd (answer, "Date", "%s, %02u %s %04u %02u:%02u:%02u GMT",
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wdays[ut.tm_wday], ut.tm_mday, mons[ut.tm_mon],
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1900 + ut.tm_year, ut.tm_hour, ut.tm_min, ut.tm_sec);
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if( query->i_proto != HTTPD_PROTO_RTSP )
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answer->i_status = 505;
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if( httpd_MsgGet( query, "Require" ) != NULL )
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answer->i_status = 551;
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httpd_MsgAdd( answer, "Unsupported", "%s",
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httpd_MsgGet( query, "Require" ) );
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switch( query->i_type )
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case HTTPD_MSG_DESCRIBE:
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{ /* Aggregate-only */
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answer->i_status = 460;
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answer->i_status = 200;
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httpd_MsgAdd( answer, "Content-Type", "%s", "application/sdp" );
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httpd_MsgAdd( answer, "Content-Base", "%s", control );
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answer->p_body = (uint8_t *)SDPGenerate( rtsp->owner, control );
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if( answer->p_body != NULL )
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answer->i_body = strlen( (char *)answer->p_body );
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answer->i_status = 500;
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case HTTPD_MSG_SETUP:
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/* Non-aggregate-only */
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answer->i_status = 459;
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psz_session = httpd_MsgGet( query, "Session" );
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answer->i_status = 461;
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for( const char *tpt = httpd_MsgGet( query, "Transport" );
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tpt = transport_next( tpt ) )
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bool b_multicast = true, b_unsupp = false;
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unsigned loport = 5004, hiport = 5005; /* from RFC3551 */
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/* Check transport protocol. */
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/* Currently, we only support RTP/AVP over UDP */
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if( strncmp( tpt, "RTP/AVP", 7 ) )
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if( strncmp( tpt, "/UDP", 4 ) == 0 )
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if( strchr( ";,", *tpt ) == NULL )
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/* Parse transport options */
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for( const char *opt = parameter_next( tpt );
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opt = parameter_next( opt ) )
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if( strncmp( opt, "multicast", 9 ) == 0)
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if( strncmp( opt, "unicast", 7 ) == 0 )
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if( sscanf( opt, "client_port=%u-%u", &loport, &hiport )
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if( strncmp( opt, "mode=", 5 ) == 0 )
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if( strncasecmp( opt + 5, "play", 4 )
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&& strncasecmp( opt + 5, "\"PLAY\"", 6 ) )
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if( strncmp( opt,"destination=", 12 ) == 0 )
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answer->i_status = 403;
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* Every other option is unsupported:
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* "source" and "append" are invalid (server-only);
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* "ssrc" also (as clarified per RFC2326bis).
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* For multicast, "port", "layers", "ttl" are set by the
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* stream output configuration.
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* For unicast, we want to decide "server_port" values.
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* "interleaved" is not implemented.
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const char *dst = id->dst;
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if( psz_session == NULL )
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/* Create a dummy session ID */
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snprintf( psz_sesbuf, sizeof( psz_sesbuf ), "%d",
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psz_session = psz_sesbuf;
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answer->i_status = 200;
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httpd_MsgAdd( answer, "Transport",
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"RTP/AVP/UDP;destination=%s;port=%u-%u;"
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dst, id->loport, id->hiport,
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( id->ttl > 0 ) ? id->ttl : 1 );
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char ip[NI_MAXNUMERICHOST], src[NI_MAXNUMERICHOST];
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rtsp_session_t *ses = NULL;
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rtsp_strack_t track = { id->sout_id, -1, false };
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if( httpd_ClientIP( cl, ip ) == NULL )
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answer->i_status = 500;
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track.fd = net_ConnectDgram( p_stream, ip, loport, -1,
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"cannot create RTP socket for %s port %u",
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answer->i_status = 500;
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net_GetSockAddress( track.fd, src, &sport );
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vlc_mutex_lock( &rtsp->lock );
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if( psz_session == NULL )
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ses = RtspClientNew( rtsp );
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snprintf( psz_sesbuf, sizeof( psz_sesbuf ), "%"PRIx64,
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psz_session = psz_sesbuf;
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/* FIXME: we probably need to remove an access out,
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* if there is already one for the same ID */
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ses = RtspClientGet( rtsp, psz_session );
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answer->i_status = 454;
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vlc_mutex_unlock( &rtsp->lock );
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INSERT_ELEM( ses->trackv, ses->trackc, ses->trackc,
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vlc_mutex_unlock( &rtsp->lock );
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httpd_ServerIP( cl, ip );
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if( strcmp( src, ip ) )
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/* Specify source IP if it is different from the RTSP
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* control connection server address */
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char *ptr = strchr( src, '%' );
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if( ptr != NULL ) *ptr = '\0'; /* remove scope ID */
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httpd_MsgAdd( answer, "Transport",
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"RTP/AVP/UDP;unicast;source=%s;"
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"client_port=%u-%u;server_port=%u-%u;"
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"ssrc=%08X;mode=play",
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src, loport, loport + 1, sport,
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sport + 1, id->ssrc );
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httpd_MsgAdd( answer, "Transport",
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"RTP/AVP/UDP;unicast;"
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"client_port=%u-%u;server_port=%u-%u;"
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"ssrc=%08X;mode=play",
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loport, loport + 1, sport, sport + 1,
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answer->i_status = 200;
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answer->i_status = 200;
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psz_session = httpd_MsgGet( query, "Session" );
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const char *range = httpd_MsgGet (query, "Range");
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if (range && strncmp (range, "npt=", 4))
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answer->i_status = 501;
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vlc_mutex_lock( &rtsp->lock );
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ses = RtspClientGet( rtsp, psz_session );
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/* FIXME: we really need to limit the number of tracks... */
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char info[ses->trackc * ( strlen( control )
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+ sizeof("/trackID=123;seq=65535, ") ) + 1];
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for( int i = 0; i < ses->trackc; i++ )
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rtsp_strack_t *tr = ses->trackv + i;
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if( ( id == NULL ) || ( tr->id == id->sout_id ) )
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rtp_add_sink( tr->id, tr->fd, false );
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infolen += sprintf( info + infolen,
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"%s/trackID=%u;seq=%u, ", control,
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rtp_get_num( tr->id ),
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rtp_get_seq( tr->id ) );
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info[infolen - 2] = '\0'; /* remove trailing ", " */
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httpd_MsgAdd( answer, "RTP-Info", "%s", info );
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vlc_mutex_unlock( &rtsp->lock );
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if( httpd_MsgGet( query, "Scale" ) != NULL )
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httpd_MsgAdd( answer, "Scale", "1." );
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case HTTPD_MSG_PAUSE:
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answer->i_status = 405;
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httpd_MsgAdd( answer, "Allow",
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"%s, TEARDOWN, PLAY, GET_PARAMETER",
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( id != NULL ) ? "SETUP" : "DESCRIBE" );
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case HTTPD_MSG_GETPARAMETER:
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if( query->i_body > 0 )
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answer->i_status = 451;
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psz_session = httpd_MsgGet( query, "Session" );
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answer->i_status = 200;
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case HTTPD_MSG_TEARDOWN:
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answer->i_status = 200;
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psz_session = httpd_MsgGet( query, "Session" );
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vlc_mutex_lock( &rtsp->lock );
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ses = RtspClientGet( rtsp, psz_session );
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if( id == NULL ) /* Delete the entire session */
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RtspClientDel( rtsp, ses );
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else /* Delete one track from the session */
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for( int i = 0; i < ses->trackc; i++ )
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if( ses->trackv[i].id == id->sout_id )
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rtp_del_sink( id->sout_id, ses->trackv[i].fd );
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REMOVE_ELEM( ses->trackv, ses->trackc, i );
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vlc_mutex_unlock( &rtsp->lock );
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httpd_MsgAdd( answer, "Session", "%s"/*;timeout=5*/, psz_session );
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httpd_MsgAdd( answer, "Content-Length", "%d", answer->i_body );
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httpd_MsgAdd( answer, "Cache-Control", "no-cache" );
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psz = httpd_MsgGet( query, "Cseq" );
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httpd_MsgAdd( answer, "Cseq", "%s", psz );
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psz = httpd_MsgGet( query, "Timestamp" );
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httpd_MsgAdd( answer, "Timestamp", "%s", psz );
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/** Aggregate RTSP callback */
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static int RtspCallback( httpd_callback_sys_t *p_args,
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httpd_message_t *answer,
741
const httpd_message_t *query )
743
return RtspHandler( (rtsp_stream_t *)p_args, NULL, cl, answer, query );
747
/** Non-aggregate RTSP callback */
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static int RtspCallbackId( httpd_callback_sys_t *p_args,
750
httpd_message_t *answer,
751
const httpd_message_t *query )
753
rtsp_stream_id_t *id = (rtsp_stream_id_t *)p_args;
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return RtspHandler( id->stream, id, cl, answer, query );