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FFmpeg/libavformat/rtsp.h
Martin Storsjö b56fc18b20 sdp: Add an option for sending RTCP packets to the source of the last packets 
An SDP description normally only contains the target IP address
and port for the packets. This means that we don't really have
any clue where to send the RTCP RR packets - previously they're
sent to the destination IP written in the SDP (at the same port),
which rarely is the actual peer. And if the source for the packets
is on a different port than the destination, it's never correct.

With a new option, we can choose to send the packets to the
address that the latest packet on each socket arrived from.
---
Some may even argue that this should be the default - perhaps,
but I'd rather keep it optional at first. Additionally, I'm not
sure if sending RTCP RR directly back to the source is
desireable for e.g. multicast.

Signed-off-by: Martin Storsjö <martin@martin.st>
2013-08-14 11:21:33 +03:00

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/*
* RTSP definitions
* Copyright (c) 2002 Fabrice Bellard
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFORMAT_RTSP_H
#define AVFORMAT_RTSP_H
#include <stdint.h>
#include "avformat.h"
#include "rtspcodes.h"
#include "rtpdec.h"
#include "network.h"
#include "httpauth.h"
#include "libavutil/log.h"
#include "libavutil/opt.h"
/**
* Network layer over which RTP/etc packet data will be transported.
*/
enum RTSPLowerTransport {
RTSP_LOWER_TRANSPORT_UDP = 0, /**< UDP/unicast */
RTSP_LOWER_TRANSPORT_TCP = 1, /**< TCP; interleaved in RTSP */
RTSP_LOWER_TRANSPORT_UDP_MULTICAST = 2, /**< UDP/multicast */
RTSP_LOWER_TRANSPORT_NB,
RTSP_LOWER_TRANSPORT_HTTP = 8, /**< HTTP tunneled - not a proper
transport mode as such,
only for use via AVOptions */
RTSP_LOWER_TRANSPORT_CUSTOM = 16, /**< Custom IO - not a public
option for lower_transport_mask,
but set in the SDP demuxer based
on a flag. */
};
/**
* Packet profile of the data that we will be receiving. Real servers
* commonly send RDT (although they can sometimes send RTP as well),
* whereas most others will send RTP.
*/
enum RTSPTransport {
RTSP_TRANSPORT_RTP, /**< Standards-compliant RTP */
RTSP_TRANSPORT_RDT, /**< Realmedia Data Transport */
RTSP_TRANSPORT_RAW, /**< Raw data (over UDP) */
RTSP_TRANSPORT_NB
};
/**
* Transport mode for the RTSP data. This may be plain, or
* tunneled, which is done over HTTP.
*/
enum RTSPControlTransport {
RTSP_MODE_PLAIN, /**< Normal RTSP */
RTSP_MODE_TUNNEL /**< RTSP over HTTP (tunneling) */
};
#define RTSP_DEFAULT_PORT 554
#define RTSP_MAX_TRANSPORTS 8
#define RTSP_TCP_MAX_PACKET_SIZE 1472
#define RTSP_DEFAULT_NB_AUDIO_CHANNELS 1
#define RTSP_DEFAULT_AUDIO_SAMPLERATE 44100
#define RTSP_RTP_PORT_MIN 5000
#define RTSP_RTP_PORT_MAX 10000
/**
* This describes a single item in the "Transport:" line of one stream as
* negotiated by the SETUP RTSP command. Multiple transports are comma-
* separated ("Transport: x-read-rdt/tcp;interleaved=0-1,rtp/avp/udp;
* client_port=1000-1001;server_port=1800-1801") and described in separate
* RTSPTransportFields.
*/
typedef struct RTSPTransportField {
/** interleave ids, if TCP transport; each TCP/RTSP data packet starts
* with a '$', stream length and stream ID. If the stream ID is within
* the range of this interleaved_min-max, then the packet belongs to
* this stream. */
int interleaved_min, interleaved_max;
/** UDP multicast port range; the ports to which we should connect to
* receive multicast UDP data. */
int port_min, port_max;
/** UDP client ports; these should be the local ports of the UDP RTP
* (and RTCP) sockets over which we receive RTP/RTCP data. */
int client_port_min, client_port_max;
/** UDP unicast server port range; the ports to which we should connect
* to receive unicast UDP RTP/RTCP data. */
int server_port_min, server_port_max;
/** time-to-live value (required for multicast); the amount of HOPs that
* packets will be allowed to make before being discarded. */
int ttl;
/** transport set to record data */
int mode_record;
struct sockaddr_storage destination; /**< destination IP address */
char source[INET6_ADDRSTRLEN + 1]; /**< source IP address */
/** data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
/** network layer transport protocol; e.g. TCP or UDP uni-/multicast */
enum RTSPLowerTransport lower_transport;
} RTSPTransportField;
/**
* This describes the server response to each RTSP command.
*/
typedef struct RTSPMessageHeader {
/** length of the data following this header */
int content_length;
enum RTSPStatusCode status_code; /**< response code from server */
/** number of items in the 'transports' variable below */
int nb_transports;
/** Time range of the streams that the server will stream. In
* AV_TIME_BASE unit, AV_NOPTS_VALUE if not used */
int64_t range_start, range_end;
/** describes the complete "Transport:" line of the server in response
* to a SETUP RTSP command by the client */
RTSPTransportField transports[RTSP_MAX_TRANSPORTS];
int seq; /**< sequence number */
/** the "Session:" field. This value is initially set by the server and
* should be re-transmitted by the client in every RTSP command. */
char session_id[512];
/** the "Location:" field. This value is used to handle redirection.
*/
char location[4096];
/** the "RealChallenge1:" field from the server */
char real_challenge[64];
/** the "Server: field, which can be used to identify some special-case
* servers that are not 100% standards-compliant. We use this to identify
* Windows Media Server, which has a value "WMServer/v.e.r.sion", where
* version is a sequence of digits (e.g. 9.0.0.3372). Helix/Real servers
* use something like "Helix [..] Server Version v.e.r.sion (platform)
* (RealServer compatible)" or "RealServer Version v.e.r.sion (platform)",
* where platform is the output of $uname -msr | sed 's/ /-/g'. */
char server[64];
/** The "timeout" comes as part of the server response to the "SETUP"
* command, in the "Session: <xyz>[;timeout=<value>]" line. It is the
* time, in seconds, that the server will go without traffic over the
* RTSP/TCP connection before it closes the connection. To prevent
* this, sent dummy requests (e.g. OPTIONS) with intervals smaller
* than this value. */
int timeout;
/** The "Notice" or "X-Notice" field value. See
* http://tools.ietf.org/html/draft-stiemerling-rtsp-announce-00
* for a complete list of supported values. */
int notice;
/** The "reason" is meant to specify better the meaning of the error code
* returned
*/
char reason[256];
/**
* Content type header
*/
char content_type[64];
} RTSPMessageHeader;
/**
* Client state, i.e. whether we are currently receiving data (PLAYING) or
* setup-but-not-receiving (PAUSED). State can be changed in applications
* by calling av_read_play/pause().
*/
enum RTSPClientState {
RTSP_STATE_IDLE, /**< not initialized */
RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */
RTSP_STATE_PAUSED, /**< initialized, but not receiving data */
RTSP_STATE_SEEKING, /**< initialized, requesting a seek */
};
/**
* Identify particular servers that require special handling, such as
* standards-incompliant "Transport:" lines in the SETUP request.
*/
enum RTSPServerType {
RTSP_SERVER_RTP, /**< Standards-compliant RTP-server */
RTSP_SERVER_REAL, /**< Realmedia-style server */
RTSP_SERVER_WMS, /**< Windows Media server */
RTSP_SERVER_NB
};
/**
* Private data for the RTSP demuxer.
*
* @todo Use AVIOContext instead of URLContext
*/
typedef struct RTSPState {
const AVClass *class; /**< Class for private options. */
URLContext *rtsp_hd; /* RTSP TCP connection handle */
/** number of items in the 'rtsp_streams' variable */
int nb_rtsp_streams;
struct RTSPStream **rtsp_streams; /**< streams in this session */
/** indicator of whether we are currently receiving data from the
* server. Basically this isn't more than a simple cache of the
* last PLAY/PAUSE command sent to the server, to make sure we don't
* send 2x the same unexpectedly or commands in the wrong state. */
enum RTSPClientState state;
/** the seek value requested when calling av_seek_frame(). This value
* is subsequently used as part of the "Range" parameter when emitting
* the RTSP PLAY command. If we are currently playing, this command is
* called instantly. If we are currently paused, this command is called
* whenever we resume playback. Either way, the value is only used once,
* see rtsp_read_play() and rtsp_read_seek(). */
int64_t seek_timestamp;
int seq; /**< RTSP command sequence number */
/** copy of RTSPMessageHeader->session_id, i.e. the server-provided session
* identifier that the client should re-transmit in each RTSP command */
char session_id[512];
/** copy of RTSPMessageHeader->timeout, i.e. the time (in seconds) that
* the server will go without traffic on the RTSP/TCP line before it
* closes the connection. */
int timeout;
/** timestamp of the last RTSP command that we sent to the RTSP server.
* This is used to calculate when to send dummy commands to keep the
* connection alive, in conjunction with timeout. */
int64_t last_cmd_time;
/** the negotiated data/packet transport protocol; e.g. RTP or RDT */
enum RTSPTransport transport;
/** the negotiated network layer transport protocol; e.g. TCP or UDP
* uni-/multicast */
enum RTSPLowerTransport lower_transport;
/** brand of server that we're talking to; e.g. WMS, REAL or other.
* Detected based on the value of RTSPMessageHeader->server or the presence
* of RTSPMessageHeader->real_challenge */
enum RTSPServerType server_type;
/** the "RealChallenge1:" field from the server */
char real_challenge[64];
/** plaintext authorization line (username:password) */
char auth[128];
/** authentication state */
HTTPAuthState auth_state;
/** The last reply of the server to a RTSP command */
char last_reply[2048]; /* XXX: allocate ? */
/** RTSPStream->transport_priv of the last stream that we read a
* packet from */
void *cur_transport_priv;
/** The following are used for Real stream selection */
//@{
/** whether we need to send a "SET_PARAMETER Subscribe:" command */
int need_subscription;
/** stream setup during the last frame read. This is used to detect if
* we need to subscribe or unsubscribe to any new streams. */
enum AVDiscard *real_setup_cache;
/** current stream setup. This is a temporary buffer used to compare
* current setup to previous frame setup. */
enum AVDiscard *real_setup;
/** the last value of the "SET_PARAMETER Subscribe:" RTSP command.
* this is used to send the same "Unsubscribe:" if stream setup changed,
* before sending a new "Subscribe:" command. */
char last_subscription[1024];
//@}
/** The following are used for RTP/ASF streams */
//@{
/** ASF demuxer context for the embedded ASF stream from WMS servers */
AVFormatContext *asf_ctx;
/** cache for position of the asf demuxer, since we load a new
* data packet in the bytecontext for each incoming RTSP packet. */
uint64_t asf_pb_pos;
//@}
/** some MS RTSP streams contain a URL in the SDP that we need to use
* for all subsequent RTSP requests, rather than the input URI; in
* other cases, this is a copy of AVFormatContext->filename. */
char control_uri[1024];
/** The following are used for parsing raw mpegts in udp */
//@{
struct MpegTSContext *ts;
int recvbuf_pos;
int recvbuf_len;
//@}
/** Additional output handle, used when input and output are done
* separately, eg for HTTP tunneling. */
URLContext *rtsp_hd_out;
/** RTSP transport mode, such as plain or tunneled. */
enum RTSPControlTransport control_transport;
/* Number of RTCP BYE packets the RTSP session has received.
* An EOF is propagated back if nb_byes == nb_streams.
* This is reset after a seek. */
int nb_byes;
/** Reusable buffer for receiving packets */
uint8_t* recvbuf;
/**
* A mask with all requested transport methods
*/
int lower_transport_mask;
/**
* The number of returned packets
*/
uint64_t packets;
/**
* Polling array for udp
*/
struct pollfd *p;
/**
* Whether the server supports the GET_PARAMETER method.
*/
int get_parameter_supported;
/**
* Do not begin to play the stream immediately.
*/
int initial_pause;
/**
* Option flags for the chained RTP muxer.
*/
int rtp_muxer_flags;
/** Whether the server accepts the x-Dynamic-Rate header */
int accept_dynamic_rate;
/**
* Various option flags for the RTSP muxer/demuxer.
*/
int rtsp_flags;
/**
* Mask of all requested media types
*/
int media_type_mask;
/**
* Minimum and maximum local UDP ports.
*/
int rtp_port_min, rtp_port_max;
/**
* Timeout to wait for incoming connections.
*/
int initial_timeout;
/**
* Size of RTP packet reordering queue.
*/
int reordering_queue_size;
} RTSPState;
#define RTSP_FLAG_FILTER_SRC 0x1 /**< Filter incoming UDP packets -
receive packets only from the right
source address and port. */
#define RTSP_FLAG_LISTEN 0x2 /**< Wait for incoming connections. */
#define RTSP_FLAG_CUSTOM_IO 0x4 /**< Do all IO via the AVIOContext. */
#define RTSP_FLAG_RTCP_TO_SOURCE 0x8 /**< Send RTCP packets to the source
address of received packets. */
typedef struct RTSPSource {
char addr[128]; /**< Source-specific multicast include source IP address (from SDP content) */
} RTSPSource;
/**
* Describe a single stream, as identified by a single m= line block in the
* SDP content. In the case of RDT, one RTSPStream can represent multiple
* AVStreams. In this case, each AVStream in this set has similar content
* (but different codec/bitrate).
*/
typedef struct RTSPStream {
URLContext *rtp_handle; /**< RTP stream handle (if UDP) */
void *transport_priv; /**< RTP/RDT parse context if input, RTP AVFormatContext if output */
/** corresponding stream index, if any. -1 if none (MPEG2TS case) */
int stream_index;
/** interleave IDs; copies of RTSPTransportField->interleaved_min/max
* for the selected transport. Only used for TCP. */
int interleaved_min, interleaved_max;
char control_url[1024]; /**< url for this stream (from SDP) */
/** The following are used only in SDP, not RTSP */
//@{
int sdp_port; /**< port (from SDP content) */
struct sockaddr_storage sdp_ip; /**< IP address (from SDP content) */
int nb_include_source_addrs; /**< Number of source-specific multicast include source IP addresses (from SDP content) */
struct RTSPSource **include_source_addrs; /**< Source-specific multicast include source IP addresses (from SDP content) */
int nb_exclude_source_addrs; /**< Number of source-specific multicast exclude source IP addresses (from SDP content) */
struct RTSPSource **exclude_source_addrs; /**< Source-specific multicast exclude source IP addresses (from SDP content) */
int sdp_ttl; /**< IP Time-To-Live (from SDP content) */
int sdp_payload_type; /**< payload type */
//@}
/** The following are used for dynamic protocols (rtpdec_*.c/rdt.c) */
//@{
/** handler structure */
RTPDynamicProtocolHandler *dynamic_handler;
/** private data associated with the dynamic protocol */
PayloadContext *dynamic_protocol_context;
//@}
/** Enable sending RTCP feedback messages according to RFC 4585 */
int feedback;
char crypto_suite[40];
char crypto_params[100];
} RTSPStream;
void ff_rtsp_parse_line(RTSPMessageHeader *reply, const char *buf,
RTSPState *rt, const char *method);
/**
* Send a command to the RTSP server without waiting for the reply.
*
* @see rtsp_send_cmd_with_content_async
*/
int ff_rtsp_send_cmd_async(AVFormatContext *s, const char *method,
const char *url, const char *headers);
/**
* Send a command to the RTSP server and wait for the reply.
*
* @param s RTSP (de)muxer context
* @param method the method for the request
* @param url the target url for the request
* @param headers extra header lines to include in the request
* @param reply pointer where the RTSP message header will be stored
* @param content_ptr pointer where the RTSP message body, if any, will
* be stored (length is in reply)
* @param send_content if non-null, the data to send as request body content
* @param send_content_length the length of the send_content data, or 0 if
* send_content is null
*
* @return zero if success, nonzero otherwise
*/
int ff_rtsp_send_cmd_with_content(AVFormatContext *s,
const char *method, const char *url,
const char *headers,
RTSPMessageHeader *reply,
unsigned char **content_ptr,
const unsigned char *send_content,
int send_content_length);
/**
* Send a command to the RTSP server and wait for the reply.
*
* @see rtsp_send_cmd_with_content
*/
int ff_rtsp_send_cmd(AVFormatContext *s, const char *method,
const char *url, const char *headers,
RTSPMessageHeader *reply, unsigned char **content_ptr);
/**
* Read a RTSP message from the server, or prepare to read data
* packets if we're reading data interleaved over the TCP/RTSP
* connection as well.
*
* @param s RTSP (de)muxer context
* @param reply pointer where the RTSP message header will be stored
* @param content_ptr pointer where the RTSP message body, if any, will
* be stored (length is in reply)
* @param return_on_interleaved_data whether the function may return if we
* encounter a data marker ('$'), which precedes data
* packets over interleaved TCP/RTSP connections. If this
* is set, this function will return 1 after encountering
* a '$'. If it is not set, the function will skip any
* data packets (if they are encountered), until a reply
* has been fully parsed. If no more data is available
* without parsing a reply, it will return an error.
* @param method the RTSP method this is a reply to. This affects how
* some response headers are acted upon. May be NULL.
*
* @return 1 if a data packets is ready to be received, -1 on error,
* and 0 on success.
*/
int ff_rtsp_read_reply(AVFormatContext *s, RTSPMessageHeader *reply,
unsigned char **content_ptr,
int return_on_interleaved_data, const char *method);
/**
* Skip a RTP/TCP interleaved packet.
*/
void ff_rtsp_skip_packet(AVFormatContext *s);
/**
* Connect to the RTSP server and set up the individual media streams.
* This can be used for both muxers and demuxers.
*
* @param s RTSP (de)muxer context
*
* @return 0 on success, < 0 on error. Cleans up all allocations done
* within the function on error.
*/
int ff_rtsp_connect(AVFormatContext *s);
/**
* Close and free all streams within the RTSP (de)muxer
*
* @param s RTSP (de)muxer context
*/
void ff_rtsp_close_streams(AVFormatContext *s);
/**
* Close all connection handles within the RTSP (de)muxer
*
* @param s RTSP (de)muxer context
*/
void ff_rtsp_close_connections(AVFormatContext *s);
/**
* Get the description of the stream and set up the RTSPStream child
* objects.
*/
int ff_rtsp_setup_input_streams(AVFormatContext *s, RTSPMessageHeader *reply);
/**
* Announce the stream to the server and set up the RTSPStream child
* objects for each media stream.
*/
int ff_rtsp_setup_output_streams(AVFormatContext *s, const char *addr);
/**
* Parse RTSP commands (OPTIONS, PAUSE and TEARDOWN) during streaming in
* listen mode.
*/
int ff_rtsp_parse_streaming_commands(AVFormatContext *s);
/**
* Parse an SDP description of streams by populating an RTSPState struct
* within the AVFormatContext; also allocate the RTP streams and the
* pollfd array used for UDP streams.
*/
int ff_sdp_parse(AVFormatContext *s, const char *content);
/**
* Receive one RTP packet from an TCP interleaved RTSP stream.
*/
int ff_rtsp_tcp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
uint8_t *buf, int buf_size);
/**
* Receive one packet from the RTSPStreams set up in the AVFormatContext
* (which should contain a RTSPState struct as priv_data).
*/
int ff_rtsp_fetch_packet(AVFormatContext *s, AVPacket *pkt);
/**
* Do the SETUP requests for each stream for the chosen
* lower transport mode.
* @return 0 on success, <0 on error, 1 if protocol is unavailable
*/
int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
int lower_transport, const char *real_challenge);
/**
* Undo the effect of ff_rtsp_make_setup_request, close the
* transport_priv and rtp_handle fields.
*/
void ff_rtsp_undo_setup(AVFormatContext *s);
/**
* Open RTSP transport context.
*/
int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st);
extern const AVOption ff_rtsp_options[];
#endif /* AVFORMAT_RTSP_H */
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