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rtp.h
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rtp.h
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/*! \file rtp.h
* \author Lorenzo Miniero <[email protected]>
* \copyright GNU General Public License v3
* \brief RTP processing (headers)
* \details Implementation of the RTP header. Since the gateway does not
* much more than relaying frames around, the only thing we're interested
* in is the RTP header and how to get its payload, and parsing extensions.
*
* \ingroup protocols
* \ref protocols
*/
#ifndef _JANUS_RTP_H
#define _JANUS_RTP_H
#include <arpa/inet.h>
#ifdef __MACH__
#include <machine/endian.h>
#define __BYTE_ORDER BYTE_ORDER
#define __BIG_ENDIAN BIG_ENDIAN
#define __LITTLE_ENDIAN LITTLE_ENDIAN
#else
#include <endian.h>
#endif
#include <inttypes.h>
#include <string.h>
#include <glib.h>
#ifdef HAVE_SRTP_2
#include <srtp2/srtp.h>
#include <openssl/rand.h>
#include <openssl/err.h>
int srtp_crypto_get_random(uint8_t *key, int len);
#else
#include <srtp/srtp.h>
#include <srtp/crypto_kernel.h>
#define srtp_err_status_t err_status_t
#define srtp_err_status_ok err_status_ok
#define srtp_err_status_replay_fail err_status_replay_fail
#define srtp_err_status_replay_old err_status_replay_old
#define srtp_crypto_policy_set_rtp_default crypto_policy_set_rtp_default
#define srtp_crypto_policy_set_rtcp_default crypto_policy_set_rtcp_default
#define srtp_crypto_policy_set_aes_cm_128_hmac_sha1_32 crypto_policy_set_aes_cm_128_hmac_sha1_32
#define srtp_crypto_policy_set_aes_cm_128_hmac_sha1_80 crypto_policy_set_aes_cm_128_hmac_sha1_80
#define srtp_crypto_get_random crypto_get_random
#endif
#define RTP_HEADER_SIZE 12
/* SRTP stuff (http://tools.ietf.org/html/rfc3711) */
#define SRTP_MASTER_KEY_LENGTH 16
#define SRTP_MASTER_SALT_LENGTH 14
#define SRTP_MASTER_LENGTH (SRTP_MASTER_KEY_LENGTH + SRTP_MASTER_SALT_LENGTH)
/*! \brief RTP Header (http://tools.ietf.org/html/rfc3550#section-5.1) */
typedef struct rtp_header
{
#if __BYTE_ORDER == __BIG_ENDIAN
uint16_t version:2;
uint16_t padding:1;
uint16_t extension:1;
uint16_t csrccount:4;
uint16_t markerbit:1;
uint16_t type:7;
#elif __BYTE_ORDER == __LITTLE_ENDIAN
uint16_t csrccount:4;
uint16_t extension:1;
uint16_t padding:1;
uint16_t version:2;
uint16_t type:7;
uint16_t markerbit:1;
#endif
uint16_t seq_number;
uint32_t timestamp;
uint32_t ssrc;
uint32_t csrc[16];
} rtp_header;
/*! \brief RTP packet */
typedef struct janus_rtp_packet {
char *data;
gint length;
gint64 created;
gint64 last_retransmit;
} janus_rtp_packet;
/*! \brief RTP extension */
typedef struct janus_rtp_header_extension {
uint16_t type;
uint16_t length;
} janus_rtp_header_extension;
/*! \brief a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level */
#define JANUS_RTP_EXTMAP_AUDIO_LEVEL "urn:ietf:params:rtp-hdrext:ssrc-audio-level"
/*! \brief a=extmap:2 urn:ietf:params:rtp-hdrext:toffset */
#define JANUS_RTP_EXTMAP_TOFFSET "urn:ietf:params:rtp-hdrext:toffset"
/*! \brief a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time */
#define JANUS_RTP_EXTMAP_ABS_SEND_TIME "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
/*! \brief a=extmap:4 urn:3gpp:video-orientation */
#define JANUS_RTP_EXTMAP_VIDEO_ORIENTATION "urn:3gpp:video-orientation"
/*! \brief a=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 */
#define JANUS_RTP_EXTMAP_CC_EXTENSIONS "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
/*! \brief a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay */
#define JANUS_RTP_EXTMAP_PLAYOUT_DELAY "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
/*! \brief Helper to quickly access the RTP payload, skipping header and extensions
* @param[in] buf The packet data
* @param[in] len The packet data length in bytes
* @param[out] plen The payload data length in bytes
* @returns A pointer to where the payload data starts, or NULL otherwise; plen is also set accordingly */
char *janus_rtp_payload(char *buf, int len, int *plen);
/*! \brief Ugly and dirty helper to quickly get the id associated with an RTP extension (extmap) in an SDP
* @param sdp The SDP to parse
* @param extension The extension namespace to look for
* @returns The extension id, if found, -1 otherwise */
int janus_rtp_header_extension_get_id(const char *sdp, const char *extension);
/*! \brief Ugly and dirty helper to quickly get the RTP extension namespace associated with an id (extmap) in an SDP
* @note This only looks for the extensions we know about, those defined in rtp.h
* @param sdp The SDP to parse
* @param id The extension id to look for
* @returns The extension namespace, if found, NULL otherwise */
const char *janus_rtp_header_extension_get_from_id(const char *sdp, int id);
/*! \brief Helper to parse a ssrc-audio-level RTP extension (https://tools.ietf.org/html/rfc6464)
* @param[in] buf The packet data
* @param[in] len The packet data length in bytes
* @param[in] id The extension ID to look for
* @param[out] level The level value in dBov (0=max, 127=min)
* @returns 0 if found, -1 otherwise */
int janus_rtp_header_extension_parse_audio_level(char *buf, int len, int id, int *level);
/*! \brief Helper to parse a video-orientation RTP extension (http://www.3gpp.org/ftp/Specs/html-info/26114.htm)
* @param[in] buf The packet data
* @param[in] len The packet data length in bytes
* @param[in] id The extension ID to look for
* @param[out] c The value of the Camera (C) bit
* @param[out] f The value of the Flip (F) bit
* @param[out] r1 The value of the first Rotation (R1) bit
* @param[out] r0 The value of the second Rotation (R0) bit
* @returns 0 if found, -1 otherwise */
int janus_rtp_header_extension_parse_video_orientation(char *buf, int len, int id,
gboolean *c, gboolean *f, gboolean *r1, gboolean *r0);
/*! \brief Helper to parse a playout-delay RTP extension (https://webrtc.org/experiments/rtp-hdrext/playout-delay)
* @param[in] buf The packet data
* @param[in] len The packet data length in bytes
* @param[in] id The extension ID to look for
* @param[out] min_delay The minimum delay value
* @param[out] max_delay The maximum delay value
* @returns 0 if found, -1 otherwise */
int janus_rtp_header_extension_parse_playout_delay(char *buf, int len, int id,
uint16_t *min_delay, uint16_t *max_delay);
/*! \brief RTP context, in order to make sure SSRC changes result in coherent seq/ts increases */
typedef struct janus_rtp_switching_context {
uint32_t a_last_ssrc, a_last_ts, a_base_ts, a_base_ts_prev,
v_last_ssrc, v_last_ts, v_base_ts, v_base_ts_prev;
uint16_t a_last_seq, a_base_seq, a_base_seq_prev,
v_last_seq, v_base_seq, v_base_seq_prev;
gboolean a_seq_reset, v_seq_reset;
} janus_rtp_switching_context;
/*! \brief Set (or reset) the context fields to their default values
* @param[in] context The context to (re)set */
void janus_rtp_switching_context_reset(janus_rtp_switching_context *context);
/*! \brief Use the context info to update the RTP header of a packet, if needed
* @param[in] header The RTP header to update
* @param[in] context The context to use as a reference
* @param[in] video Whether this is an audio or a video packet
* @param[in] step The expected timestamp step */
void janus_rtp_header_update(rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step);
/*! \brief Helper method to get a string representation of a libsrtp error code
* @param[in] error The libsrtp error code
* @returns A string representation of the error code */
const char *janus_srtp_error_str(int error);
#endif