/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2009, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 * Joshua Colp <jcolp@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 * \brief Pluggable RTP Architecture
 * \author Joshua Colp <jcolp@digium.com>
 * \ref AstRTPEngine
 */

/*!
 * \page AstRTPEngine Asterisk RTP Engine API
 *
 * The purpose of this API is to provide a way for multiple RTP stacks to be
 * used inside of Asterisk without any module that uses RTP knowing any
 * different. To the module each RTP stack behaves the same.
 *
 * An RTP session is called an instance and is made up of a combination of codec
 * information, RTP engine, RTP properties, and address information. An engine
 * name may be passed in to explicitly choose an RTP stack to be used but a
 * default one will be used if none is provided. An address to use for RTP may
 * also be provided but the underlying RTP engine may choose a different address
 * depending on it's configuration.
 *
 * An RTP engine is the layer between the RTP engine core and the RTP stack
 * itself. The RTP engine core provides a set of callbacks to do various things
 * (such as write audio out) that the RTP engine has to have implemented.
 *
 * Glue is what binds an RTP instance to a channel. It is used to retrieve RTP
 * instance information when performing remote or local bridging and is used to
 * have the channel driver tell the remote side to change destination of the RTP
 * stream.
 *
 * Statistics from an RTP instance can be retrieved using the
 * ast_rtp_instance_get_stats API call. This essentially asks the RTP engine in
 * use to fill in a structure with the requested values. It is not required for
 * an RTP engine to support all statistic values.
 *
 * Properties allow behavior of the RTP engine and RTP engine core to be
 * changed. For example, there is a property named AST_RTP_PROPERTY_NAT which is
 * used to tell the RTP engine to enable symmetric RTP if it supports it. It is
 * not required for an RTP engine to support all properties.
 *
 * Codec information is stored using a separate data structure which has it's
 * own set of API calls to add/remove/retrieve information. They are used by the
 * module after an RTP instance is created so that payload information is
 * available for the RTP engine.
 */

#ifndef _ASTERISK_RTP_ENGINE_H
#define _ASTERISK_RTP_ENGINE_H

#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
#endif

#include "asterisk/astobj2.h"
#include "asterisk/frame.h"
#include "asterisk/format_cap.h"
#include "asterisk/netsock2.h"
#include "asterisk/sched.h"
#include "asterisk/res_srtp.h"
#include "asterisk/stasis.h"
#include "asterisk/vector.h"
#include "asterisk/logger_category.h"

/*! Maximum number of payload types RTP can support. */
#define AST_RTP_MAX_PT 128

/*!
 * Last RTP payload type statically assigned, see
 * http://www.iana.org/assignments/rtp-parameters
 */
#define AST_RTP_PT_LAST_STATIC 34

/*! First dynamic RTP payload type */
#define AST_RTP_PT_FIRST_DYNAMIC 96

/*! Last reassignable RTP payload type */
#define AST_RTP_PT_LAST_REASSIGN 63

/*! Maximum number of generations */
#define AST_RED_MAX_GENERATION 5

/*!
 * Maximum size of an internal Asterisk channel unique ID.
 *
 * \note Must match the AST_MAX_UNIQUEID(AST_MAX_PUBLIC_UNIQUEID) value.
 * We don't use that defined value directly here to avoid a hard
 * dependency on channel.h.
 */
#define MAX_CHANNEL_ID 152

/*!< DTMF samples per second */
#define DEFAULT_DTMF_SAMPLE_RATE_MS    8000

struct ast_rtp_instance;
struct ast_rtp_glue;

/*! RTP Properties that can be set on an RTP instance */
enum ast_rtp_property {
	/*! Enable symmetric RTP support */
	AST_RTP_PROPERTY_NAT = 0,
	/*! RTP instance will be carrying DTMF (using RFC2833) */
	AST_RTP_PROPERTY_DTMF,
	/*! Expect unreliable DTMF from remote party */
	AST_RTP_PROPERTY_DTMF_COMPENSATE,
	/*! Enable STUN support */
	AST_RTP_PROPERTY_STUN,
	/*! Enable RTCP support */
	AST_RTP_PROPERTY_RTCP,
	/*! Enable Asymmetric RTP Codecs */
	AST_RTP_PROPERTY_ASYMMETRIC_CODEC,
	/*! Enable packet retransmission for received packets */
	AST_RTP_PROPERTY_RETRANS_RECV,
	/*! Enable packet retransmission for sent packets */
	AST_RTP_PROPERTY_RETRANS_SEND,
	/*! Enable REMB sending and receiving passthrough support */
	AST_RTP_PROPERTY_REMB,

	/*!
	 * \brief Maximum number of RTP properties supported
	 *
	 * \note THIS MUST BE THE LAST ENTRY IN THIS ENUM.
	 */
	AST_RTP_PROPERTY_MAX,
};

/*! Additional RTP options */
enum ast_rtp_options {
	/*! Remote side is using non-standard G.726 */
	AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
};

/*! RTP DTMF Modes */
enum ast_rtp_dtmf_mode {
	/*! No DTMF is being carried over the RTP stream */
	AST_RTP_DTMF_MODE_NONE = 0,
	/*! DTMF is being carried out of band using RFC2833 */
	AST_RTP_DTMF_MODE_RFC2833,
	/*! DTMF is being carried inband over the RTP stream */
	AST_RTP_DTMF_MODE_INBAND,
};

/*! Result codes when RTP glue is queried for information */
enum ast_rtp_glue_result {
	/*! No remote or local bridging is permitted */
	AST_RTP_GLUE_RESULT_FORBID = 0,
	/*! Move RTP stream to be remote between devices directly */
	AST_RTP_GLUE_RESULT_REMOTE,
	/*! Perform RTP engine level bridging if possible */
	AST_RTP_GLUE_RESULT_LOCAL,
};

/*! Field statistics that can be retrieved from an RTP instance */
enum ast_rtp_instance_stat_field {
	/*! Retrieve quality information */
	AST_RTP_INSTANCE_STAT_FIELD_QUALITY = 0,
	/*! Retrieve quality information about jitter */
	AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER,
	/*! Retrieve quality information about packet loss */
	AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS,
	/*! Retrieve quality information about round trip time */
	AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT,
	/*! Retrieve quality information about Media Experience Score */
	AST_RTP_INSTANCE_STAT_FIELD_QUALITY_MES,
};

/*! Statistics that can be retrieved from an RTP instance */
enum ast_rtp_instance_stat {
	/*! Retrieve all statistics */
	AST_RTP_INSTANCE_STAT_ALL = 0,
	/*! Retrieve number of packets transmitted */
	AST_RTP_INSTANCE_STAT_TXCOUNT,
	/*! Retrieve number of packets received */
	AST_RTP_INSTANCE_STAT_RXCOUNT,
	/*! Retrieve ALL statistics relating to packet loss */
	AST_RTP_INSTANCE_STAT_COMBINED_LOSS,
	/*! Retrieve number of packets lost for transmitting */
	AST_RTP_INSTANCE_STAT_TXPLOSS,
	/*! Retrieve number of packets lost for receiving */
	AST_RTP_INSTANCE_STAT_RXPLOSS,
	/*! Retrieve maximum number of packets lost on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS,
	/*! Retrieve minimum number of packets lost on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS,
	/*! Retrieve average number of packets lost on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS,
	/*! Retrieve standard deviation of packets lost on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS,
	/*! Retrieve maximum number of packets lost on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS,
	/*! Retrieve minimum number of packets lost on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS,
	/*! Retrieve average number of packets lost on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS,
	/*! Retrieve standard deviation of packets lost on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS,
	/*! Retrieve ALL statistics relating to jitter */
	AST_RTP_INSTANCE_STAT_COMBINED_JITTER,
	/*! Retrieve jitter on transmitted packets */
	AST_RTP_INSTANCE_STAT_TXJITTER,
	/*! Retrieve jitter on received packets */
	AST_RTP_INSTANCE_STAT_RXJITTER,
	/*! Retrieve maximum jitter on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER,
	/*! Retrieve minimum jitter on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER,
	/*! Retrieve average jitter on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER,
	/*! Retrieve standard deviation jitter on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER,
	/*! Retrieve maximum jitter on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER,
	/*! Retrieve minimum jitter on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER,
	/*! Retrieve average jitter on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER,
	/*! Retrieve standard deviation jitter on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER,
	/*! Retrieve ALL statistics relating to round trip time */
	AST_RTP_INSTANCE_STAT_COMBINED_RTT,
	/*! Retrieve round trip time */
	AST_RTP_INSTANCE_STAT_RTT,
	/*! Retrieve maximum round trip time */
	AST_RTP_INSTANCE_STAT_MAX_RTT,
	/*! Retrieve minimum round trip time */
	AST_RTP_INSTANCE_STAT_MIN_RTT,
	/*! Retrieve average round trip time */
	AST_RTP_INSTANCE_STAT_NORMDEVRTT,
	/*! Retrieve standard deviation round trip time */
	AST_RTP_INSTANCE_STAT_STDEVRTT,
	/*! Retrieve local SSRC */
	AST_RTP_INSTANCE_STAT_LOCAL_SSRC,
	/*! Retrieve remote SSRC */
	AST_RTP_INSTANCE_STAT_REMOTE_SSRC,
	/*! Retrieve channel unique ID */
	AST_RTP_INSTANCE_STAT_CHANNEL_UNIQUEID,
	/*! Retrieve number of octets transmitted */
	AST_RTP_INSTANCE_STAT_TXOCTETCOUNT,
	/*! Retrieve number of octets received */
	AST_RTP_INSTANCE_STAT_RXOCTETCOUNT,

	/*! Retrieve ALL statistics relating to Media Experience Score */
	AST_RTP_INSTANCE_STAT_COMBINED_MES,
	/*! Retrieve MES on transmitted packets */
	AST_RTP_INSTANCE_STAT_TXMES,
	/*! Retrieve MES on received packets */
	AST_RTP_INSTANCE_STAT_RXMES,
	/*! Retrieve maximum MES on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_MAXMES,
	/*! Retrieve minimum MES on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_MINMES,
	/*! Retrieve average MES on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVMES,
	/*! Retrieve standard deviation MES on remote side */
	AST_RTP_INSTANCE_STAT_REMOTE_STDEVMES,
	/*! Retrieve maximum MES on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_MAXMES,
	/*! Retrieve minimum MES on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_MINMES,
	/*! Retrieve average MES on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVMES,
	/*! Retrieve standard deviation MES on local side */
	AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES,
};

enum ast_rtp_instance_rtcp {
	/*! RTCP should not be sent/received */
	AST_RTP_INSTANCE_RTCP_DISABLED = 0,
	/*! RTCP should be sent/received based on standard port rules */
	AST_RTP_INSTANCE_RTCP_STANDARD,
	/*! RTCP should be sent/received on the same port as RTP */
	AST_RTP_INSTANCE_RTCP_MUX,
};

/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
/*! DTMF (RFC2833) */
#define AST_RTP_DTMF                    (1 << 0)
/*! 'Comfort Noise' (RFC3389) */
#define AST_RTP_CN                      (1 << 1)
/*! DTMF (Cisco Proprietary) */
#define AST_RTP_CISCO_DTMF              (1 << 2)
/*! Maximum RTP-specific code */
#define AST_RTP_MAX                     AST_RTP_CISCO_DTMF

/*! Structure that represents a payload */
struct ast_rtp_payload_type {
	/*! If asterisk_format is set, this is the internal
	 * asterisk format represented by the payload */
	struct ast_format *format;
	/*! Is this an Asterisk value */
	int asterisk_format;
	/*! Actual internal RTP specific value of the payload */
	int rtp_code;
	/*! Actual payload number */
	int payload;
	/*! TRUE if this is the primary mapping to the format. */
	unsigned int primary_mapping:1;
	/*! When the payload type became non-primary. */
	struct timeval when_retired;
	/*! Sample rate to over-ride mime type defaults */
	unsigned int sample_rate;
};

/* Common RTCP report types */
/*! Sender Report */
#define AST_RTP_RTCP_SR 200
/*! Receiver Report */
#define AST_RTP_RTCP_RR 201
/*! Transport Layer Feed Back (From RFC4585 also RFC5104) */
#define AST_RTP_RTCP_RTPFB	205
/*! Payload Specific Feed Back (From RFC4585 also RFC5104) */
#define AST_RTP_RTCP_PSFB	206

/* Common RTCP feedback message types */
/*! Generic NACK (From RFC4585 also RFC5104) */
#define AST_RTP_RTCP_FMT_NACK	1
/*! Picture loss indication (From RFC4585) */
#define AST_RTP_RTCP_FMT_PLI	1
/*! Full INTRA-frame Request (From RFC5104) */
#define AST_RTP_RTCP_FMT_FIR	4
/*! REMB Information (From draft-alvestrand-rmcat-remb-03) */
#define AST_RTP_RTCP_FMT_REMB	15
/*! Transport-wide congestion control feedback (From draft-holmer-rmcat-transport-wide-cc-extensions-01) */
#define AST_RTP_RTCP_FMT_TRANSPORT_WIDE_CC 15

/*!
 * \since 12
 * \brief A report block within a SR/RR report */
struct ast_rtp_rtcp_report_block {
	unsigned int source_ssrc;         /*< The SSRC of the source for this report block */
	struct {
		unsigned short fraction;      /*< The fraction of packets lost since last SR/RR */
		unsigned int packets;         /*< The cumulative packets since the beginning */
	} lost_count;                     /*< Statistics regarding missed packets */
	unsigned int highest_seq_no;      /*< Extended highest sequence number received */
	unsigned int ia_jitter;           /*< Calculated interarrival jitter */
	unsigned int lsr;                 /*< The time the last SR report was received */
	unsigned int dlsr;                /*< Delay in sending this report */
};

/*!
 * \since 12
 * \brief An object that represents data sent during a SR/RR RTCP report */
struct ast_rtp_rtcp_report {
	unsigned short reception_report_count;     /*< The number of report blocks */
	unsigned int ssrc;                         /*< Our SSRC */
	unsigned int type;                         /*< The type of report. 200=SR; 201=RR */
	struct {
		struct timeval ntp_timestamp;          /*< Our NTP timestamp */
		unsigned int rtp_timestamp;            /*< Our last RTP timestamp */
		unsigned int packet_count;             /*< Number of packets sent */
		unsigned int octet_count;              /*< Number of bytes sent */
	} sender_information;                      /*< Sender information for SR */
	/*! A dynamic array of report blocks. The number of elements is given by
	 * \c reception_report_count.
	 */
	struct ast_rtp_rtcp_report_block *report_block[0];
};

/*!
 * \since 15.4.0
 * \brief A REMB feedback message (see draft-alvestrand-rmcat-remb-03 for details) */
struct ast_rtp_rtcp_feedback_remb {
	unsigned int br_exp;		/*!< Exponential scaling of the mantissa for the maximum total media bit rate value */
	unsigned int br_mantissa;	/*!< The mantissa of the maximum total media bit rate */
};

/*!
 * \since 15.4.0
 * \brief An object that represents data received in a feedback report */
struct ast_rtp_rtcp_feedback {
	unsigned int fmt; /*!< The feedback message type */
	union {
		struct ast_rtp_rtcp_feedback_remb remb; /*!< REMB feedback information */
	};
};

/*! Structure that represents statistics from an RTP instance */
struct ast_rtp_instance_stats {
	/*! Number of packets transmitted */
	unsigned int txcount;
	/*! Number of packets received */
	unsigned int rxcount;
	/*! Jitter on transmitted packets */
	double txjitter;
	/*! Jitter on received packets */
	double rxjitter;
	/*! Maximum jitter on remote side */
	double remote_maxjitter;
	/*! Minimum jitter on remote side */
	double remote_minjitter;
	/*! Average jitter on remote side */
	double remote_normdevjitter;
	/*! Standard deviation jitter on remote side */
	double remote_stdevjitter;
	/*! Maximum jitter on local side */
	double local_maxjitter;
	/*! Minimum jitter on local side */
	double local_minjitter;
	/*! Average jitter on local side */
	double local_normdevjitter;
	/*! Standard deviation jitter on local side */
	double local_stdevjitter;
	/*! Number of transmitted packets lost */
	unsigned int txploss;
	/*! Number of received packets lost */
	unsigned int rxploss;
	/*! Maximum number of packets lost on remote side */
	double remote_maxrxploss;
	/*! Minimum number of packets lost on remote side */
	double remote_minrxploss;
	/*! Average number of packets lost on remote side */
	double remote_normdevrxploss;
	/*! Standard deviation packets lost on remote side */
	double remote_stdevrxploss;
	/*! Maximum number of packets lost on local side */
	double local_maxrxploss;
	/*! Minimum number of packets lost on local side */
	double local_minrxploss;
	/*! Average number of packets lost on local side */
	double local_normdevrxploss;
	/*! Standard deviation packets lost on local side */
	double local_stdevrxploss;
	/*! Total round trip time */
	double rtt;
	/*! Maximum round trip time */
	double maxrtt;
	/*! Minimum round trip time */
	double minrtt;
	/*! Average round trip time */
	double normdevrtt;
	/*! Standard deviation round trip time */
	double stdevrtt;
	/*! Our SSRC */
	unsigned int local_ssrc;
	/*! Their SSRC */
	unsigned int remote_ssrc;
	/*! The Asterisk channel's unique ID that owns this instance */
	char channel_uniqueid[MAX_CHANNEL_ID];
	/*! Number of octets transmitted */
	unsigned int txoctetcount;
	/*! Number of octets received */
	unsigned int rxoctetcount;

	/*! Media Experience Score on transmitted packets */
	double txmes;
	/*! Media Experience Score on received packets */
	double rxmes;
	/*! Maximum MES on remote side */
	double remote_maxmes;
	/*! Minimum MES on remote side */
	double remote_minmes;
	/*! Average MES on remote side */
	double remote_normdevmes;
	/*! Standard deviation MES on remote side */
	double remote_stdevmes;
	/*! Maximum MES on local side */
	double local_maxmes;
	/*! Minimum MES on local side */
	double local_minmes;
	/*! Average MES on local side */
	double local_normdevmes;
	/*! Standard deviation MES on local side */
	double local_stdevmes;
};

#define AST_RTP_STAT_SET(current_stat, combined, placement, value) \
if (stat == current_stat || stat == AST_RTP_INSTANCE_STAT_ALL || (combined >= 0 && combined == stat)) { \
placement = value; \
if (stat == current_stat) { \
return 0; \
} \
}

#define AST_RTP_STAT_STRCPY(current_stat, combined, placement, value) \
if (stat == current_stat || stat == AST_RTP_INSTANCE_STAT_ALL || (combined >= 0 && combined == stat)) { \
	ast_copy_string(placement, value, sizeof(placement)); \
	if (stat == current_stat) { \
		return 0; \
	} \
}

#define AST_RTP_STAT_TERMINATOR(combined) \
if (stat == combined) { \
return 0; \
}

/*! \brief ICE candidate types */
enum ast_rtp_ice_candidate_type {
	AST_RTP_ICE_CANDIDATE_TYPE_HOST,    /*!< ICE host candidate. A host candidate represents the actual local transport address in the host. */
	AST_RTP_ICE_CANDIDATE_TYPE_SRFLX,   /*!< ICE server reflexive candidate, which represents the public mapped address of the local address. */
	AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, /*!< ICE relayed candidate, which represents the address allocated in TURN server. */
};

/*! \brief ICE component types */
enum ast_rtp_ice_component_type {
	AST_RTP_ICE_COMPONENT_RTP = 1,
	AST_RTP_ICE_COMPONENT_RTCP = 2,
};

/*! \brief ICE role during negotiation */
enum ast_rtp_ice_role {
	AST_RTP_ICE_ROLE_CONTROLLED,
	AST_RTP_ICE_ROLE_CONTROLLING,
};

/*! \brief Structure for an ICE candidate */
struct ast_rtp_engine_ice_candidate {
	char *foundation;                     /*!< Foundation identifier */
	enum ast_rtp_ice_component_type id;   /*!< Component identifier */
	char *transport;                      /*!< Transport for the media */
	int priority;                         /*!< Priority which is used if multiple candidates can be used */
	struct ast_sockaddr address;          /*!< Address of the candidate */
	struct ast_sockaddr relay_address;    /*!< Relay address for the candidate */
	enum ast_rtp_ice_candidate_type type; /*!< Type of candidate */
};

/*! \brief Structure that represents the optional ICE support within an RTP engine */
struct ast_rtp_engine_ice {
	/*! Callback for setting received authentication information */
	void (*set_authentication)(struct ast_rtp_instance *instance, const char *ufrag, const char *password);
	/*! Callback for adding a remote candidate */
	void (*add_remote_candidate)(struct ast_rtp_instance *instance, const struct ast_rtp_engine_ice_candidate *candidate);
	/*! Callback for starting ICE negotiation */
	void (*start)(struct ast_rtp_instance *instance);
	/*! Callback for stopping ICE support */
	void (*stop)(struct ast_rtp_instance *instance);
	/*! Callback for getting local username */
	const char *(*get_ufrag)(struct ast_rtp_instance *instance);
	/*! Callback for getting local password */
	const char *(*get_password)(struct ast_rtp_instance *instance);
	/*! Callback for getting local candidates */
	struct ao2_container *(*get_local_candidates)(struct ast_rtp_instance *instance);
	/*! Callback for telling the ICE support that it is talking to an ice-lite implementation */
	void (*ice_lite)(struct ast_rtp_instance *instance);
	/*! Callback for changing our role in negotiation */
	void (*set_role)(struct ast_rtp_instance *instance, enum ast_rtp_ice_role role);
	/*! Callback for requesting a TURN session */
	void (*turn_request)(struct ast_rtp_instance *instance, enum ast_rtp_ice_component_type component,
		enum ast_transport transport, const char *server, unsigned int port,
		const char *username, const char *password);
	/*! Callback to alter the number of ICE components on a session */
	void (*change_components)(struct ast_rtp_instance *instance, int num_components);
};

/*! \brief DTLS setup types */
enum ast_rtp_dtls_setup {
	AST_RTP_DTLS_SETUP_ACTIVE,   /*!< Endpoint is willing to inititate connections */
	AST_RTP_DTLS_SETUP_PASSIVE,  /*!< Endpoint is willing to accept connections */
	AST_RTP_DTLS_SETUP_ACTPASS,  /*!< Endpoint is willing to both accept and initiate connections */
	AST_RTP_DTLS_SETUP_HOLDCONN, /*!< Endpoint does not want the connection to be established right now */
};

/*! \brief DTLS connection states */
enum ast_rtp_dtls_connection {
	AST_RTP_DTLS_CONNECTION_NEW,      /*!< Endpoint wants to use a new connection */
	AST_RTP_DTLS_CONNECTION_EXISTING, /*!< Endpoint wishes to use existing connection */
};

/*! \brief DTLS fingerprint hashes */
enum ast_rtp_dtls_hash {
	AST_RTP_DTLS_HASH_SHA256, /*!< SHA-256 fingerprint hash */
	AST_RTP_DTLS_HASH_SHA1,   /*!< SHA-1 fingerprint hash */
};

/*! \brief DTLS verification settings */
enum ast_rtp_dtls_verify {
	AST_RTP_DTLS_VERIFY_NONE = 0,               /*!< Don't verify anything */
	AST_RTP_DTLS_VERIFY_FINGERPRINT = (1 << 0), /*!< Verify the fingerprint */
	AST_RTP_DTLS_VERIFY_CERTIFICATE = (1 << 1), /*!< Verify the certificate */
};

/*!
 * \brief Known RTP extensions
 */
enum ast_rtp_extension {
	/*! Per the RFC 0 should not be used, so we treat it as an unsupported extension placeholder */
	AST_RTP_EXTENSION_UNSUPPORTED = 0,
	/*! abs-send-time from https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 */
	AST_RTP_EXTENSION_ABS_SEND_TIME,
	/*! transport-cc from https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01 */
	AST_RTP_EXTENSION_TRANSPORT_WIDE_CC,
	/*! The maximum number of known RTP extensions */
	AST_RTP_EXTENSION_MAX,
};

/*! \brief DTLS configuration structure */
struct ast_rtp_dtls_cfg {
	unsigned int enabled:1;                /*!< Whether DTLS support is enabled or not */
	unsigned int rekey;                    /*!< Interval at which to renegotiate and rekey - defaults to 0 (off) */
	enum ast_rtp_dtls_setup default_setup; /*!< Default setup type to use for outgoing */
	enum ast_srtp_suite suite;             /*!< Crypto suite in use */
	enum ast_rtp_dtls_hash hash;		   /*!< Hash to use for fingerprint */
	enum ast_rtp_dtls_verify verify;	   /*!< What should be verified */
	char *certfile;                        /*!< Certificate file */
	char *pvtfile;                         /*!< Private key file */
	char *cipher;                          /*!< Cipher to use */
	char *cafile;                          /*!< Certificate authority file */
	char *capath;                          /*!< Path to certificate authority */
	unsigned int ephemeral_cert:1;         /*!< Whether to not to generate an ephemeral certificate - defaults to 0 (off) */
};

/*! \brief Structure that represents the optional DTLS SRTP support within an RTP engine */
struct ast_rtp_engine_dtls {
	/*! Set the configuration of the DTLS support on the instance */
	int (*set_configuration)(struct ast_rtp_instance *instance, const struct ast_rtp_dtls_cfg *dtls_cfg);
	/*! Get if the DTLS SRTP support is active or not */
	int (*active)(struct ast_rtp_instance *instance);
	/*! Stop and terminate DTLS SRTP support */
	void (*stop)(struct ast_rtp_instance *instance);
	/*! Reset the connection and start fresh */
	void (*reset)(struct ast_rtp_instance *instance);
	/*! Get the current connection state */
	enum ast_rtp_dtls_connection (*get_connection)(struct ast_rtp_instance *instance);
	/*! Get the current setup state */
	enum ast_rtp_dtls_setup (*get_setup)(struct ast_rtp_instance *instance);
	/*! Set the remote setup state */
	void (*set_setup)(struct ast_rtp_instance *instance, enum ast_rtp_dtls_setup setup);
	/*! Set the remote fingerprint */
	void (*set_fingerprint)(struct ast_rtp_instance *instance, enum ast_rtp_dtls_hash hash, const char *fingerprint);
	/*! Get the local fingerprint hash type */
	enum ast_rtp_dtls_hash (*get_fingerprint_hash)(struct ast_rtp_instance *instance);
	/*! Get the local fingerprint */
	const char *(*get_fingerprint)(struct ast_rtp_instance *instance);
};

#ifdef TEST_FRAMEWORK
/*! \brief Structure that represents the test functionality for res_rtp_asterisk unit tests */
struct ast_rtp_engine_test {
	/*! Drops RTP packets while this has a value greater than 0 */
	int packets_to_drop;
	/*! Sends a SR/RR instead of RTP the next time RTP would be sent */
	int send_report;
	/*! Set to 1 whenever SDES is received */
	int sdes_received;
	/*! Get the number of packets in the receive buffer for a RTP instance */
	size_t (*recv_buffer_count)(struct ast_rtp_instance *instance);
	/*! Get the maximum number of packets the receive buffer can hold for a RTP instance */
	size_t (*recv_buffer_max)(struct ast_rtp_instance *instance);
	/*! Get the number of packets in the send buffer for a RTP instance */
	size_t (*send_buffer_count)(struct ast_rtp_instance *instance);
	/*! Set the schedid for RTCP */
	void (*set_schedid)(struct ast_rtp_instance *instance, int id);
};
#endif

/*! Structure that represents an RTP stack (engine) */
struct ast_rtp_engine {
	/*! Name of the RTP engine, used when explicitly requested */
	const char *name;
	/*! Module this RTP engine came from, used for reference counting */
	struct ast_module *mod;
	/*! Callback for setting up a new RTP instance */
	int (*new)(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *sa, void *data);
	/*! Callback for destroying an RTP instance */
	int (*destroy)(struct ast_rtp_instance *instance);
	/*! Callback for writing out a frame */
	int (*write)(struct ast_rtp_instance *instance, struct ast_frame *frame);
	/*! Callback for stopping the RTP instance */
	void (*stop)(struct ast_rtp_instance *instance);
	/*! Callback for starting RFC2833 DTMF transmission */
	int (*dtmf_begin)(struct ast_rtp_instance *instance, char digit);
	/*! Callback for stopping RFC2833 DTMF transmission */
	int (*dtmf_end)(struct ast_rtp_instance *instance, char digit);
	int (*dtmf_end_with_duration)(struct ast_rtp_instance *instance, char digit, unsigned int duration);
	/*! Callback to indicate that we should update the marker bit */
	void (*update_source)(struct ast_rtp_instance *instance);
	/*! Callback to indicate that we should update the marker bit and ssrc */
	void (*change_source)(struct ast_rtp_instance *instance);
	/*! Callback for setting an extended RTP property */
	int (*extended_prop_set)(struct ast_rtp_instance *instance, int property, void *value);
	/*! Callback for getting an extended RTP property */
	void *(*extended_prop_get)(struct ast_rtp_instance *instance, int property);
	/*! Callback for setting an RTP property */
	void (*prop_set)(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
	/*! Callback for setting a payload.  If asterisk  is to be used, asterisk_format will be set, otherwise value in code is used. */
	void (*payload_set)(struct ast_rtp_instance *instance, int payload, int asterisk_format, struct ast_format *format, int code);
	/*! Callback for setting the remote address that RTP is to be sent to */
	void (*remote_address_set)(struct ast_rtp_instance *instance, struct ast_sockaddr *sa);
	/*! Callback for changing DTMF mode */
	int (*dtmf_mode_set)(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode);
	/*! Callback for getting DTMF mode */
	enum ast_rtp_dtmf_mode (*dtmf_mode_get)(struct ast_rtp_instance *instance);
	/*! Callback for retrieving statistics */
	int (*get_stat)(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat);
	/*! Callback for setting QoS values */
	int (*qos)(struct ast_rtp_instance *instance, int tos, int cos, const char *desc);
	/*! Callback for retrieving a file descriptor to poll on, not always required */
	int (*fd)(struct ast_rtp_instance *instance, int rtcp);
	/*! Callback for initializing RED support */
	int (*red_init)(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations);
	/*! Callback for buffering a frame using RED */
	int (*red_buffer)(struct ast_rtp_instance *instance, struct ast_frame *frame);
	/*! Callback for reading a frame from the RTP engine */
	struct ast_frame *(*read)(struct ast_rtp_instance *instance, int rtcp);
	/*! Callback to locally bridge two RTP instances */
	int (*local_bridge)(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1);
	/*! Callback to set the read format */
	int (*set_read_format)(struct ast_rtp_instance *instance, struct ast_format *format);
	/*! Callback to set the write format */
	int (*set_write_format)(struct ast_rtp_instance *instance, struct ast_format *format);
	/*! Callback to make two instances compatible */
	int (*make_compatible)(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
	/*! Callback to see if two instances are compatible with DTMF */
	int (*dtmf_compatible)(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
	/*! Callback to indicate that packets will now flow */
	int (*activate)(struct ast_rtp_instance *instance);
	/*! Callback to request that the RTP engine send a STUN BIND request */
	void (*stun_request)(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
	/*! Callback to get the transcodeable formats supported. result returned in ast_format_cap *result */
	void (*available_formats)(struct ast_rtp_instance *instance, struct ast_format_cap *to_endpoint, struct ast_format_cap *to_asterisk, struct ast_format_cap *result);
	/*! Callback to send CNG */
	int (*sendcng)(struct ast_rtp_instance *instance, int level);
	/*! Callback to retrieve local SSRC */
	unsigned int (*ssrc_get)(struct ast_rtp_instance *instance);
	/*! Callback to retrieve RTCP SDES CNAME */
	const char *(*cname_get)(struct ast_rtp_instance *instance);
	/*! Callback to bundle an RTP instance to another */
	int (*bundle)(struct ast_rtp_instance *child, struct ast_rtp_instance *parent);
	/*! Callback to set remote SSRC information */
	void (*set_remote_ssrc)(struct ast_rtp_instance *instance, unsigned int ssrc);
	/*! Callback to set the stream identifier */
	void (*set_stream_num)(struct ast_rtp_instance *instance, int stream_num);
	/*! Callback to pointer for optional ICE support */
	struct ast_rtp_engine_ice *ice;
	/*! Callback to pointer for optional DTLS SRTP support */
	struct ast_rtp_engine_dtls *dtls;
#ifdef TEST_FRAMEWORK
	/*! Callback to pointer for test callbacks for RTP/RTCP unit tests */
	struct ast_rtp_engine_test *test;
#endif
	/*! Callback to enable an RTP extension (returns non-zero if supported) */
	int (*extension_enable)(struct ast_rtp_instance *instance, enum ast_rtp_extension extension);
	/*! Linked list information */
	AST_RWLIST_ENTRY(ast_rtp_engine) entry;
};

/*! Structure that represents codec and packetization information */
struct ast_rtp_codecs {
	/*! RW lock that protects elements in this structure */
	ast_rwlock_t codecs_lock;
	/*! Rx payload type mapping exceptions */
	AST_VECTOR(, struct ast_rtp_payload_type *) payload_mapping_rx;
	/*! Tx payload type mapping */
	AST_VECTOR(, struct ast_rtp_payload_type *) payload_mapping_tx;
	/*! The framing for this media session */
	unsigned int framing;
	/*! The preferred format, as the mappings are numerically sorted */
	struct ast_format *preferred_format;
	/*! The preferred dtmf sample rate */
	int preferred_dtmf_rate;
	/*! The preferred dtmf payload type */
	int preferred_dtmf_pt;
};

#define AST_RTP_CODECS_NULL_INIT \
    { .codecs_lock = AST_RWLOCK_INIT_VALUE, .payload_mapping_rx = { 0, }, .payload_mapping_tx = { 0, }, .framing = 0, .preferred_format = NULL, \
	.preferred_dtmf_rate = -1, .preferred_dtmf_pt = -1}

/*! Structure that represents the glue that binds an RTP instance to a channel */
struct ast_rtp_glue {
	/*! Name of the channel driver that this glue is responsible for */
	const char *type;
	/*! Module that the RTP glue came from */
	struct ast_module *mod;
	/*!
	 * \brief Callback for retrieving the RTP instance carrying audio
	 * \note This function increases the reference count on the returned RTP instance.
	 */
	enum ast_rtp_glue_result (*get_rtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
	/*!
	 * \brief Used to prevent two channels from remotely bridging audio rtp if the channel tech has a
	 *        reason for prohibiting it based on qualities that need to be compared from both channels.
	 * \note This function may be NULL for a given channel driver. This should be accounted for and if that is the case, this function is not used.
	 */
	int (*allow_rtp_remote)(struct ast_channel *chan1, struct ast_rtp_instance *instance);
	/*!
	 * \brief Callback for retrieving the RTP instance carrying video
	 * \note This function increases the reference count on the returned RTP instance.
	 * \note This function may be NULL for a given channel driver. This should be accounted for and if that is the case, this function is not used.
	 */
	enum ast_rtp_glue_result (*get_vrtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
	/*!
	 * \brief Used to prevent two channels from remotely bridging video rtp if the channel tech has a
	 *        reason for prohibiting it based on qualities that need to be compared from both channels.
	 * \note This function may be NULL for a given channel driver. This should be accounted for and if that is the case, this function is not used.
	 */
	int (*allow_vrtp_remote)(struct ast_channel *chan1, struct ast_rtp_instance *instance);

	/*!
	 * \brief Callback for retrieving the RTP instance carrying text
	 * \note This function increases the reference count on the returned RTP instance.
	 * \note This function may be NULL for a given channel driver. This should be accounted for and if that is the case, this function is not used.
	 */
	enum ast_rtp_glue_result (*get_trtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
	/*! Callback for updating the destination that the remote side should send RTP to */
	int (*update_peer)(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active);
	/*!
	 * \brief Callback for retrieving codecs that the channel can do.  Result returned in result_cap.
	 * \note This function may be NULL for a given channel driver. This should be accounted for and if that is the case, this function is not used.
	 */
	void (*get_codec)(struct ast_channel *chan, struct ast_format_cap *result_cap);
	/*! Linked list information */
	AST_RWLIST_ENTRY(ast_rtp_glue) entry;
};

/*!
 * \brief Directions for RTP extensions
 */
enum ast_rtp_extension_direction {
	/*! The extension is not negotiated and is not flowing */
	AST_RTP_EXTENSION_DIRECTION_NONE,
	/*! Send and receive */
	AST_RTP_EXTENSION_DIRECTION_SENDRECV,
	/*! Send only */
	AST_RTP_EXTENSION_DIRECTION_SENDONLY,
	/*! Receive only */
	AST_RTP_EXTENSION_DIRECTION_RECVONLY,
	/*! Negotiated but not sending or receiving */
	AST_RTP_EXTENSION_DIRECTION_INACTIVE,
};

/*!
 * \brief Allocation routine for \ref ast_rtp_payload_type
 *
 * \retval NULL on error
 * \return An ao2 ref counted \c ast_rtp_payload_type on success.
 *
 * \note The \c ast_rtp_payload_type returned by this function is an
 *       ao2 ref counted object.
 *
 */
struct ast_rtp_payload_type *ast_rtp_engine_alloc_payload_type(void);

#define ast_rtp_engine_register(engine) ast_rtp_engine_register2(engine, AST_MODULE_SELF)

/*!
 * \brief Register an RTP engine
 *
 * \param engine Structure of the RTP engine to register
 * \param module Module that the RTP engine is part of
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_engine_register2(&example_rtp_engine, NULL);
 * \endcode
 *
 * This registers the RTP engine declared as example_rtp_engine with the RTP engine core, but does not
 * associate a module with it.
 *
 * \note It is recommended that you use the ast_rtp_engine_register macro so that the module is
 *       associated with the RTP engine and use counting is performed.
 *
 * \since 1.8
 */
int ast_rtp_engine_register2(struct ast_rtp_engine *engine, struct ast_module *module);

/*!
 * \brief Unregister an RTP engine
 *
 * \param engine Structure of the RTP engine to unregister
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_engine_unregister(&example_rtp_engine);
 * \endcode
 *
 * This unregisters the RTP engine declared as example_rtp_engine from the RTP engine core. If a module
 * reference was provided when it was registered then this will only be called once the RTP engine is no longer in use.
 *
 * \since 1.8
 */
int ast_rtp_engine_unregister(struct ast_rtp_engine *engine);

int ast_rtp_engine_register_srtp(struct ast_srtp_res *srtp_res, struct ast_srtp_policy_res *policy_res);

void ast_rtp_engine_unregister_srtp(void);
int ast_rtp_engine_srtp_is_registered(void);

#define ast_rtp_glue_register(glue) ast_rtp_glue_register2(glue, AST_MODULE_SELF)

/*!
 * \brief Register RTP glue
 *
 * \param glue The glue to register
 * \param module Module that the RTP glue is part of
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_glue_register2(&example_rtp_glue, NULL);
 * \endcode
 *
 * This registers the RTP glue declared as example_rtp_glue with the RTP engine core, but does not
 * associate a module with it.
 *
 * \note It is recommended that you use the ast_rtp_glue_register macro so that the module is
 *       associated with the RTP glue and use counting is performed.
 *
 * \since 1.8
 */
int ast_rtp_glue_register2(struct ast_rtp_glue *glue, struct ast_module *module);

/*!
 * \brief Unregister RTP glue
 *
 * \param glue The glue to unregister
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_glue_unregister(&example_rtp_glue);
 * \endcode
 *
 * This unregisters the RTP glue declared as example_rtp_gkue from the RTP engine core. If a module
 * reference was provided when it was registered then this will only be called once the RTP engine is no longer in use.
 *
 * \since 1.8
 */
int ast_rtp_glue_unregister(struct ast_rtp_glue *glue);

/*!
 * \brief Create a new RTP instance
 *
 * \param engine_name Name of the engine to use for the RTP instance
 * \param sched Scheduler context that the RTP engine may want to use
 * \param sa Address we want to bind to
 * \param data Unique data for the engine
 *
 * \retval non-NULL success
 * \retval NULL failure
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_instance *instance = NULL;
 * instance = ast_rtp_instance_new(NULL, sched, &sin, NULL);
 * \endcode
 *
 * This creates a new RTP instance using the default engine and asks the RTP engine to bind to the address given
 * in the address structure.
 *
 * \note The RTP engine does not have to use the address provided when creating an RTP instance. It may choose to use
 *       another depending on it's own configuration.
 *
 * \since 1.8
 */
struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name,
                struct ast_sched_context *sched, const struct ast_sockaddr *sa,
                void *data);

/*!
 * \brief Destroy an RTP instance
 *
 * \param instance The RTP instance to destroy
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_destroy(instance);
 * \endcode
 *
 * This destroys the RTP instance pointed to by instance. Once this function returns instance no longer points to valid
 * memory and may not be used again.
 *
 * \since 1.8
 */
int ast_rtp_instance_destroy(struct ast_rtp_instance *instance);

/*!
 * \brief Set the data portion of an RTP instance
 *
 * \param instance The RTP instance to manipulate
 * \param data Pointer to data
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_data(instance, blob);
 * \endcode
 *
 * This sets the data pointer on the RTP instance pointed to by 'instance' to
 * blob.
 *
 * \since 1.8
 */
void ast_rtp_instance_set_data(struct ast_rtp_instance *instance, void *data);

/*!
 * \brief Get the data portion of an RTP instance
 *
 * \param instance The RTP instance we want the data portion from
 *
 * Example usage:
 *
 * \code
 * struct *blob = ast_rtp_instance_get_data(instance);
 ( \endcode
 *
 * This gets the data pointer on the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
void *ast_rtp_instance_get_data(struct ast_rtp_instance *instance);

/*!
 * \brief Send a frame out over RTP
 *
 * \param instance The RTP instance to send frame out on
 * \param frame the frame to send out
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_write(instance, frame);
 * \endcode
 *
 * This gives the frame pointed to by frame to the RTP engine being used for the instance
 * and asks that it be transmitted to the current remote address set on the RTP instance.
 *
 * \since 1.8
 */
int ast_rtp_instance_write(struct ast_rtp_instance *instance, struct ast_frame *frame);

/*!
 * \brief Receive a frame over RTP
 *
 * \param instance The RTP instance to receive frame on
 * \param rtcp Whether to read in RTCP or not
 *
 * \retval non-NULL success
 * \retval NULL failure
 *
 * Example usage:
 *
 * \code
 * struct ast_frame *frame;
 * frame = ast_rtp_instance_read(instance, 0);
 * \endcode
 *
 * This asks the RTP engine to read in RTP from the instance and return it as an Asterisk frame.
 *
 * \since 1.8
 */
struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp);

/*!
 * \brief Set the incoming source address of the remote endpoint that we are sending RTP to
 *
 * This sets the incoming source address the engine is sending RTP to. Usually this
 * will be the same as the requested target address, however in the case where
 * the engine "learns" the address (for instance, symmetric RTP enabled) this
 * will then contain the learned address.
 *
 * \param instance The RTP instance to change the address on
 * \param address Address to set it to
 *
 * \retval 0 success
 * \retval -1 failure
 */
int ast_rtp_instance_set_incoming_source_address(struct ast_rtp_instance *instance,
						 const struct ast_sockaddr *address);

/*!
 * \brief Set the requested target address of the remote endpoint
 *
 * This should always be the address of the remote endpoint. Consequently, this can differ
 * from the address the engine is sending RTP to.  However, usually they will be the same
 * except in some circumstances (for instance when the engine "learns" the address if
 * symmetric RTP is enabled).
 *
 * \param instance The RTP instance to change the address on
 * \param address Address to set it to
 *
 * \retval 0 success
 * \retval -1 failure
 */
int ast_rtp_instance_set_requested_target_address(struct ast_rtp_instance *instance,
						  const struct ast_sockaddr *address);

/*!
 * \brief Set the address of the remote endpoint that we are sending RTP to
 *
 * \param instance The RTP instance to change the address on
 * \param address Address to set it to
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_remote_address(instance, &sin);
 * \endcode
 *
 * This changes the remote address that RTP will be sent to on instance to the address given in the sin
 * structure.
 *
 * \since 1.8
 */
#define ast_rtp_instance_set_remote_address(instance, address) \
	ast_rtp_instance_set_requested_target_address((instance), (address))

/*!
 * \brief Set the address that we are expecting to receive RTP on
 *
 * \param instance The RTP instance to change the address on
 * \param address Address to set it to
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_local_address(instance, &sin);
 * \endcode
 *
 * This changes the local address that RTP is expected on to the address given in the sin
 * structure.
 *
 * \since 1.8
 */
int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance,
                const struct ast_sockaddr *address);

/*!
 * \brief Get the local address that we are expecting RTP on
 *
 * \param instance The RTP instance to get the address from
 * \param address The variable to store the address in
 *
 * Example usage:
 *
 * \code
 * struct ast_sockaddr address;
 * ast_rtp_instance_get_local_address(instance, &address);
 * \endcode
 *
 * This gets the local address that we are expecting RTP on and stores it in the 'address' structure.
 *
 * \since 1.8
 */
void ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct ast_sockaddr *address);

/*!
 * \brief Get the address of the local endpoint that we are sending RTP to, comparing its address to another
 *
 * \param instance The instance that we want to get the local address for
 * \param address An initialized address that may be overwritten if the local address is different
 *
 * \retval 0 address was not changed
 * \retval 1 address was changed
 * Example usage:
 *
 * \code
 * struct ast_sockaddr address;
 * int ret;
 * ret = ast_rtp_instance_get_and_cmp_local_address(instance, &address);
 * \endcode
 *
 * This retrieves the current local address set on the instance pointed to by instance and puts the value
 * into the address structure.
 *
 * \since 1.8
 */
int ast_rtp_instance_get_and_cmp_local_address(struct ast_rtp_instance *instance, struct ast_sockaddr *address);

/*!
 * \brief Get the incoming source address of the remote endpoint
 *
 * This returns the remote address the engine is sending RTP to. Usually this
 * will be the same as the requested target address, however in the case where
 * the engine "learns" the address (for instance, symmetric RTP enabled) this
 * will then contain the learned address.
 *
 * \param instance The instance that we want to get the incoming source address for
 * \param address A structure to put the address into
 */
void ast_rtp_instance_get_incoming_source_address(struct ast_rtp_instance *instance, struct ast_sockaddr *address);

/*!
 * \brief Get the requested target address of the remote endpoint
 *
 * This returns the explicitly set address of a remote endpoint. Meaning this won't change unless
 * specifically told to change. In most cases this should be the same as the incoming source
 * address, except in cases where the engine "learns" the address in which case this and the
 * incoming source address might differ.
 *
 * \param instance The instance that we want to get the requested target address for
 * \param address A structure to put the address into
 */
void ast_rtp_instance_get_requested_target_address(struct ast_rtp_instance *instance, struct ast_sockaddr *address);

/*!
 * \brief Get the address of the remote endpoint that we are sending RTP to
 *
 * \param instance The instance that we want to get the remote address for
 * \param address A structure to put the address into
 *
 * Example usage:
 *
 * \code
 * struct ast_sockaddr address;
 * ast_rtp_instance_get_remote_address(instance, &address);
 * \endcode
 *
 * This retrieves the current remote address set on the instance pointed to by instance and puts the value
 * into the address structure.
 *
 * \since 1.8
 */
#define ast_rtp_instance_get_remote_address(instance, address) \
	ast_rtp_instance_get_incoming_source_address((instance), (address))

/*!
 * \brief Get the requested target address of the remote endpoint and
 *        compare it to the given address
 *
 * \param instance The instance that we want to get the remote address for
 * \param address An initialized address that may be overwritten addresses differ
 *
 * \retval 0 address was not changed
 * \retval 1 address was changed
 */
int ast_rtp_instance_get_and_cmp_requested_target_address(struct ast_rtp_instance *instance, struct ast_sockaddr *address);

/*!
 * \brief Get the address of the remote endpoint that we are sending RTP to, comparing its address to another
 *
 * \param instance The instance that we want to get the remote address for
 * \param address An initialized address that may be overwritten if the remote address is different
 *
 * \retval 0 address was not changed
 * \retval 1 address was changed
 * Example usage:
 *
 * \code
 * struct ast_sockaddr address;
 * int ret;
 * ret = ast_rtp_instance_get_and_cmp_remote_address(instance, &address);
 * \endcode
 *
 * This retrieves the current remote address set on the instance pointed to by instance and puts the value
 * into the address structure.
 *
 * \since 1.8
 */
#define ast_rtp_instance_get_and_cmp_remote_address(instance, address) \
	ast_rtp_instance_get_and_cmp_requested_target_address((instance), (address))

/*!
 * \brief Set the value of an RTP instance extended property
 *
 * \param instance The RTP instance to set the extended property on
 * \param property The extended property to set
 * \param value The value to set the extended property to
 *
 * \since 1.8
 */
void ast_rtp_instance_set_extended_prop(struct ast_rtp_instance *instance, int property, void *value);

/*!
 * \brief Get the value of an RTP instance extended property
 *
 * \param instance The RTP instance to get the extended property on
 * \param property The extended property to get
 *
 * \since 1.8
 */
void *ast_rtp_instance_get_extended_prop(struct ast_rtp_instance *instance, int property);

/*!
 * \brief Set the value of an RTP instance property
 *
 * \param instance The RTP instance to set the property on
 * \param property The property to modify
 * \param value The value to set the property to
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_NAT, 1);
 * \endcode
 *
 * This enables the AST_RTP_PROPERTY_NAT property on the instance pointed to by instance.
 *
 * \since 1.8
 */
void ast_rtp_instance_set_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);

/*!
 * \brief Get the value of an RTP instance property
 *
 * \param instance The RTP instance to get the property from
 * \param property The property to get
 *
 * \return Current value of the property
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT);
 * \endcode
 *
 * This returns the current value of the NAT property on the instance pointed to by instance.
 *
 * \since 1.8
 */
int ast_rtp_instance_get_prop(struct ast_rtp_instance *instance, enum ast_rtp_property property);

/*!
 * \brief Get the codecs structure of an RTP instance
 *
 * \param instance The RTP instance to get the codecs structure from
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_codecs *codecs = ast_rtp_instance_get_codecs(instance);
 * \endcode
 *
 * This gets the codecs structure on the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance);

/*!
 * \brief Enable support for an RTP extension on an instance
 *
 * \param instance The RTP instance to enable the extension on
 * \param id The unique local identifier to use for this extension (-1 to have one auto selected)
 * \param extension The RTP extension
 * \param direction The initial direction that the RTP extension should be used in
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * \since 15.5.0
 */
int ast_rtp_instance_extmap_enable(struct ast_rtp_instance *instance, int id, enum ast_rtp_extension extension,
	enum ast_rtp_extension_direction direction);

/*!
 * \brief Negotiate received RTP extension information
 *
 * \param instance The RTP instance to set the extension on
 * \param id The local identifier for the extension
 * \param direction The direction that the extension should be used in
 * \param uri The unique URI for the extension
 * \param attributes Attributes specific to this extension (if NULL or empty then no attributes)
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * \since 15.5.0
 */
int ast_rtp_instance_extmap_negotiate(struct ast_rtp_instance *instance, int id, enum ast_rtp_extension_direction direction,
	const char *uri, const char *attributes);

/*!
 * \brief Clear negotiated RTP extension information
 *
 * \param instance The RTP instance to clear negotiated extension information on
 */
void ast_rtp_instance_extmap_clear(struct ast_rtp_instance *instance);

/*!
 * \brief Retrieve the id for an RTP extension
 *
 * \param instance The RTP instance to retrieve the id from
 * \param extension The RTP extension
 *
 * \retval -1 not negotiated
 * \retval id if negotiated
 *
 * \since 15.5.0
 */
int ast_rtp_instance_extmap_get_id(struct ast_rtp_instance *instance, enum ast_rtp_extension extension);

/*!
 * \brief Get the number of known unique identifiers
 *
 * \param instance The RTP instance to retrieve the count from
 *
 * \return the number of known unique identifiers
 *
 * \since 15.5.0
 */
size_t ast_rtp_instance_extmap_count(struct ast_rtp_instance *instance);

/*!
 * \brief Retrieve the extension for an RTP extension id
 *
 * \param instance The RTP instance to retrieve the extension from
 * \param id The negotiated RTP extension id
 *
 * \return extension the extension that maps to the id
 *
 * \since 15.5.0
 *
 * \note This will return AST_RTP_EXTENSION_UNSUPPORTED if an extension was proposed for this unique identifier
 * but it is not supported or if the unique identifier is unused.
 */
enum ast_rtp_extension ast_rtp_instance_extmap_get_extension(struct ast_rtp_instance *instance, int id);

/*!
 * \brief Retrieve the negotiated direction for an RTP extension id
 *
 * \param instance The RTP instance to retrieve the direction from
 * \param id The negotiated RTP extension id
 *
 * \return direction the direction that has been negotiated
 *
 * \since 15.5.0
 */
enum ast_rtp_extension_direction ast_rtp_instance_extmap_get_direction(struct ast_rtp_instance *instance, int id);

/*!
 * \brief Retrieve the URI for an RTP extension id
 *
 * \param instance The RTP instance to retrieve the direction from
 * \param id The negotiated RTP extension id
 *
 * \return The URI for the RTP extension
 *
 * \since 15.5.0
 */
const char *ast_rtp_instance_extmap_get_uri(struct ast_rtp_instance *instance, int id);

/*!
 * \brief Initialize an RTP codecs structure
 *
 * \param codecs The codecs structure to initialize
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_codecs codecs;
 * ast_rtp_codecs_payloads_initialize(&codecs);
 * \endcode
 *
 * \since 11
 */
int ast_rtp_codecs_payloads_initialize(struct ast_rtp_codecs *codecs);

/*!
 * \brief Destroy the contents of an RTP codecs structure (but not the structure itself)
 *
 * \param codecs The codecs structure to destroy the contents of
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_codecs codecs;
 * ast_rtp_codecs_payloads_destroy(&codecs);
 * \endcode
 *
 * \since 11
 */
void ast_rtp_codecs_payloads_destroy(struct ast_rtp_codecs *codecs);

/*!
 * \brief Clear rx and tx payload mapping information from an RTP instance
 *
 * \param codecs The codecs structure that payloads will be cleared from
 * \param instance Optionally the instance that the codecs structure belongs to
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_codecs codecs;
 * ast_rtp_codecs_payloads_clear(&codecs, NULL);
 * \endcode
 *
 * This clears the codecs structure and puts it into a pristine state.
 *
 * \since 1.8
 */
void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance);

/*!
 * \brief Copy payload information from one RTP instance to another
 *
 * \param src The source codecs structure
 * \param dest The destination codecs structure that the values from src will be copied to
 * \param instance Optionally the instance that the dst codecs structure belongs to
 *
 * Example usage:
 *
 * \code
 * ast_rtp_codecs_payloads_copy(&codecs0, &codecs1, NULL);
 * \endcode
 *
 * This copies the payloads from the codecs0 structure to the codecs1 structure, overwriting any current values.
 *
 * \since 1.8
 */
void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance);

/*!
 * \brief Crossover copy the tx payload mapping of src to the rx payload mapping of dest.
 * \since 14.0.0
 *
 * \param src The source codecs structure
 * \param dest The destination codecs structure that the values from src will be copied to
 * \param instance Optionally the instance that the dst codecs structure belongs to
 */
void ast_rtp_codecs_payloads_xover(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance);

/*!
 * \brief Record tx payload type information that was seen in an m= SDP line
 *
 * \param codecs The codecs structure to muck with
 * \param instance Optionally the instance that the codecs structure belongs to
 * \param payload Numerical payload that was seen in the m= SDP line
 *
 * Example usage:
 *
 * \code
 * ast_rtp_codecs_payloads_set_m_type(&codecs, NULL, 0);
 * \endcode
 *
 * This records that the numerical payload '0' was seen in the codecs structure.
 *
 * \since 1.8
 */
void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload);

/*!
 * \brief Record tx payload type information that was seen in an a=rtpmap: SDP line
 *
 * \param codecs The codecs structure to muck with
 * \param instance Optionally the instance that the codecs structure belongs to
 * \param payload Numerical payload that was seen in the a=rtpmap: SDP line
 * \param mimetype The string mime type that was seen
 * \param mimesubtype The string mime sub type that was seen
 * \param options Optional options that may change the behavior of this specific payload
 *
 * \retval 0 success
 * \retval -1 failure, invalid payload numbe
 * \retval -2 failure, unknown mimetype
 *
 * Example usage:
 *
 * \code
 * ast_rtp_codecs_payloads_set_rtpmap_type(&codecs, NULL, 0, "audio", "PCMU", 0);
 * \endcode
 *
 * This records that the numerical payload '0' was seen with mime type 'audio' and sub mime type 'PCMU' in the codecs structure.
 *
 * \since 1.8
 */
int ast_rtp_codecs_payloads_set_rtpmap_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload, char *mimetype, char *mimesubtype, enum ast_rtp_options options);

/*!
 * \brief Set tx payload type to a known MIME media type for a codec with a specific sample rate
 *
 * \param codecs RTP structure to modify
 * \param instance Optionally the instance that the codecs structure belongs to
 * \param pt Payload type entry to modify
 * \param mimetype top-level MIME type of media stream (typically "audio", "video", "text", etc.)
 * \param mimesubtype MIME subtype of media stream (typically a codec name)
 * \param options Zero or more flags from the ast_rtp_options enum
 * \param sample_rate The sample rate of the media stream
 *
 * This function 'fills in' an entry in the list of possible formats for
 * a media stream associated with an RTP structure.
 *
 * \retval 0 on success
 * \retval -1 if the payload type is out of range
 * \retval -2 if the mimeType/mimeSubtype combination was not found
 *
 * \since 1.8
 */
int ast_rtp_codecs_payloads_set_rtpmap_type_rate(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int pt,
				  char *mimetype, char *mimesubtype,
				  enum ast_rtp_options options,
				  unsigned int sample_rate);

/*!
 * \brief Remove tx payload type mapped information
 *
 * \param codecs The codecs structure to muck with
 * \param instance Optionally the instance that the codecs structure belongs to
 * \param payload Numerical payload to unset
 *
 * Example usage:
 *
 * \code
 * ast_rtp_codecs_payloads_unset(&codecs, NULL, 0);
 * \endcode
 *
 * This clears the payload '0' from the codecs structure. It will be as if it was never set.
 *
 * \since 1.8
 */
void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload);

/*!
 * \brief Determine the type of RTP stream media from the codecs mapped.
 * \since 13.19.0
 *
 * \param codecs Codecs structure to look in
 *
 * \return Media type or AST_MEDIA_TYPE_UNKNOWN if no codecs mapped.
 */
enum ast_media_type ast_rtp_codecs_get_stream_type(struct ast_rtp_codecs *codecs);

/*!
 * \brief Retrieve rx payload mapped information by payload type
 *
 * \param codecs Codecs structure to look in
 * \param payload Numerical payload to look up
 *
 * \return Payload information.
 * \retval NULL if payload does not exist.
 *
 * \note The payload returned by this function has its reference count increased.
 *       Callers are responsible for decrementing the reference count.
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_payload_type *payload_type;
 * payload_type = ast_rtp_codecs_get_payload(&codecs, 0);
 * \endcode
 *
 * This looks up the information for payload '0' from the codecs structure.
 */
struct ast_rtp_payload_type *ast_rtp_codecs_get_payload(struct ast_rtp_codecs *codecs, int payload);

/*!
 * \brief Retrieve rx preferred format
 *
 * \param codecs Codecs structure to look in
 *
 * \return format information.
 * \retval NULL if format does not exist.
 *
 * \note The format returned by this function has its reference count increased.
 *       Callers are responsible for decrementing the reference count.
 *
 * Example usage:
 *
 * \code
 * struct ast_format *payload_format;
 * payload_format = ast_rtp_codecs_get_preferred_format(&codecs);
 * \endcode
 *
 * This looks up the preferred format on the codec
 */
struct ast_format *ast_rtp_codecs_get_preferred_format(struct ast_rtp_codecs *codecs);

/*!
 * \brief Set the preferred format
 *
 * \param codecs Codecs structure to set the preferred format in
 * \param format Preferred format to set.
 *
 * \return 0
 *
 * \note The format passed this function has its reference count increased. If an existing
 * 		 format is set, that format is replaced.
 *
 * Example usage:
 *
 * \code
 * struct ast_format *preferred_format = ast_format_cap_get_format(joint, 0);
 * ast_rtp_codecs_set_preferred_format(&codecs, preferred_format));
 * \endcode
 *
 * This sets the first joint format as the preferred format.
 */
int ast_rtp_codecs_set_preferred_format(struct ast_rtp_codecs *codecs, struct ast_format *format);

/*!
 * \brief Retrieve rx preferred dtmf format payload type
 *
 * \param codecs Codecs structure to look in
 *
 * \return Payload type of preferred dtmf format.
 * \retval -1 if not set.
 *
 * Example usage:
 *
 * \code
 * int payload;
 * payload = ast_rtp_codecs_get_preferred_dtmf_format_pt(codec);
 * \endcode
 *
 * This looks up the preferred dtmf format pt on the codec
 */
int ast_rtp_codecs_get_preferred_dtmf_format_pt(struct ast_rtp_codecs *codecs);

/*!
 * \brief Retrieve rx preferred dtmf format sample rate
 *
 * \param codecs Codecs structure to look in
 *
 * \return Sample rate of preferred dtmf format.
 * \retval -1 if not set.
 *
 * Example usage:
 *
 * \code
 * int sample_rate;
 * sample_rate = ast_rtp_codecs_get_preferred_dtmf_format_rate(codec);
 * \endcode
 *
 * This looks up the preferred dtmf format sample rate on the codec
 */
int ast_rtp_codecs_get_preferred_dtmf_format_rate(struct ast_rtp_codecs *codecs);

/*!
 * \brief Set the preferred dtmf format pt and sample rate
 *
 * \param codecs Codecs structure to set the preferred format in
 * \param pt Preferred dtmf payload type to set.
 * \param rate Preferred dtmf payload rate to set.
 *
 * \return 0
 *
 * \note The format passed this function has its reference count increased. If an existing
 * 		 format is set, that format is replaced.
 *
 * Example usage:
 *
 * \code
 * int dtmf_code = atoi(dtmf_pt);
 * int dtmf_rate = clock_rate;
 * ast_rtp_codecs_set_preferred_dtmf_format(codecs, dtmf_code, dtmf_rate);
 * \endcode
 *
 * This sets the preferred dtmf_code and dtmf_rate on the codec.
 */
int ast_rtp_codecs_set_preferred_dtmf_format(struct ast_rtp_codecs *codecs, int pt, int rate);


/*!
 * \brief Update the format associated with a tx payload type in a codecs structure
 *
 * \param codecs Codecs structure to operate on
 * \param payload Numerical payload type to look up
 * \param format The format to replace the existing one
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * \since 13
 */
int ast_rtp_codecs_payload_replace_format(struct ast_rtp_codecs *codecs, int payload, struct ast_format *format);

/*!
 * \brief Retrieve the actual ast_format stored on the codecs structure for a specific tx payload type
 *
 * \param codecs Codecs structure to look in
 * \param payload Numerical payload type to look up
 *
 * \return pointer to format structure on success
 * \retval NULL on failure
 *
 * \note The format returned by this function has its reference count increased.
 *       Callers are responsible for decrementing the reference count.
 *
 * \since 10.0
 */
struct ast_format *ast_rtp_codecs_get_payload_format(struct ast_rtp_codecs *codecs, int payload);

/*!
 * \brief Set the framing used for a set of codecs
 *
 * \param codecs Codecs structure to set framing on
 * \param framing The framing value to set on the codecs
 *
 * \since 13.0.0
 */
void ast_rtp_codecs_set_framing(struct ast_rtp_codecs *codecs, unsigned int framing);

/*!
 * \brief Get the framing used for a set of codecs
 *
 * \param codecs Codecs structure to get the framing from
 *
 * \return The framing to be used for the media stream associated with these codecs
 *
 * \since 13.0.0
 */
unsigned int ast_rtp_codecs_get_framing(struct ast_rtp_codecs *codecs);

/*!
 * \brief Get the sample rate associated with known RTP payload types
 *
 * \param asterisk_format True if the value in format is to be used.
 * \param format An asterisk format
 * \param code from AST_RTP list
 *
 * \return the sample rate if the format was found, zero if it was not found
 *
 * \since 1.8
 */
unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format,
	const struct ast_format *format, int code);

/*!
 * \brief Retrieve all formats that were found
 *
 * \param codecs Codecs structure to look in
 * \param astformats A capabilities structure to put the Asterisk formats in.
 * \param nonastformats An integer to put the non-Asterisk formats in
 *
 * Example usage:
 *
 * \code
 * struct ast_format_cap *astformats = ast_format_cap_alloc_nolock()
 * int nonastformats;
 * ast_rtp_codecs_payload_formats(&codecs, astformats, &nonastformats);
 * \endcode
 *
 * This retrieves all the formats known about in the codecs structure and puts the Asterisk ones in the integer
 * pointed to by astformats and the non-Asterisk ones in the integer pointed to by nonastformats.
 *
 * \since 1.8
 */
void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, struct ast_format_cap *astformats, int *nonastformats);

/*!
 * \brief Retrieve a rx mapped payload type based on whether it is an Asterisk format and the code
 *
 * \param codecs Codecs structure to look in
 * \param asterisk_format Non-zero if the given Asterisk format is present
 * \param format Asterisk format to look for
 * \param code The format to look for
 *
 * \details
 * Find the currently assigned rx mapped payload type based on whether it
 * is an Asterisk format or non-format code.  If one is currently not
 * assigned then create a rx payload type mapping.
 *
 * \return Numerical payload type
 * \retval -1 if could not assign.
 *
 * Example usage:
 *
 * \code
 * int payload = ast_rtp_codecs_payload_code(&codecs, 1, ast_format_ulaw, 0);
 * \endcode
 *
 * This looks for the numerical payload for ULAW in the codecs structure.
 *
 * \since 1.8
 */
int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code);


/*!
 * \brief Retrieve a rx mapped payload type based on whether it is an Asterisk format, the code and the sample rate.
 *
 * \param codecs Codecs structure to look in
 * \param asterisk_format Non-zero if the given Asterisk format is present
 * \param format Asterisk format to look for
 * \param code The format to look for
 * \param sample_rate Non-zero if we want to also match on sample rate.
 *
 * \details
 * Find the currently assigned rx mapped payload type based on whether it
 * is an Asterisk format or non-format code.  If one is currently not
 * assigned then create a rx payload type mapping.
 *
 * \return Numerical payload type
 * \retval -1 if could not assign.
 *
 * Example usage:
 *
 * \code
 * int payload = ast_rtp_codecs_payload_code_sample_rate(&codecs, 0, NULL, AST_RTP_DTMF, 8000);
 * \endcode
 *
 * This looks for the numerical payload for a DTMF type with a sample rate of 8kHz in the codecs structure.
 *
 * \since 22.0.0
 */
int ast_rtp_codecs_payload_code_sample_rate(struct ast_rtp_codecs *codecs, int asterisk_format, struct ast_format *format, int code, unsigned int sample_rate);

/*!
 * \brief Set a payload code for use with a specific Asterisk format
 *
 * \param codecs Codecs structure to manipulate
 * \param code The payload code
 * \param format Asterisk format
 *
 * \retval 0 Payload was set to the given format
 * \retval -1 Payload was in use or could not be set
 *
 * \since 15.0.0
 */
int ast_rtp_codecs_payload_set_rx(struct ast_rtp_codecs *codecs, int code, struct ast_format *format);

/*!
 * \brief Set a payload code with sample rate for use with a specific Asterisk format
 *
 * \param codecs Codecs structure to manipulate
 * \param code The payload code
 * \param format Asterisk format
 * \param sample_rate Sample rate of the format, 0 to use the format's default
 *
 * \retval 0 Payload was set to the given format
 * \retval -1 Payload was in use or could not be set
 *
 * \since 22.0.0
 */
int ast_rtp_codecs_payload_set_rx_sample_rate(struct ast_rtp_codecs *codecs, int code, struct ast_format *format, unsigned int sample_rate);

/*!
 * \brief Retrieve a tx mapped payload type based on whether it is an Asterisk format and the code
 * \since 14.0.0
 *
 * \param codecs Codecs structure to look in
 * \param asterisk_format Non-zero if the given Asterisk format is present
 * \param format Asterisk format to look for
 * \param code The format to look for
 *
 * \return Numerical payload type
 * \retval -1 if not found.
 */
int ast_rtp_codecs_payload_code_tx(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code);

/*!
 * \brief Retrieve a tx mapped payload type based on whether it is an Asterisk format and the code
 * \since 22.0.0
 *
 * \param codecs Codecs structure to look in
 * \param asterisk_format Non-zero if the given Asterisk format is present
 * \param format Asterisk format to look for
 * \param code The format to look for
 * \param sample_rate The sample rate to look for, zero if we don't care
 *
 * \return Numerical payload type
 * \retval -1 if not found.
 */
int ast_rtp_codecs_payload_code_tx_sample_rate(struct ast_rtp_codecs *codecs, int asterisk_format, const struct ast_format *format, int code, unsigned int sample_rate);

/*!
 * \brief Search for the tx payload type in the ast_rtp_codecs structure
 *
 * \param codecs Codecs structure to look in
 * \param payload The payload type format to look for
 *
 * \return Numerical payload type or -1 if unable to find payload in codecs
 *
 * Example usage:
 *
 * \code
 * int payload = ast_rtp_codecs_find_payload_code(&codecs, 0);
 * \endcode
 *
 * This looks for the numerical payload for ULAW in the codecs structure.
 */
int ast_rtp_codecs_find_payload_code(struct ast_rtp_codecs *codecs, int payload);

/*!
 * \brief Retrieve mime subtype information on a payload
 *
 * \param asterisk_format Non-zero to look up using Asterisk format
 * \param format Asterisk format to look up
 * \param code RTP code to look up
 * \param options Additional options that may change the result
 *
 * \return Mime subtype success
 * \retval NULL failure
 *
 * Example usage:
 *
 * \code
 * const char *subtype = ast_rtp_lookup_mime_subtype2(1, ast_format_ulaw, 0, 0);
 * \endcode
 *
 * This looks up the mime subtype for the ULAW format.
 *
 * \since 1.8
 */
const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format,
	const struct ast_format *format, int code, enum ast_rtp_options options);

/*!
 * \brief Convert formats into a string and put them into a buffer
 *
 * \param buf Buffer to put the mime output into
 * \param ast_format_capability Asterisk Formats we are looking up.
 * \param rtp_capability RTP codes that we are looking up
 * \param asterisk_format Non-zero if the ast_format_capability structure is to be used, 0 if rtp_capability is to be used
 * \param options Additional options that may change the result
 *
 * \retval non-NULL success
 * \retval NULL failure
 *
 * Example usage:
 *
 * \code
 * char buf[256] = "";
 * struct ast_format tmp_fmt;
 * struct ast_format_cap *cap = ast_format_cap_alloc_nolock();
 * ast_format_cap_append(cap, ast_format_ulaw, 0);
 * ast_format_cap_append(cap, ast_format_ulaw, 0);
 * char *mime = ast_rtp_lookup_mime_multiple2(&buf, sizeof(buf), cap, 0, 1, 0);
 * ast_format_cap_destroy(cap);
 * \endcode
 *
 * This returns the mime values for ULAW and ALAW in the buffer pointed to by buf.
 *
 * \since 1.8
 */
char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, struct ast_format_cap *ast_format_capability, int rtp_capability, const int asterisk_format, enum ast_rtp_options options);

/*!
 * \brief Begin sending a DTMF digit
 *
 * \param instance The RTP instance to send the DTMF on
 * \param digit What DTMF digit to send
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_dtmf_begin(instance, '1');
 * \endcode
 *
 * This starts sending the DTMF '1' on the RTP instance pointed to by instance. It will
 * continue being sent until it is ended.
 *
 * \since 1.8
 */
int ast_rtp_instance_dtmf_begin(struct ast_rtp_instance *instance, char digit);

/*!
 * \brief Stop sending a DTMF digit
 *
 * \param instance The RTP instance to stop the DTMF on
 * \param digit What DTMF digit to stop
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_dtmf_end(instance, '1');
 * \endcode
 *
 * This stops sending the DTMF '1' on the RTP instance pointed to by instance.
 *
 * \since 1.8
 */
int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit);
int ast_rtp_instance_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration);

/*!
 * \brief Set the DTMF mode that should be used
 *
 * \param instance the RTP instance to set DTMF mode on
 * \param dtmf_mode The DTMF mode that is in use
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_dtmf_mode_set(instance, AST_RTP_DTMF_MODE_RFC2833);
 * \endcode
 *
 * This sets the RTP instance to use RFC2833 for DTMF transmission and receiving.
 *
 * \since 1.8
 */
int ast_rtp_instance_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode);

/*!
 * \brief Get the DTMF mode of an RTP instance
 *
 * \param instance The RTP instance to get the DTMF mode of
 *
 * \return DTMF mode
 *
 * Example usage:
 *
 * \code
 * enum ast_rtp_dtmf_mode dtmf_mode = ast_rtp_instance_dtmf_mode_get(instance);
 * \endcode
 *
 * This gets the DTMF mode set on the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
enum ast_rtp_dtmf_mode ast_rtp_instance_dtmf_mode_get(struct ast_rtp_instance *instance);

/*!
 * \brief Indicate that the RTP marker bit should be set on an RTP stream
 *
 * \param instance Instance that the new media source is feeding into
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_update_source(instance);
 * \endcode
 *
 * This indicates that the source of media that is feeding the instance pointed to by
 * instance has been updated and that the marker bit should be set.
 *
 * \since 1.8
 */
void ast_rtp_instance_update_source(struct ast_rtp_instance *instance);

/*!
 * \brief Indicate a new source of audio has dropped in and the ssrc should change
 *
 * \param instance Instance that the new media source is feeding into
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_change_source(instance);
 * \endcode
 *
 * This indicates that the source of media that is feeding the instance pointed to by
 * instance has changed and that the marker bit should be set and the SSRC updated.
 *
 * \since 1.8
 */
void ast_rtp_instance_change_source(struct ast_rtp_instance *instance);

/*!
 * \brief Set QoS parameters on an RTP session
 *
 * \param instance Instance to set the QoS parameters on
 * \param tos Terms of service value
 * \param cos Class of service value
 * \param desc What is setting the QoS values
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_qos(instance, 0, 0, "Example");
 * \endcode
 *
 * This sets the TOS and COS values to 0 on the instance pointed to by instance.
 *
 * \since 1.8
 */
int ast_rtp_instance_set_qos(struct ast_rtp_instance *instance, int tos, int cos, const char *desc);

/*!
 * \brief Stop an RTP instance
 *
 * \param instance Instance that media is no longer going to at this time
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_stop(instance);
 * \endcode
 *
 * This tells the RTP engine being used for the instance pointed to by instance
 * that media is no longer going to it at this time, but may in the future.
 *
 * \since 1.8
 */
void ast_rtp_instance_stop(struct ast_rtp_instance *instance);

/*!
 * \brief Get the file descriptor for an RTP session (or RTCP)
 *
 * \param instance Instance to get the file descriptor for
 * \param rtcp Whether to retrieve the file descriptor for RTCP or not
 *
 * \retval fd success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * int rtp_fd = ast_rtp_instance_fd(instance, 0);
 * \endcode
 *
 * This retrieves the file descriptor for the socket carrying media on the instance
 * pointed to by instance.
 *
 * \since 1.8
 */
int ast_rtp_instance_fd(struct ast_rtp_instance *instance, int rtcp);

/*!
 * \brief Get the RTP glue that binds a channel to the RTP engine
 *
 * \param type Name of the glue we want
 *
 * \retval non-NULL success
 * \retval NULL failure
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_glue *glue = ast_rtp_instance_get_glue("Example");
 * \endcode
 *
 * This retrieves the RTP glue that has the name 'Example'.
 *
 * \since 1.8
 */
struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type);

/*!
 * \brief Get the unique ID of the channel that owns this RTP instance
 *
 * Note that this should remain valid for the lifetime of the RTP instance.
 *
 * \param instance The RTP instance
 *
 * \return The unique ID of the channel
 * \retval Empty string if no channel owns this RTP instance
 *
 * \since 12
 */
const char *ast_rtp_instance_get_channel_id(struct ast_rtp_instance *instance);

/*!
 * \brief Set the channel that owns this RTP instance
 *
 * \param instance The RTP instance
 * \param uniqueid The uniqueid of the channel
 *
 * \since 12
 */
void ast_rtp_instance_set_channel_id(struct ast_rtp_instance *instance, const char *uniqueid);

/*!
 * \brief Get the other RTP instance that an instance is bridged to
 *
 * \param instance The RTP instance that we want
 *
 * \retval non-NULL success
 * \retval NULL failure
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_instance *bridged = ast_rtp_instance_get_bridged(instance0);
 * \endcode
 *
 * This gets the RTP instance that instance0 is bridged to.
 *
 * \since 1.8
 */
struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance);

/*!
 * \brief Set the other RTP instance that an instance is bridged to
 *
 * \param instance The RTP instance that we want to set the bridged value on
 * \param bridged The RTP instance they are bridged to
 *
 * \since 12
 */
void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged);

/*!
 * \brief Make two channels compatible for early bridging
 *
 * \param c_dst Destination channel to copy to
 * \param c_src Source channel to copy from
 *
 * \since 1.8
 */
void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c_dst, struct ast_channel *c_src);

/*!
 * \brief Early bridge two channels that use RTP instances
 *
 * \param c0 First channel part of the bridge
 * \param c1 Second channel part of the bridge
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * \note This should only be used by channel drivers in their technology declaration.
 *
 * \since 1.8
 */
int ast_rtp_instance_early_bridge(struct ast_channel *c0, struct ast_channel *c1);

/*!
 * \brief Initialize RED support on an RTP instance
 *
 * \param instance The instance to initialize RED support on
 * \param buffer_time How long to buffer before sending
 * \param payloads Payload values
 * \param generations Number of generations
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * \since 1.8
 */
int ast_rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations);

/*!
 * \brief Buffer a frame in an RTP instance for RED
 *
 * \param instance The instance to buffer the frame on
 * \param frame Frame that we want to buffer
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * \since 1.8
 */
int ast_rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame);

/*!
 * \brief Retrieve statistics about an RTP instance
 *
 * \param instance Instance to get statistics on
 * \param stats Structure to put results into
 * \param stat What statistic(s) to retrieve
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_instance_stats stats;
 * ast_rtp_instance_get_stats(instance, &stats, AST_RTP_INSTANCE_STAT_ALL);
 * \endcode
 *
 * This retrieves all statistics the underlying RTP engine supports and puts the values into the
 * stats structure.
 *
 * \since 1.8
 */
int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat);

/*!
 * \brief Set standard statistics from an RTP instance on a channel
 *
 * \param chan Channel to set the statistics on
 * \param instance The RTP instance that statistics will be retrieved from
 *
 * \note Absolutely _NO_ channel locks should be held before calling this function.
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_stats_vars(chan, rtp);
 * \endcode
 *
 * This retrieves standard statistics from the RTP instance rtp and sets it on the channel pointed to
 * by chan.
 *
 * \since 1.8
 */
void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance);

/*!
 * \brief Retrieve quality statistics about an RTP instance
 *
 * \param instance Instance to get statistics on
 * \param field What quality statistic to retrieve
 * \param buf What buffer to put the result into
 * \param size Size of the above buffer
 *
 * \retval non-NULL success
 * \retval NULL failure
 *
 * Example usage:
 *
 * \code
 * char quality[AST_MAX_USER_FIELD];
 * ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, &buf, sizeof(buf));
 * \endcode
 *
 * This retrieves general quality statistics and places a text representation into the buf pointed to by buf.
 *
 * \since 1.8
 */
char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_rtp_instance_stat_field field, char *buf, size_t size);

/*!
 * \brief Request that the underlying RTP engine provide audio frames in a specific format
 *
 * \param instance The RTP instance to change read format on
 * \param format Format that frames are wanted in
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * struct ast_format tmp_fmt;
 * ast_rtp_instance_set_read_format(instance, ast_format_ulaw);
 * \endcode
 *
 * This requests that the RTP engine provide audio frames in the ULAW format.
 *
 * \since 1.8
 */
int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format);

/*!
 * \brief Tell underlying RTP engine that audio frames will be provided in a specific format
 *
 * \param instance The RTP instance to change write format on
 * \param format Format that frames will be provided in
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * struct ast_format tmp_fmt;
 * ast_rtp_instance_set_write_format(instance, ast_format_ulaw);
 * \endcode
 *
 * This tells the underlying RTP engine that audio frames will be provided to it in ULAW format.
 *
 * \since 1.8
 */
int ast_rtp_instance_set_write_format(struct ast_rtp_instance *instance, struct ast_format *format);

/*!
 * \brief Request that the underlying RTP engine make two RTP instances compatible with eachother
 *
 * \param chan Our own Asterisk channel
 * \param instance The first RTP instance
 * \param peer The peer Asterisk channel
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_make_compatible(instance, peer);
 * \endcode
 *
 * This makes the RTP instance for 'peer' compatible with 'instance' and vice versa.
 *
 * \since 1.8
 */
int ast_rtp_instance_make_compatible(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_channel *peer);

/*! \brief Request the formats that can be transcoded
 *
 * \param instance The RTP instance
 * \param to_endpoint Formats being sent/received towards the endpoint
 * \param to_asterisk Formats being sent/received towards Asterisk
 * \param result capabilities structure to store and return supported formats in.
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_available_formats(instance, to_capabilities, from_capabilities, result_capabilities);
 * \endcode
 *
 * This sees if it is possible to have ulaw communicated to the endpoint but signed linear received into Asterisk.
 *
 * \since 1.8
 */
void ast_rtp_instance_available_formats(struct ast_rtp_instance *instance, struct ast_format_cap *to_endpoint, struct ast_format_cap *to_asterisk, struct ast_format_cap *result);

/*!
 * \brief Indicate to the RTP engine that packets are now expected to be sent/received on the RTP instance
 *
 * \param instance The RTP instance
 *
 * \retval 0 success
 * \retval -1 failure
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_activate(instance);
 * \endcode
 *
 * This tells the underlying RTP engine of instance that packets will now flow.
 *
 * \since 1.8
 */
int ast_rtp_instance_activate(struct ast_rtp_instance *instance);

/*!
 * \brief Request that the underlying RTP engine send a STUN BIND request
 *
 * \param instance The RTP instance
 * \param suggestion The suggested destination
 * \param username Optionally a username for the request
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_stun_request(instance, NULL, NULL);
 * \endcode
 *
 * This requests that the RTP engine send a STUN BIND request on the session pointed to by
 * 'instance'.
 *
 * \since 1.8
 */
void ast_rtp_instance_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);

/*!
 * \brief Set the RTP timeout value
 *
 * \param instance The RTP instance
 * \param timeout Value to set the timeout to
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_timeout(instance, 5000);
 * \endcode
 *
 * This sets the RTP timeout value on 'instance' to be 5000.
 *
 * \since 1.8
 */
void ast_rtp_instance_set_timeout(struct ast_rtp_instance *instance, int timeout);

/*!
 * \brief Set the RTP timeout value for when the instance is on hold
 *
 * \param instance The RTP instance
 * \param timeout Value to set the timeout to
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_hold_timeout(instance, 5000);
 * \endcode
 *
 * This sets the RTP hold timeout value on 'instance' to be 5000.
 *
 * \since 1.8
 */
void ast_rtp_instance_set_hold_timeout(struct ast_rtp_instance *instance, int timeout);

/*!
 * \brief Set the RTP keepalive interval
 *
 * \param instance The RTP instance
 * \param timeout Value to set the keepalive interval to
 *
 * Example usage:
 *
 * \code
 * ast_rtp_instance_set_keepalive(instance, 5000);
 * \endcode
 *
 * This sets the RTP keepalive interval on 'instance' to be 5000.
 *
 * \since 1.8
 */
void ast_rtp_instance_set_keepalive(struct ast_rtp_instance *instance, int timeout);

/*!
 * \brief Get the RTP timeout value
 *
 * \param instance The RTP instance
 *
 * \return timeout value
 *
 * Example usage:
 *
 * \code
 * int timeout = ast_rtp_instance_get_timeout(instance);
 * \endcode
 *
 * This gets the RTP timeout value for the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
int ast_rtp_instance_get_timeout(struct ast_rtp_instance *instance);

/*!
 * \brief Get the RTP timeout value for when an RTP instance is on hold
 *
 * \param instance The RTP instance
 *
 * \return timeout value
 *
 * Example usage:
 *
 * \code
 * int timeout = ast_rtp_instance_get_hold_timeout(instance);
 * \endcode
 *
 * This gets the RTP hold timeout value for the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
int ast_rtp_instance_get_hold_timeout(struct ast_rtp_instance *instance);

/*!
 * \brief Get the RTP keepalive interval
 *
 * \param instance The RTP instance
 *
 * \return period Keepalive interval value
 *
 * Example usage:
 *
 * \code
 * int interval = ast_rtp_instance_get_keepalive(instance);
 * \endcode
 *
 * This gets the RTP keepalive interval value for the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
int ast_rtp_instance_get_keepalive(struct ast_rtp_instance *instance);

/*!
 * \brief Get the RTP engine in use on an RTP instance
 *
 * \param instance The RTP instance
 *
 * \return pointer to the engine
 *
 * Example usage:
 *
 * \code
 * struct ast_rtp_engine *engine = ast_rtp_instance_get_engine(instance);
 * \endcode
 *
 * This gets the RTP engine currently in use on the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
struct ast_rtp_engine *ast_rtp_instance_get_engine(struct ast_rtp_instance *instance);

/*!
 * \brief Get the RTP glue in use on an RTP instance
 *
 * \param instance The RTP instance
 *
 * \return pointer to the glue
 *
 * Example:
 *
 * \code
 * struct ast_rtp_glue *glue = ast_rtp_instance_get_active_glue(instance);
 * \endcode
 *
 * This gets the RTP glue currently in use on the RTP instance pointed to by 'instance'.
 *
 * \since 1.8
 */
struct ast_rtp_glue *ast_rtp_instance_get_active_glue(struct ast_rtp_instance *instance);

/*!
 * \brief Send a comfort noise packet to the RTP instance
 *
 * \param instance The RTP instance
 * \param level Magnitude of the noise level
 *
 * \retval 0 Success
 * \retval non-zero Failure
 */
int ast_rtp_instance_sendcng(struct ast_rtp_instance *instance, int level);

/*!
 * \brief Add or replace the SRTP policies for the given RTP instance
 *
 * \param instance the RTP instance
 * \param remote_policy the remote endpoint's policy
 * \param local_policy our policy for this RTP instance's remote endpoint
 * \param rtcp 1 for dedicated RTCP policies
 *
 * \retval 0 Success
 * \retval non-zero Failure
 *
 * \note If no remote policy is provided any existing SRTP policies are left and the new local policy is added
 */
int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy* remote_policy, struct ast_srtp_policy *local_policy, int rtcp);

/*!
 * \brief Obtain the SRTP instance associated with an RTP instance
 *
 * \param instance the RTP instance
 * \param rtcp 1 to request instance for RTCP
 * \return the SRTP instance on success
 * \retval NULL if no SRTP instance exists
 */
struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance, int rtcp);

/*! \brief Custom formats declared in codecs.conf at startup must be communicated to the rtp_engine
 * so their mime type can payload number can be initialized. */
int ast_rtp_engine_load_format(struct ast_format *format);

/*! \brief Formats requiring the use of a format attribute interface must have that
 * interface registered in order for the rtp engine to handle it correctly.  If an
 * attribute interface is unloaded, this function must be called to notify the rtp_engine. */
int ast_rtp_engine_unload_format(struct ast_format *format);

/*!
 * \brief Obtain a pointer to the ICE support present on an RTP instance
 *
 * \param instance the RTP instance
 *
 * \return ICE support if present
 * \retval NULL if no ICE support available
 */
struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *instance);

#ifdef TEST_FRAMEWORK
/*!
 * \brief Obtain a pointer to the test callbacks on an RTP instance
 *
 * \param instance the RTP instance
 *
 * \return test callbacks if present
 * \retval NULL if not present
 */
struct ast_rtp_engine_test *ast_rtp_instance_get_test(struct ast_rtp_instance *instance);
#endif

/*!
 * \brief Obtain a pointer to the DTLS support present on an RTP instance
 *
 * \param instance the RTP instance
 *
 * \return DTLS support if present
 * \retval NULL if no DTLS support available
 */
struct ast_rtp_engine_dtls *ast_rtp_instance_get_dtls(struct ast_rtp_instance *instance);

/*!
 * \brief Parse DTLS related configuration options
 *
 * \param dtls_cfg a DTLS configuration structure
 * \param name name of the configuration option
 * \param value value of the configuration option
 *
 * \retval 0 if handled
 * \retval -1 if not handled
 */
int ast_rtp_dtls_cfg_parse(struct ast_rtp_dtls_cfg *dtls_cfg, const char *name, const char *value);

/*!
 * \brief Validates DTLS related configuration options
 *
 * \param dtls_cfg a DTLS configuration structure
 *
 * \retval 0 if valid
 * \retval -1 if invalid
 */
int ast_rtp_dtls_cfg_validate(struct ast_rtp_dtls_cfg *dtls_cfg);

/*!
 * \brief Copy contents of a DTLS configuration structure
 *
 * \param src_cfg source DTLS configuration structure
 * \param dst_cfg destination DTLS configuration structure
 */
void ast_rtp_dtls_cfg_copy(const struct ast_rtp_dtls_cfg *src_cfg, struct ast_rtp_dtls_cfg *dst_cfg);

/*!
 * \brief Free contents of a DTLS configuration structure
 *
 * \param dtls_cfg a DTLS configuration structure
 */
void ast_rtp_dtls_cfg_free(struct ast_rtp_dtls_cfg *dtls_cfg);

struct ast_json;

/*!
 * \brief Allocate an ao2 ref counted instance of \ref ast_rtp_rtcp_report
 *
 * \param report_blocks The number of report blocks to allocate
 * \return An ao2 ref counted \ref ast_rtp_rtcp_report object on success
 * \retval NULL on error
 */
struct ast_rtp_rtcp_report *ast_rtp_rtcp_report_alloc(unsigned int report_blocks);

/*!
 * \since 12
 * \brief Publish an RTCP message to \ref stasis
 *
 * \param rtp The rtp instance object
 * \param message_type The RTP message type to publish
 * \param report The RTCP report object to publish. This should be an ao2 ref counted
 *  object. This routine will increase the reference count of the object.
 * \param blob Additional JSON objects to publish along with the RTCP information
 */
void ast_rtp_publish_rtcp_message(struct ast_rtp_instance *rtp,
		struct stasis_message_type *message_type,
		struct ast_rtp_rtcp_report *report,
		struct ast_json *blob);

/*!
 * \brief Get the last RTP transmission time
 *
 * \param rtp The instance from which to get the last transmission time
 * \return The last RTP transmission time
 */
time_t ast_rtp_instance_get_last_tx(const struct ast_rtp_instance *rtp);

/*!
 * \brief Set the last RTP transmission time
 *
 * \param rtp The instance on which to set the last transmission time
 * \param time The last transmission time
 */
void ast_rtp_instance_set_last_tx(struct ast_rtp_instance *rtp, time_t time);

/*!
 * \brief Get the last RTP reception time
 *
 * \param rtp The instance from which to get the last reception time
 * \return The last RTP reception time
 */
time_t ast_rtp_instance_get_last_rx(const struct ast_rtp_instance *rtp);

/*!
 * \brief Set the last RTP reception time
 *
 * \param rtp The instance on which to set the last reception time
 * \param time The last reception time
 */
void ast_rtp_instance_set_last_rx(struct ast_rtp_instance *rtp, time_t time);

/*!
 * \brief Retrieve the local SSRC value that we will be using
 *
 * \param rtp The RTP instance
 * \return The SSRC value
 */
unsigned int ast_rtp_instance_get_ssrc(struct ast_rtp_instance *rtp);

/*!
 * \brief Retrieve the CNAME used in RTCP SDES items
 *
 * This is a pointer directly into the RTP struct, not a copy.
 *
 * \param rtp The RTP instance
 * \return the CNAME
 */
const char *ast_rtp_instance_get_cname(struct ast_rtp_instance *rtp);

/*!
 * \brief Request that an RTP instance be bundled with another
 * \since 15.0.0
 *
 * \param child The child RTP instance
 * \param parent The parent RTP instance the child should be bundled with
 *
 * \retval 0 success
 * \retval -1 failure
 */
int ast_rtp_instance_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent);

/*!
 * \brief Set the remote SSRC for an RTP instance
 * \since 15.0.0
 *
 * \param rtp The RTP instance
 * \param ssrc The remote SSRC
 */
void ast_rtp_instance_set_remote_ssrc(struct ast_rtp_instance *rtp, unsigned int ssrc);

/*!
 * \brief Set the stream number for an RTP instance
 * \since 15.0.0
 *
 * \param instance The RTP instance
 * \param stream_num The stream identifier number
 */
void ast_rtp_instance_set_stream_num(struct ast_rtp_instance *instance, int stream_num);

/*! \addtogroup StasisTopicsAndMessages
 * @{
 */

/*!
 * \since 12
 * \brief Message type for an RTCP message sent from this Asterisk instance
 *
 * \return A stasis message type
 */
struct stasis_message_type *ast_rtp_rtcp_sent_type(void);

/*!
 * \since 12
 * \brief Message type for an RTCP message received from some external source
 *
 * \return A stasis message type
 */
struct stasis_message_type *ast_rtp_rtcp_received_type(void);

/*! @} */

#ifdef TEST_FRAMEWORK
/*!
 * \brief Get the maximum size of the receive buffer
 *
 * \param instance The RTP instance
 * \return The recv_buffer max size if it exists, else 0
 */
size_t ast_rtp_instance_get_recv_buffer_max(struct ast_rtp_instance *instance);

/*!
 * \brief Get the current size of the receive buffer
 *
 * \param instance The RTP instance
 * \return The recv_buffer size if it exists, else 0
 */
size_t ast_rtp_instance_get_recv_buffer_count(struct ast_rtp_instance *instance);

/*!
 * \brief Get the current size of the send buffer
 *
 * \param instance The RTP instance
 * \return The send_buffer size if it exists, else 0
 */
size_t ast_rtp_instance_get_send_buffer_count(struct ast_rtp_instance *instance);

/*!
 * \brief Set the schedid for RTCP
 *
 * \param instance The RTP instance
 * \param id The number to set schedid to
 */
void ast_rtp_instance_set_schedid(struct ast_rtp_instance *instance, int id);

/*!
 * \brief Set the number of packets to drop on RTP read
 *
 * \param instance The RTP instance
 * \param num The number of packets to drop
 */
void ast_rtp_instance_drop_packets(struct ast_rtp_instance *instance, int num);

/*!
 * \brief Sends a SR/RR report the next time RTP would be sent
 *
 * \param instance The RTP instance
 */
void ast_rtp_instance_queue_report(struct ast_rtp_instance *instance);

/*!
 * \brief Get the value of sdes_received on the test engine
 *
 * \param instance The RTP instance
 * \retval 1 if sdes_received, else 0
 */
int ast_rtp_instance_get_sdes_received(struct ast_rtp_instance *instance);

/*!
 * \brief Resets all the fields to default values for the test engine
 *
 * \param instance The RTP instance
 */
void ast_rtp_instance_reset_test_engine(struct ast_rtp_instance *instance);
#endif

/*!
 * \brief Convert given stat instance into json format
 * \param stats
 * \return A json format stat
 */
struct ast_json *ast_rtp_convert_stats_json(const struct ast_rtp_instance_stats *stats);

/*!
 * \brief Retrieve statistics about an RTP instance in json format
 * \param instance
 * \return json object of stats
 */
struct ast_json *ast_rtp_instance_get_stats_all_json(struct ast_rtp_instance *instance);

/*!
 * \brief Retrieve the sample rate of a format according to RTP specifications
 * \since 16.7.0
 * \since 17.1.0
 *
 * \param format The media format
 *
 * \return The sample rate
 */
int ast_rtp_get_rate(const struct ast_format *format);

/*!
 * \since 12
 * \brief \ref stasis topic for RTP and RTCP related messages
 *
 * \return A \ref stasis topic
 */
struct stasis_topic *ast_rtp_topic(void);

/*!
 * \brief Determine if a type of payload is already present in mappings.
 * \since 18
 *
 * \param codecs Codecs to be checked for mappings.
 * \param to_match Payload type object to compare against.
 *
 * \note It is assumed that codecs is not locked before calling.
 *
 * \retval 0 not found
 * \retval 1 found
 */
int ast_rtp_payload_mapping_tx_is_present(struct ast_rtp_codecs *codecs, const struct ast_rtp_payload_type *to_match);

/* RTP debug logging category name */
#define AST_LOG_CATEGORY_RTP "rtp"
/* RTP packet debug logging category name */
#define AST_LOG_CATEGORY_RTP_PACKET "rtp_packet"
/* RTCP debug logging category name */
#define AST_LOG_CATEGORY_RTCP "rtcp"
/* RTCP packet debug logging category name */
#define AST_LOG_CATEGORY_RTCP_PACKET "rtcp_packet"
/* DTLS debug logging category name */
#define AST_LOG_CATEGORY_DTLS "dtls"
/* DTLS packet debug logging category name */
#define AST_LOG_CATEGORY_DTLS_PACKET "dtls_packet"
/* ICE debug logging category name */
#define AST_LOG_CATEGORY_ICE "ice"

uintmax_t ast_debug_category_rtp_id(void);
uintmax_t ast_debug_category_rtp_packet_id(void);
uintmax_t ast_debug_category_rtcp_id(void);
uintmax_t ast_debug_category_rtcp_packet_id(void);
uintmax_t ast_debug_category_dtls_id(void);
uintmax_t ast_debug_category_dtls_packet_id(void);
uintmax_t ast_debug_category_ice_id(void);

#define AST_DEBUG_CATEGORY_RTP ast_debug_category_rtp_id() /* RTP debug logging category id */
#define AST_DEBUG_CATEGORY_RTP_PACKET ast_debug_category_rtp_packet_id() /* RTP packet debug logging category id */
#define AST_DEBUG_CATEGORY_RTCP ast_debug_category_rtcp_id() /* RTCP debug logging category id */
#define AST_DEBUG_CATEGORY_RTCP_PACKET ast_debug_category_rtcp_packet_id() /* RTCP packet debug logging category id */
#define AST_DEBUG_CATEGORY_DTLS ast_debug_category_dtls_id() /* DTLS debug logging category id */
#define AST_DEBUG_CATEGORY_DTLS_PACKET ast_debug_category_dtls_packet_id() /* DTLS packet debug logging category id */
#define AST_DEBUG_CATEGORY_ICE ast_debug_category_ice_id() /* ICE debug logging category id */

/*!
 * \brief Log debug level RTP information
 *
 * \param sublevel Debug output sublevel (>= 0)
 * \param ... String format and any associated arguments
 */
#define ast_debug_rtp(sublevel, ...) \
	ast_debug_category(sublevel, AST_DEBUG_CATEGORY_RTP,  __VA_ARGS__)

/* Allow logging of RTP? */
#define ast_debug_rtp_is_allowed \
	ast_debug_category_is_allowed(AST_LOG_CATEGORY_ENABLED, AST_DEBUG_CATEGORY_RTP)

/* Allow logging of RTP packets? */
#define ast_debug_rtp_packet_is_allowed \
	ast_debug_category_is_allowed(AST_LOG_CATEGORY_ENABLED, AST_DEBUG_CATEGORY_RTP_PACKET)

/*!
 * \brief Log debug level RTCP information
 *
 * \param sublevel Debug output sublevel (>= 0)
 * \param ... String format and any associated arguments
 */
#define ast_debug_rtcp(sublevel, ...) \
	ast_debug_category(sublevel, AST_DEBUG_CATEGORY_RTCP, __VA_ARGS__)

/* Allow logging of RTCP? */
#define ast_debug_rtcp_is_allowed \
	ast_debug_category_is_allowed(AST_LOG_CATEGORY_ENABLED, AST_DEBUG_CATEGORY_RTCP)

/* Allow logging of RTCP packets? */
#define ast_debug_rtcp_packet_is_allowed \
	ast_debug_category_is_allowed(AST_LOG_CATEGORY_ENABLED, AST_DEBUG_CATEGORY_RTCP_PACKET)

/*!
 * \brief Log debug level DTLS information
 *
 * \param sublevel Debug output sublevel (>= 0)
 * \param ... String format and any associated arguments
 */
#define ast_debug_dtls(sublevel, ...) \
	ast_debug_category(sublevel, AST_DEBUG_CATEGORY_DTLS, __VA_ARGS__)

/* Allow logging of DTLS packets? */
#define ast_debug_dtls_packet_is_allowed \
	ast_debug_category_is_allowed(AST_LOG_CATEGORY_ENABLED, AST_DEBUG_CATEGORY_DTLS_PACKET)
/*!
 * \brief Log debug level ICE information
 *
 * \param sublevel Debug output sublevel (>= 0)
 * \param ... String format and any associated arguments
 */
#define ast_debug_ice(sublevel, ...) \
	ast_debug_category(sublevel, AST_DEBUG_CATEGORY_ICE, __VA_ARGS__)

#if defined(__cplusplus) || defined(c_plusplus)
}
#endif

#endif /* _ASTERISK_RTP_ENGINE_H */
