[asterisk-commits] mmichelson: branch group/issue8824 r186524 - in /team/group/issue8824: includ...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Apr 3 17:26:09 CDT 2009
Author: mmichelson
Date: Fri Apr 3 17:26:01 2009
New Revision: 186524
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=186524
Log:
Add missing files from rtp engine merge.
Added:
team/group/issue8824/include/asterisk/rtp_engine.h (with props)
team/group/issue8824/include/asterisk/stun.h (with props)
team/group/issue8824/main/rtp_engine.c (with props)
team/group/issue8824/main/stun.c (with props)
team/group/issue8824/res/res_rtp_asterisk.c (with props)
Added: team/group/issue8824/include/asterisk/rtp_engine.h
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/include/asterisk/rtp_engine.h?view=auto&rev=186524
==============================================================================
--- team/group/issue8824/include/asterisk/rtp_engine.h (added)
+++ team/group/issue8824/include/asterisk/rtp_engine.h Fri Apr 3 17:26:01 2009
@@ -1,0 +1,1594 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 1999 - 2009, Digium, Inc.
+ *
+ * Mark Spencer <markster at digium.com>
+ * Joshua Colp <jcolp at 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 at 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"
+
+/* Maximum number of payloads supported */
+#define AST_RTP_MAX_PT 256
+
+/* Maximum number of generations */
+#define AST_RED_MAX_GENERATION 5
+
+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,
+ /*! Maximum number of RTP properties supported */
+ 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,
+};
+
+/*! 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,
+};
+
+/* 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 {
+ /*! Is this an Asterisk value */
+ int asterisk_format;
+ /*! Actual internal value of the payload */
+ int code;
+};
+
+/*! 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 */
+ unsigned int txjitter;
+ /*! Jitter on received packets */
+ unsigned int 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 */
+ unsigned int 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;
+};
+
+#define AST_RTP_STAT_SET(current_stat, combined, placement, value) \
+if (stat == current_stat || stat == AST_RTP_INSTANCE_STAT_ALL || (combined >= 0 && combined == current_stat)) { \
+placement = value; \
+if (stat == current_stat) { \
+return 0; \
+} \
+}
+
+#define AST_RTP_STAT_TERMINATOR(combined) \
+if (stat == combined) { \
+return 0; \
+}
+
+/*! 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 sched_context *sched, struct sockaddr_in *sin, 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);
+ /*! Callback to indicate that a new source of media has come in */
+ void (*new_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 */
+ void (*payload_set)(struct ast_rtp_instance *instance, int payload, int astformat, int format);
+ /*! Callback for setting packetization preferences */
+ void (*packetization_set)(struct ast_rtp_instance *instance, struct ast_codec_pref *pref);
+ /*! Callback for setting the remote address that RTP is to be sent to */
+ void (*remote_address_set)(struct ast_rtp_instance *instance, struct sockaddr_in *sin);
+ /*! Callback for changing DTMF mode */
+ int (*dtmf_mode_set)(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode);
+ /*! 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, int format);
+ /*! Callback to set the write format */
+ int (*set_write_format)(struct ast_rtp_instance *instance, int 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 sockaddr_in *suggestion, const char *username);
+ /*! Linked list information */
+ AST_RWLIST_ENTRY(ast_rtp_engine) entry;
+};
+
+/*! Structure that represents codec and packetization information */
+struct ast_rtp_codecs {
+ /*! Codec packetization preferences */
+ struct ast_codec_pref pref;
+ /*! Payloads present */
+ struct ast_rtp_payload_type payloads[AST_RTP_MAX_PT];
+};
+
+/*! 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 Callback for retrieving the RTP instance carrying video
+ * \note This function increases the reference count on the returned RTP instance.
+ */
+ enum ast_rtp_glue_result (*get_vrtp_info)(struct ast_channel *chan, 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.
+ */
+ 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, int codecs, int nat_active);
+ /*! Callback for retrieving codecs that the channel can do */
+ int (*get_codec)(struct ast_channel *chan);
+ /*! Linked list information */
+ AST_RWLIST_ENTRY(ast_rtp_glue) entry;
+};
+
+#define ast_rtp_engine_register(engine) ast_rtp_engine_register2(engine, ast_module_info->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 1.6.3
+ */
+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 1.6.3
+ */
+int ast_rtp_engine_unregister(struct ast_rtp_engine *engine);
+
+#define ast_rtp_glue_register(glue) ast_rtp_glue_register2(glue, ast_module_info->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 1.6.3
+ */
+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 1.6.3
+ */
+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 sin 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 sin 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 1.6.3
+ */
+struct ast_rtp_instance *ast_rtp_instance_new(const char *engine_name, struct sched_context *sched, struct sockaddr_in *sin, 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 1.6.3
+ */
+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 1.6.3
+ */
+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 1.6.3
+ */
+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
+ *
+ * \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 1.6.3
+ */
+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 1.6.3
+ */
+struct ast_frame *ast_rtp_instance_read(struct ast_rtp_instance *instance, int rtcp);
+
+/*!
+ * \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 1.6.3
+ */
+int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *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 1.6.3
+ */
+int ast_rtp_instance_set_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *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
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct sockaddr_in sin;
+ * ast_rtp_instance_get_local_address(instance, &sin);
+ * \endcode
+ *
+ * This gets the local address that we are expecting RTP on and stores it in the 'sin' structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_local_address(struct ast_rtp_instance *instance, struct sockaddr_in *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
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ *
+ * Example usage:
+ *
+ * \code
+ * struct sockaddr_in sin;
+ * ast_rtp_instance_get_remote_address(instance, &sin);
+ * \endcode
+ *
+ * This retrieves the current remote address set on the instance pointed to by instance and puts the value
+ * into the sin structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_instance_get_remote_address(struct ast_rtp_instance *instance, struct sockaddr_in *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 1.6.3
+ */
+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 1.6.3
+ */
+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 1.6.3
+ */
+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
+ *
+ * \retval 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 1.6.3
+ */
+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 1.6.3
+ */
+struct ast_rtp_codecs *ast_rtp_instance_get_codecs(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Clear payload 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 1.6.3
+ */
+void ast_rtp_codecs_payloads_clear(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Set payload information on an RTP instance to the default
+ *
+ * \param codecs The codecs structure to set defaults on
+ * \param instance Optionally the instance that the codecs structure belongs to
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_codecs codecs;
+ * ast_rtp_codecs_payloads_default(&codecs, NULL);
+ * \endcode
+ *
+ * This sets the default payloads on the codecs structure.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_payloads_default(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 dst 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 1.6.3
+ */
+void ast_rtp_codecs_payloads_copy(struct ast_rtp_codecs *src, struct ast_rtp_codecs *dest, struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Record payload 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 1.6.3
+ */
+void ast_rtp_codecs_payloads_set_m_type(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload);
+
+/*!
+ * \brief Record payload 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 strin 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
+ *
+ * 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 1.6.3
+ */
+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 payload type to a known MIME media type for a codec with a specific sample rate
+ *
+ * \param rtp 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 1.6.3
+ */
+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 payload 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 1.6.3
+ */
+void ast_rtp_codecs_payloads_unset(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, int payload);
+
+/*!
+ * \brief Retrieve payload information by payload
+ *
+ * \param codecs Codecs structure to look in
+ * \param payload Numerical payload to look up
+ *
+ * \retval Payload information
+ *
+ * Example usage:
+ *
+ * \code
+ * struct ast_rtp_payload_type payload_type;
+ * payload_type = ast_rtp_codecs_payload_lookup(&codecs, 0);
+ * \endcode
+ *
+ * This looks up the information for payload '0' from the codecs structure.
+ *
+ * \since 1.6.3
+ */
+struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload);
+
+/*!
+ * \brief Get the sample rate associated with known RTP payload types
+ *
+ * \param asterisk_format True if the value in the 'code' parameter is an AST_FORMAT value
+ * \param code Format code, either from AST_FORMAT list or from AST_RTP list
+ *
+ * \return the sample rate if the format was found, zero if it was not found
+ *
+ * \since 1.6.3
+ */
+unsigned int ast_rtp_lookup_sample_rate2(int asterisk_format, int code);
+
+/*!
+ * \brief Retrieve all formats that were found
+ *
+ * \param codecs Codecs structure to look in
+ * \param astFormats An integer to put the Asterisk formats in
+ * \param nonastformats An integer to put the non-Asterisk formats in
+ *
+ * Example usage:
+ *
+ * \code
+ * int astformats, 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 1.6.3
+ */
+void ast_rtp_codecs_payload_formats(struct ast_rtp_codecs *codecs, int *astformats, int *nonastformats);
+
+/*!
+ * \brief Retrieve a payload 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 code is an Asterisk format value
+ * \param code The format to look for
+ *
+ * \retval Numerical payload
+ *
+ * Example usage:
+ *
+ * \code
+ * int payload = ast_rtp_codecs_payload_code(&codecs, 1, AST_FORMAT_ULAW);
+ * \endcode
+ *
+ * This looks for the numerical payload for ULAW in the codecs structure.
+ *
+ * \since 1.6.3
+ */
+int ast_rtp_codecs_payload_code(struct ast_rtp_codecs *codecs, const int asterisk_format, const int code);
+
+/*!
+ * \brief Retrieve mime subtype information on a payload
+ *
+ * \param asterisk_format Non-zero if the given code is an Asterisk format value
+ * \param code Format to look up
+ * \param options Additional options that may change the result
+ *
+ * \retval Mime subtype success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * const char *subtype = ast_rtp_lookup_mime_subtype2(1, AST_FORMAT_ULAW, 0);
+ * \endcode
+ *
+ * This looks up the mime subtype for the ULAW format.
+ *
+ * \since 1.6.3
+ */
+const char *ast_rtp_lookup_mime_subtype2(const int asterisk_format, const 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 capability Formats that we are looking up
+ * \param asterisk_format Non-zero if the given capability are Asterisk format capabilities
+ * \param options Additional options that may change the result
+ *
+ * \retval non-NULL success
+ * \retval NULL failure
+ *
+ * Example usage:
+ *
+ * \code
+ * char buf[256] = "";
+ * char *mime = ast_rtp_lookup_mime_multiple2(&buf, sizeof(buf), AST_FORMAT_ULAW | AST_FORMAT_ALAW, 1, 0);
+ * \endcode
+ *
+ * This returns the mime values for ULAW and ALAW in the buffer pointed to by buf.
+ *
+ * \since 1.6.3
+ */
+char *ast_rtp_lookup_mime_multiple2(struct ast_str *buf, const int capability, const int asterisk_format, enum ast_rtp_options options);
+
+/*!
+ * \brief Set codec packetization preferences
+ *
+ * \param codecs Codecs structure to muck with
+ * \param instance Optionally the instance that the codecs structure belongs to
+ * \param prefs Codec packetization preferences
+ *
+ * Example usage:
+ *
+ * \code
+ * ast_rtp_codecs_packetization_set(&codecs, NULL, &prefs);
+ * \endcode
+ *
+ * This sets the packetization preferences pointed to by prefs on the codecs structure pointed to by codecs.
+ *
+ * \since 1.6.3
+ */
+void ast_rtp_codecs_packetization_set(struct ast_rtp_codecs *codecs, struct ast_rtp_instance *instance, struct ast_codec_pref *prefs);
+
+/*!
+ * \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 1.6.3
+ */
+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 1.6.3
+ */
+int ast_rtp_instance_dtmf_end(struct ast_rtp_instance *instance, char digit);
+
+/*!
+ * \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 1.6.3
+ */
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