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asterisk/include/asterisk/rtp_engine.h

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/*
* 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"
Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-09 15:31:32 +00:00
#include "asterisk/frame.h"
#include "asterisk/netsock2.h"
#include "asterisk/sched.h"
#include "asterisk/res_srtp.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,
/*!
* \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,
};
/*! 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;
/*! If asterisk_format is set, this is the internal
* asterisk format represented by the payload */
struct ast_format format;
/*! Actual internal RTP specific value of the payload */
int rtp_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 */
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;
};
#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 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 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 ast_sockaddr *sa);
/*! Callback for setting an alternate remote address */
void (*alt_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);
/*! 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, const struct ast_format_cap *cap, int nat_active);
/*! Callback for retrieving codecs that the channel can do. Result returned in result_cap*/
void (*get_codec)(struct ast_channel *chan, struct ast_format_cap *result_cap);
/*! 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 1.8
*/
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_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 1.8
*/
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 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
*/
int ast_rtp_instance_set_remote_address(struct ast_rtp_instance *instance, const struct ast_sockaddr *address);
/*!
* \brief Set the address of an an alternate RTP address to receive from
*
* \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_alt_remote_address(instance, &address);
* \endcode
*
* This changes the alternate remote address that RTP will be sent to on instance to the address given in the sin
* structure.
*
* \since 1.8
*/
int ast_rtp_instance_set_alt_remote_address(struct ast_rtp_instance *instance, const struct ast_sockaddr *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
*/
Merged revisions 293803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@293809 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03 18:43:18 +00:00
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 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
*/
Merged revisions 293803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@293809 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-03 18:43:18 +00:00
void ast_rtp_instance_get_remote_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
*/
int ast_rtp_instance_get_and_cmp_remote_address(struct ast_rtp_instance *instance, struct ast_sockaddr *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
*
* \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 1.8
*/
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 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 1.8
*/
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 1.8
*/
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 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 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 1.8
*/
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, 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 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 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 1.8
*/
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 1.8
*/
struct ast_rtp_payload_type ast_rtp_codecs_payload_lookup(struct ast_rtp_codecs *codecs, int payload);
/*!
* \brief Retrieve the actual ast_format stored on the codecs structure for a specific payload
*
* \param codecs Codecs structure to look in
* \param payload Numerical payload to look up
*
* \retval pointer to format structure on success
* \retval NULL on failure
*
* \since 1.10
*/
struct ast_format *ast_rtp_codecs_get_payload_format(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 format is to be used.
* \param 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, 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 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 Asterisk format is present
* \param format Asterisk format to look for
* \param code The format to look for
*
* \retval Numerical payload
*
* Example usage:
*
* \code
* int payload = ast_rtp_codecs_payload_code(&codecs, 1, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0), 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, const struct ast_format *format, int code);
/*!
* \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
*
* \retval Mime subtype success
* \retval NULL failure
*
* Example usage:
*
* \code
* const char *subtype = ast_rtp_lookup_mime_subtype2(1, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0), 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, 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_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0));
* ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_GSM, 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 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 1.8
*/
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 1.8
*/
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
*
* \retval 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 Bridge two channels that use RTP instances
*
* \param c0 First channel part of the bridge
* \param c1 Second channel part of the bridge
* \param flags Bridging flags
* \param fo If a frame needs to be passed up it is stored here
* \param rc Channel that passed the above frame up
* \param timeoutms How long the channels should be bridged for
*
* \retval Bridge result
*
* \note This should only be used by channel drivers in their technology declaration.
*
* \since 1.8
*/
enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
/*!
* \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 Make two channels compatible for early bridging
*
* \param c0 First channel part of the bridge
* \param c1 Second channel part of the bridge
*
* \since 1.8
*/
void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c0, struct ast_channel *c1);
/*!
* \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
*
* 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_set(&tmp_fmt, AST_FORMAT_ULAW, 0));
* \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_set(&tmp_fmt, AST_FORMAT_ULAW, 0));
* \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 period 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
*
* \retval 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
*
* \retval 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
*
* \retval 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
*
* \retval 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
*
* \retval 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 Get the channel that is associated with an RTP instance while in a bridge
*
* \param instance The RTP instance
*
* \retval pointer to the channel
*
* Example:
*
* \code
* struct ast_channel *chan = ast_rtp_instance_get_chan(instance);
* \endcode
*
* This gets the channel associated with the RTP instance pointed to by 'instance'.
*
* \note This will only return a channel while in a local or remote bridge.
*
* \since 1.8
*/
struct ast_channel *ast_rtp_instance_get_chan(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);
int ast_rtp_instance_add_srtp_policy(struct ast_rtp_instance *instance, struct ast_srtp_policy *policy);
struct ast_srtp *ast_rtp_instance_get_srtp(struct ast_rtp_instance *instance);
/*! \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(const 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(const struct ast_format *format);
#if defined(__cplusplus) || defined(c_plusplus)
}
#endif
#endif /* _ASTERISK_RTP_ENGINE_H */