mirror of
https://github.com/servalproject/serval-dna.git
synced 2024-12-23 15:02:27 +00:00
287 lines
9.7 KiB
C
287 lines
9.7 KiB
C
/*
|
|
* Asterisk -- An open source telephony toolkit.
|
|
*
|
|
* Copyright (C) 1999 - 2006, Digium, Inc.
|
|
*
|
|
* Mark Spencer <markster@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 rtp.h
|
|
* \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
|
|
*
|
|
* RTP is defined in RFC 3550.
|
|
*/
|
|
|
|
#ifndef _ASTERISK_RTP_H
|
|
#define _ASTERISK_RTP_H
|
|
|
|
#include <netinet/in.h>
|
|
|
|
#include "asterisk/frame.h"
|
|
#include "asterisk/io.h"
|
|
#include "asterisk/sched.h"
|
|
#include "asterisk/channel.h"
|
|
#include "asterisk/linkedlists.h"
|
|
|
|
#if defined(__cplusplus) || defined(c_plusplus)
|
|
extern "C" {
|
|
#endif
|
|
|
|
/* 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
|
|
|
|
#define MAX_RTP_PT 256
|
|
|
|
enum ast_rtp_options {
|
|
AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
|
|
};
|
|
|
|
enum ast_rtp_get_result {
|
|
/*! Failed to find the RTP structure */
|
|
AST_RTP_GET_FAILED = 0,
|
|
/*! RTP structure exists but true native bridge can not occur so try partial */
|
|
AST_RTP_TRY_PARTIAL,
|
|
/*! RTP structure exists and native bridge can occur */
|
|
AST_RTP_TRY_NATIVE,
|
|
};
|
|
|
|
struct ast_rtp;
|
|
|
|
struct ast_rtp_protocol {
|
|
/*! Get RTP struct, or NULL if unwilling to transfer */
|
|
enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
|
|
/*! Get RTP struct, or NULL if unwilling to transfer */
|
|
enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
|
|
/*! Set RTP peer */
|
|
int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active);
|
|
int (* const get_codec)(struct ast_channel *chan);
|
|
const char * const type;
|
|
AST_LIST_ENTRY(ast_rtp_protocol) list;
|
|
};
|
|
|
|
struct ast_rtp_quality {
|
|
unsigned int local_ssrc; /* Our SSRC */
|
|
unsigned int local_lostpackets; /* Our lost packets */
|
|
double local_jitter; /* Our calculated jitter */
|
|
unsigned int local_count; /* Number of received packets */
|
|
unsigned int remote_ssrc; /* Their SSRC */
|
|
unsigned int remote_lostpackets; /* Their lost packets */
|
|
double remote_jitter; /* Their reported jitter */
|
|
unsigned int remote_count; /* Number of transmitted packets */
|
|
double rtt; /* Round trip time */
|
|
};
|
|
|
|
|
|
#define FLAG_3389_WARNING (1 << 0)
|
|
|
|
typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
|
|
|
|
/*!
|
|
* \brief Get the amount of space required to hold an RTP session
|
|
* \return number of bytes required
|
|
*/
|
|
size_t ast_rtp_alloc_size(void);
|
|
|
|
/*!
|
|
* \brief Initializate a RTP session.
|
|
*
|
|
* \param sched
|
|
* \param io
|
|
* \param rtcpenable
|
|
* \param callbackmode
|
|
* \returns A representation (structure) of an RTP session.
|
|
*/
|
|
struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode);
|
|
|
|
/*!
|
|
* \brief Initializate a RTP session using an in_addr structure.
|
|
*
|
|
* This fuction gets called by ast_rtp_new().
|
|
*
|
|
* \param sched
|
|
* \param io
|
|
* \param rtcpenable
|
|
* \param callbackmode
|
|
* \param in
|
|
* \returns A representation (structure) of an RTP session.
|
|
*/
|
|
struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in);
|
|
|
|
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
|
|
|
|
/*!
|
|
* \since 1.4.26
|
|
* \brief set potential alternate source for RTP media
|
|
*
|
|
* This function may be used to give the RTP stack a hint that there is a potential
|
|
* second source of media. One case where this is used is when the SIP stack receives
|
|
* a REINVITE to which it will be replying with a 491. In such a scenario, the IP and
|
|
* port information in the SDP of that REINVITE lets us know that we may receive media
|
|
* from that source/those sources even though the SIP transaction was unable to be completed
|
|
* successfully
|
|
*
|
|
* \param rtp The RTP structure we wish to set up an alternate host/port on
|
|
* \param alt The address information for the alternate media source
|
|
* \retval void
|
|
*/
|
|
void ast_rtp_set_alt_peer(struct ast_rtp *rtp, struct sockaddr_in *alt);
|
|
|
|
/* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
|
|
int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
|
|
|
|
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
|
|
|
|
struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
|
|
|
|
void ast_rtp_destroy(struct ast_rtp *rtp);
|
|
|
|
void ast_rtp_reset(struct ast_rtp *rtp);
|
|
|
|
void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
|
|
|
|
void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
|
|
|
|
void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
|
|
|
|
int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
|
|
|
|
struct ast_frame *ast_rtp_read(struct ast_rtp *rtp);
|
|
|
|
struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp);
|
|
|
|
int ast_rtp_fd(struct ast_rtp *rtp);
|
|
|
|
int ast_rtcp_fd(struct ast_rtp *rtp);
|
|
|
|
int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
|
|
|
|
int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
|
|
|
|
int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
|
|
|
|
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
|
|
|
|
/*! \brief Indicate that we need to set the marker bit */
|
|
void ast_rtp_new_source(struct ast_rtp *rtp);
|
|
|
|
/*! \brief Indicate that we need to set the marker bit and change the ssrc */
|
|
void ast_rtp_change_source(struct ast_rtp *rtp);
|
|
|
|
/*! \brief Setting RTP payload types from lines in a SDP description: */
|
|
void ast_rtp_pt_clear(struct ast_rtp* rtp);
|
|
/*! \brief Set payload types to defaults */
|
|
void ast_rtp_pt_default(struct ast_rtp* rtp);
|
|
|
|
/*! \brief Copy payload types between RTP structures */
|
|
void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
|
|
|
|
/*! \brief Activate payload type */
|
|
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
|
|
|
|
/*! \brief clear payload type */
|
|
void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
|
|
|
|
/*! \brief Initiate payload type to a known MIME media type for a codec */
|
|
int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
|
|
char *mimeType, char *mimeSubtype,
|
|
enum ast_rtp_options options);
|
|
|
|
/*! \brief Mapping between RTP payload format codes and Asterisk codes: */
|
|
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
|
|
int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
|
|
|
|
void ast_rtp_get_current_formats(struct ast_rtp* rtp,
|
|
int* astFormats, int* nonAstFormats);
|
|
|
|
/*! \brief Mapping an Asterisk code into a MIME subtype (string): */
|
|
const char *ast_rtp_lookup_mime_subtype(int isAstFormat, int code,
|
|
enum ast_rtp_options options);
|
|
|
|
/*! \brief Build a string of MIME subtype names from a capability list */
|
|
char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
|
|
const int isAstFormat, enum ast_rtp_options options);
|
|
|
|
void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
|
|
|
|
int ast_rtp_getnat(struct ast_rtp *rtp);
|
|
|
|
/*! \brief Indicate whether this RTP session is carrying DTMF or not */
|
|
void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
|
|
|
|
/*! \brief Compensate for devices that send RFC2833 packets all at once */
|
|
void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
|
|
|
|
/*! \brief Enable STUN capability */
|
|
void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
|
|
|
|
int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
|
|
|
|
int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
|
|
|
|
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
|
|
|
|
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
|
|
|
|
/*! \brief If possible, create an early bridge directly between the devices without
|
|
having to send a re-invite later */
|
|
int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src);
|
|
|
|
void ast_rtp_stop(struct ast_rtp *rtp);
|
|
|
|
/*! \brief Return RTCP quality string */
|
|
char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual);
|
|
|
|
/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
|
|
int ast_rtcp_send_h261fur(void *data);
|
|
|
|
void ast_rtp_new_init(struct ast_rtp *rtp);
|
|
|
|
void ast_rtp_init(void);
|
|
|
|
int ast_rtp_reload(void);
|
|
|
|
int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
|
|
|
|
struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
|
|
|
|
int ast_rtp_codec_getformat(int pt);
|
|
|
|
/*! \brief Set rtp timeout */
|
|
void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
|
|
/*! \brief Set rtp hold timeout */
|
|
void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
|
|
/*! \brief set RTP keepalive interval */
|
|
void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
|
|
/*! \brief Get RTP keepalive interval */
|
|
int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
|
|
/*! \brief Get rtp hold timeout */
|
|
int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
|
|
/*! \brief Get rtp timeout */
|
|
int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
|
|
/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
|
|
void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
|
|
|
|
#if defined(__cplusplus) || defined(c_plusplus)
|
|
}
|
|
#endif
|
|
|
|
#endif /* _ASTERISK_RTP_H */
|