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https://github.com/servalproject/serval-dna.git
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287 lines
9.7 KiB
C
287 lines
9.7 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2006, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file rtp.h
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* \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
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*
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* RTP is defined in RFC 3550.
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*/
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#ifndef _ASTERISK_RTP_H
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#define _ASTERISK_RTP_H
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#include <netinet/in.h>
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#include "asterisk/frame.h"
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#include "asterisk/io.h"
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#include "asterisk/sched.h"
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#include "asterisk/channel.h"
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#include "asterisk/linkedlists.h"
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#if defined(__cplusplus) || defined(c_plusplus)
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extern "C" {
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#endif
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/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
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/*! DTMF (RFC2833) */
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#define AST_RTP_DTMF (1 << 0)
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/*! 'Comfort Noise' (RFC3389) */
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#define AST_RTP_CN (1 << 1)
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/*! DTMF (Cisco Proprietary) */
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#define AST_RTP_CISCO_DTMF (1 << 2)
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/*! Maximum RTP-specific code */
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#define AST_RTP_MAX AST_RTP_CISCO_DTMF
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#define MAX_RTP_PT 256
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enum ast_rtp_options {
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AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
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};
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enum ast_rtp_get_result {
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/*! Failed to find the RTP structure */
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AST_RTP_GET_FAILED = 0,
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/*! RTP structure exists but true native bridge can not occur so try partial */
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AST_RTP_TRY_PARTIAL,
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/*! RTP structure exists and native bridge can occur */
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AST_RTP_TRY_NATIVE,
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};
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struct ast_rtp;
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struct ast_rtp_protocol {
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/*! Get RTP struct, or NULL if unwilling to transfer */
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enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
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/*! Get RTP struct, or NULL if unwilling to transfer */
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enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
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/*! Set RTP peer */
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int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active);
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int (* const get_codec)(struct ast_channel *chan);
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const char * const type;
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AST_LIST_ENTRY(ast_rtp_protocol) list;
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};
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struct ast_rtp_quality {
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unsigned int local_ssrc; /* Our SSRC */
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unsigned int local_lostpackets; /* Our lost packets */
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double local_jitter; /* Our calculated jitter */
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unsigned int local_count; /* Number of received packets */
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unsigned int remote_ssrc; /* Their SSRC */
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unsigned int remote_lostpackets; /* Their lost packets */
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double remote_jitter; /* Their reported jitter */
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unsigned int remote_count; /* Number of transmitted packets */
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double rtt; /* Round trip time */
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};
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#define FLAG_3389_WARNING (1 << 0)
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typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
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/*!
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* \brief Get the amount of space required to hold an RTP session
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* \return number of bytes required
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*/
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size_t ast_rtp_alloc_size(void);
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/*!
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* \brief Initializate a RTP session.
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*
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* \param sched
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* \param io
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* \param rtcpenable
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* \param callbackmode
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* \returns A representation (structure) of an RTP session.
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*/
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struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode);
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/*!
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* \brief Initializate a RTP session using an in_addr structure.
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*
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* This fuction gets called by ast_rtp_new().
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*
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* \param sched
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* \param io
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* \param rtcpenable
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* \param callbackmode
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* \param in
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* \returns A representation (structure) of an RTP session.
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*/
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struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in);
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void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
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/*!
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* \since 1.4.26
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* \brief set potential alternate source for RTP media
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*
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* This function may be used to give the RTP stack a hint that there is a potential
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* second source of media. One case where this is used is when the SIP stack receives
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* a REINVITE to which it will be replying with a 491. In such a scenario, the IP and
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* port information in the SDP of that REINVITE lets us know that we may receive media
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* from that source/those sources even though the SIP transaction was unable to be completed
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* successfully
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*
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* \param rtp The RTP structure we wish to set up an alternate host/port on
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* \param alt The address information for the alternate media source
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* \retval void
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*/
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void ast_rtp_set_alt_peer(struct ast_rtp *rtp, struct sockaddr_in *alt);
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/* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
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int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
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void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
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struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
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void ast_rtp_destroy(struct ast_rtp *rtp);
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void ast_rtp_reset(struct ast_rtp *rtp);
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void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
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void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
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void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
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int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
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struct ast_frame *ast_rtp_read(struct ast_rtp *rtp);
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struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp);
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int ast_rtp_fd(struct ast_rtp *rtp);
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int ast_rtcp_fd(struct ast_rtp *rtp);
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int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
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int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
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int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
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int ast_rtp_settos(struct ast_rtp *rtp, int tos);
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/*! \brief Indicate that we need to set the marker bit */
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void ast_rtp_new_source(struct ast_rtp *rtp);
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/*! \brief Indicate that we need to set the marker bit and change the ssrc */
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void ast_rtp_change_source(struct ast_rtp *rtp);
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/*! \brief Setting RTP payload types from lines in a SDP description: */
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void ast_rtp_pt_clear(struct ast_rtp* rtp);
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/*! \brief Set payload types to defaults */
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void ast_rtp_pt_default(struct ast_rtp* rtp);
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/*! \brief Copy payload types between RTP structures */
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void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
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/*! \brief Activate payload type */
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void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
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/*! \brief clear payload type */
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void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
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/*! \brief Initiate payload type to a known MIME media type for a codec */
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int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
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char *mimeType, char *mimeSubtype,
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enum ast_rtp_options options);
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/*! \brief Mapping between RTP payload format codes and Asterisk codes: */
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struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
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int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
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void ast_rtp_get_current_formats(struct ast_rtp* rtp,
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int* astFormats, int* nonAstFormats);
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/*! \brief Mapping an Asterisk code into a MIME subtype (string): */
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const char *ast_rtp_lookup_mime_subtype(int isAstFormat, int code,
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enum ast_rtp_options options);
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/*! \brief Build a string of MIME subtype names from a capability list */
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char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
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const int isAstFormat, enum ast_rtp_options options);
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void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
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int ast_rtp_getnat(struct ast_rtp *rtp);
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/*! \brief Indicate whether this RTP session is carrying DTMF or not */
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void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
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/*! \brief Compensate for devices that send RFC2833 packets all at once */
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void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
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/*! \brief Enable STUN capability */
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void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
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int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
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int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
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void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
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int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
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/*! \brief If possible, create an early bridge directly between the devices without
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having to send a re-invite later */
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int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src);
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void ast_rtp_stop(struct ast_rtp *rtp);
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/*! \brief Return RTCP quality string */
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char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual);
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/*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message in SIP */
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int ast_rtcp_send_h261fur(void *data);
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void ast_rtp_new_init(struct ast_rtp *rtp);
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void ast_rtp_init(void);
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int ast_rtp_reload(void);
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int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
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struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
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int ast_rtp_codec_getformat(int pt);
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/*! \brief Set rtp timeout */
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void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
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/*! \brief Set rtp hold timeout */
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void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
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/*! \brief set RTP keepalive interval */
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void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
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/*! \brief Get RTP keepalive interval */
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int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
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/*! \brief Get rtp hold timeout */
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int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
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/*! \brief Get rtp timeout */
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int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
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/* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
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void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
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#if defined(__cplusplus) || defined(c_plusplus)
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}
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#endif
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#endif /* _ASTERISK_RTP_H */
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