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asterisk/include/asterisk/rtp.h
Luigi Rizzo e950538bdd Small improvement to the STUN support so it can be used by 
sockets other than RTP ones.

The main change is a new API function in main/rtp.c (see there
for a description)

    int ast_stun_request(int s, struct sockaddr_in *dst,
        const char *username, struct sockaddr_in *answer)

which can be used to send an STUN request on a socket, and
optionally wait for a reply and store the STUN_MAPPED_ADDRESS
into the 'answer' argument (obviously, the version that
waits for a reply is blocking, but this is no different
from DNS resolutions).

Internally there are minor modifications to let stun_handle_packet()
be somewhat configurable on how to parse the body of responses.

At the moment i am not committing any change to the clients,
but adding STUN client support is extremely simple, e.g. chan_sip.c
could do something like this:

    + add a variable to store the stun server address;

	static struct sockaddr_in stunaddr = { 0, };   /*!< stun server address */

    + add code to parse a config file of the form "stunaddr=my.stun.server.org:3478"
      (not shown for brevity);

    + right after binding the main sip socket, talk to the stun server to
      determine the externally visible address

	    if (stunaddr.sin_addr.s_addr != 0)
		ast_stun_request(sipsock, &stunaddr, NULL, &externip);

      so now 'externip' is set with the externally visible address.

so it is really trivial.

Similarly ast_stun_request could be called when creating the RTP
socket (possibly adding a struct sockaddr_in field in the struct
ast_rtp to store the externalip).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-13 16:22:09 +00:00

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/*
* 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
/*! Maxmum number of payload defintions for a RTP session */
#define MAX_RTP_PT 256
#define FLAG_3389_WARNING (1 << 0)
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;
/*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem
*/
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);
/*! Get RTP struct, or NULL if unwilling to transfer */
enum ast_rtp_get_result (* const get_trtp_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, struct ast_rtp *tpeer, int codecs, int nat_active);
int (* const get_codec)(struct ast_channel *chan);
const char * const type;
AST_LIST_ENTRY(ast_rtp_protocol) list;
};
/*! \brief RTCP quality report storage */
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 */
};
/*! RTP callback structure */
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);
/* 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);
/*! Destroy RTP session */
void ast_rtp_destroy(struct ast_rtp *rtp);
void ast_rtp_reset(struct ast_rtp *rtp);
/*! Stop RTP session, do not destroy structure */
void ast_rtp_stop(struct ast_rtp *rtp);
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_setqos(struct ast_rtp *rtp, int tos, int cos);
/*! \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);
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
void 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);
/*! \brief Generic STUN request
* send a generic stun request to the server specified.
* \param s the socket used to send the request
* \param dst the address of the STUN server
* \param username if non null, add the username in the request
* \param answer if non null, the function waits for a response and
* puts here the externally visible address.
* \return 0 on success, other values on error.
* The interface it may change in the future.
*/
int ast_stun_request(int s, struct sockaddr_in *dst,
const char *username, struct sockaddr_in *answer);
/*! \brief Send STUN request for an RTP socket
* Deprecated, this is just a wrapper for ast_rtp_stun_request()
*/
void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
/*! \brief The RTP bridge.
\arg \ref AstRTPbridge
*/
int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
/*! \brief Register an RTP channel client */
int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
/*! \brief Unregister an RTP channel client */
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 *c0, struct ast_channel *c1);
/*! \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_init(void); /*! Initialize RTP subsystem */
int ast_rtp_reload(void); /*! reload rtp configuration */
void ast_rtp_new_init(struct ast_rtp *rtp);
/*! Set codec preference */
int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
/*! Get codec preference */
struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
/*! get format from predefined dynamic payload format */
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 */
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