Wed Aug 18 22:33:56 2010

Asterisk developer's documentation


rtp.h

Go to the documentation of this file.
00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2006, Digium, Inc.
00005  *
00006  * Mark Spencer <markster@digium.com>
00007  *
00008  * See http://www.asterisk.org for more information about
00009  * the Asterisk project. Please do not directly contact
00010  * any of the maintainers of this project for assistance;
00011  * the project provides a web site, mailing lists and IRC
00012  * channels for your use.
00013  *
00014  * This program is free software, distributed under the terms of
00015  * the GNU General Public License Version 2. See the LICENSE file
00016  * at the top of the source tree.
00017  */
00018 
00019 /*!
00020  * \file rtp.h
00021  * \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
00022  *
00023  * RTP is defined in RFC 3550.
00024  */
00025 
00026 #ifndef _ASTERISK_RTP_H
00027 #define _ASTERISK_RTP_H
00028 
00029 #include "asterisk/network.h"
00030 
00031 #include "asterisk/frame.h"
00032 #include "asterisk/io.h"
00033 #include "asterisk/sched.h"
00034 #include "asterisk/channel.h"
00035 #include "asterisk/linkedlists.h"
00036 
00037 #if defined(__cplusplus) || defined(c_plusplus)
00038 extern "C" {
00039 #endif
00040 
00041 /* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
00042 /*! DTMF (RFC2833) */
00043 #define AST_RTP_DTMF             (1 << 0)
00044 /*! 'Comfort Noise' (RFC3389) */
00045 #define AST_RTP_CN               (1 << 1)
00046 /*! DTMF (Cisco Proprietary) */
00047 #define AST_RTP_CISCO_DTMF       (1 << 2)
00048 /*! Maximum RTP-specific code */
00049 #define AST_RTP_MAX              AST_RTP_CISCO_DTMF
00050 
00051 /*! Maxmum number of payload defintions for a RTP session */
00052 #define MAX_RTP_PT         256
00053 
00054 /*! T.140 Redundancy Maxium number of generations */
00055 #define RED_MAX_GENERATION 5
00056 
00057 #define FLAG_3389_WARNING     (1 << 0)
00058 
00059 enum ast_rtp_options {
00060    AST_RTP_OPT_G726_NONSTANDARD = (1 << 0),
00061 };
00062 
00063 enum ast_rtp_get_result {
00064    /*! Failed to find the RTP structure */
00065    AST_RTP_GET_FAILED = 0,
00066    /*! RTP structure exists but true native bridge can not occur so try partial */
00067    AST_RTP_TRY_PARTIAL,
00068    /*! RTP structure exists and native bridge can occur */
00069    AST_RTP_TRY_NATIVE,
00070 };
00071 
00072 /*! \brief Variables used in ast_rtcp_get function */
00073 enum ast_rtp_qos_vars {
00074    AST_RTP_TXCOUNT,
00075    AST_RTP_RXCOUNT,
00076    AST_RTP_TXJITTER,
00077    AST_RTP_RXJITTER,
00078    AST_RTP_RXPLOSS,
00079    AST_RTP_TXPLOSS,
00080    AST_RTP_RTT
00081 };
00082 
00083 struct ast_rtp;
00084 /*! T.140 Redundancy structure*/
00085 struct rtp_red;
00086 
00087 /*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem 
00088 */
00089 struct ast_rtp_protocol {
00090    /*! Get RTP struct, or NULL if unwilling to transfer */
00091    enum ast_rtp_get_result (* const get_rtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
00092    /*! Get RTP struct, or NULL if unwilling to transfer */
00093    enum ast_rtp_get_result (* const get_vrtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
00094    /*! Get RTP struct, or NULL if unwilling to transfer */
00095    enum ast_rtp_get_result (* const get_trtp_info)(struct ast_channel *chan, struct ast_rtp **rtp);
00096    /*! Set RTP peer */
00097    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);
00098    int (* const get_codec)(struct ast_channel *chan);
00099    const char * const type;
00100    AST_LIST_ENTRY(ast_rtp_protocol) list;
00101 };
00102 
00103 enum ast_rtp_quality_type {
00104    RTPQOS_SUMMARY = 0,
00105    RTPQOS_JITTER,
00106    RTPQOS_LOSS,
00107    RTPQOS_RTT
00108 };
00109 
00110 /*! \brief RTCP quality report storage */
00111 struct ast_rtp_quality {
00112    unsigned int local_ssrc;          /*!< Our SSRC */
00113    unsigned int local_lostpackets;   /*!< Our lost packets */
00114    double       local_jitter;        /*!< Our calculated jitter */
00115    unsigned int local_count;         /*!< Number of received packets */
00116    unsigned int remote_ssrc;         /*!< Their SSRC */
00117    unsigned int remote_lostpackets;  /*!< Their lost packets */
00118    double       remote_jitter;       /*!< Their reported jitter */
00119    unsigned int remote_count;        /*!< Number of transmitted packets */
00120    double       rtt;                 /*!< Round trip time */
00121 };
00122 
00123 /*! RTP callback structure */
00124 typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
00125 
00126 /*!
00127  * \brief Get the amount of space required to hold an RTP session
00128  * \return number of bytes required
00129  */
00130 size_t ast_rtp_alloc_size(void);
00131 
00132 /*!
00133  * \brief Initializate a RTP session.
00134  *
00135  * \param sched
00136  * \param io
00137  * \param rtcpenable
00138  * \param callbackmode
00139  * \returns A representation (structure) of an RTP session.
00140  */
00141 struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode);
00142 
00143 /*!
00144  * \brief Initializate a RTP session using an in_addr structure.
00145  *
00146  * This fuction gets called by ast_rtp_new().
00147  *
00148  * \param sched
00149  * \param io
00150  * \param rtcpenable
00151  * \param callbackmode
00152  * \param in
00153  * \returns A representation (structure) of an RTP session.
00154  */
00155 struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in);
00156 
00157 void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
00158 
00159 /*! 
00160  * \since 1.4.26
00161  * \brief set potential alternate source for RTP media
00162  *
00163  * This function may be used to give the RTP stack a hint that there is a potential
00164  * second source of media. One case where this is used is when the SIP stack receives
00165  * a REINVITE to which it will be replying with a 491. In such a scenario, the IP and
00166  * port information in the SDP of that REINVITE lets us know that we may receive media
00167  * from that source/those sources even though the SIP transaction was unable to be completed
00168  * successfully
00169  *
00170  * \param rtp The RTP structure we wish to set up an alternate host/port on
00171  * \param alt The address information for the alternate media source
00172  * \retval void
00173  */
00174 void ast_rtp_set_alt_peer(struct ast_rtp *rtp, struct sockaddr_in *alt);
00175 
00176 /* Copies from rtp to them and returns 1 if there was a change or 0 if it was already the same */
00177 int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
00178 
00179 void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
00180 
00181 struct ast_rtp *ast_rtp_get_bridged(struct ast_rtp *rtp);
00182 
00183 /*! Destroy RTP session */
00184 void ast_rtp_destroy(struct ast_rtp *rtp);
00185 
00186 void ast_rtp_reset(struct ast_rtp *rtp);
00187 
00188 /*! Stop RTP session, do not destroy structure */
00189 void ast_rtp_stop(struct ast_rtp *rtp);
00190 
00191 void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
00192 
00193 void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
00194 
00195 int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
00196 
00197 struct ast_frame *ast_rtp_read(struct ast_rtp *rtp);
00198 
00199 struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp);
00200 
00201 int ast_rtp_fd(struct ast_rtp *rtp);
00202 
00203 int ast_rtcp_fd(struct ast_rtp *rtp);
00204 
00205 int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
00206 
00207 int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
00208 
00209 int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
00210 
00211 int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
00212 
00213 /*! \brief Indicate that we need to set the marker bit */
00214 void ast_rtp_new_source(struct ast_rtp *rtp);
00215 
00216 /*! \brief Indicate that we need to set the marker bit and change the ssrc */
00217 void ast_rtp_change_source(struct ast_rtp *rtp);
00218 
00219 /*! \brief  Setting RTP payload types from lines in a SDP description: */
00220 void ast_rtp_pt_clear(struct ast_rtp* rtp);
00221 /*! \brief Set payload types to defaults */
00222 void ast_rtp_pt_default(struct ast_rtp* rtp);
00223 
00224 /*! \brief Copy payload types between RTP structures */
00225 void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src);
00226 
00227 /*! \brief Activate payload type */
00228 void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
00229 
00230 /*! \brief clear payload type */
00231 void ast_rtp_unset_m_type(struct ast_rtp* rtp, int pt);
00232 
00233 /*! \brief Initiate payload type to a known MIME media type for a codec */
00234 int ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
00235               char *mimeType, char *mimeSubtype,
00236               enum ast_rtp_options options);
00237 
00238 /*! \brief  Mapping between RTP payload format codes and Asterisk codes: */
00239 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
00240 int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
00241 
00242 void ast_rtp_get_current_formats(struct ast_rtp* rtp,
00243               int* astFormats, int* nonAstFormats);
00244 
00245 /*! \brief  Mapping an Asterisk code into a MIME subtype (string): */
00246 const char *ast_rtp_lookup_mime_subtype(int isAstFormat, int code,
00247                enum ast_rtp_options options);
00248 
00249 /*! \brief Build a string of MIME subtype names from a capability list */
00250 char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
00251                const int isAstFormat, enum ast_rtp_options options);
00252 
00253 void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
00254 
00255 int ast_rtp_getnat(struct ast_rtp *rtp);
00256 
00257 /*! \brief Indicate whether this RTP session is carrying DTMF or not */
00258 void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf);
00259 
00260 /*! \brief Compensate for devices that send RFC2833 packets all at once */
00261 void ast_rtp_setdtmfcompensate(struct ast_rtp *rtp, int compensate);
00262 
00263 /*! \brief Enable STUN capability */
00264 void ast_rtp_setstun(struct ast_rtp *rtp, int stun_enable);
00265 
00266 /*! \brief Generic STUN request
00267  * send a generic stun request to the server specified.
00268  * \param s the socket used to send the request
00269  * \param dst the address of the STUN server
00270  * \param username if non null, add the username in the request
00271  * \param answer if non null, the function waits for a response and
00272  *    puts here the externally visible address.
00273  * \return 0 on success, other values on error.
00274  * The interface it may change in the future.
00275  */
00276 int ast_stun_request(int s, struct sockaddr_in *dst,
00277    const char *username, struct sockaddr_in *answer);
00278 
00279 /*! \brief Send STUN request for an RTP socket
00280  * Deprecated, this is just a wrapper for ast_rtp_stun_request()
00281  */
00282 void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username);
00283 
00284 /*! \brief The RTP bridge.
00285    \arg \ref AstRTPbridge
00286 */
00287 int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
00288 
00289 /*! \brief Register an RTP channel client */
00290 int ast_rtp_proto_register(struct ast_rtp_protocol *proto);
00291 
00292 /*! \brief Unregister an RTP channel client */
00293 void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
00294 
00295 int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media);
00296 
00297 /*! \brief If possible, create an early bridge directly between the devices without
00298            having to send a re-invite later */
00299 int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1);
00300 
00301 /*! \brief Get QOS stats on a RTP channel
00302  * \since 1.6.1
00303  */
00304 int ast_rtp_get_qos(struct ast_rtp *rtp, const char *qos, char *buf, unsigned int buflen);
00305 
00306 /*! \brief Return RTP and RTCP QoS values
00307  * \since 1.6.1
00308  */
00309 unsigned int ast_rtp_get_qosvalue(struct ast_rtp *rtp, enum ast_rtp_qos_vars value);
00310 
00311 /*! \brief Set RTPAUDIOQOS(...) variables on a channel when it is being hung up
00312  * \since 1.6.1
00313  */
00314 void ast_rtp_set_vars(struct ast_channel *chan, struct ast_rtp *rtp);
00315 
00316 /*! \brief Return RTCP quality string 
00317  *
00318  *  \param rtp An rtp structure to get qos information about.
00319  *
00320  *  \param qual An (optional) rtp quality structure that will be 
00321  *              filled with the quality information described in 
00322  *              the ast_rtp_quality structure. This structure is
00323  *              not dependent on any qtype, so a call for any
00324  *              type of information would yield the same results
00325  *              because ast_rtp_quality is not a data type 
00326  *              specific to any qos type.
00327  *
00328  *  \param qtype The quality type you'd like, default should be
00329  *               RTPQOS_SUMMARY which returns basic information
00330  *               about the call. The return from RTPQOS_SUMMARY
00331  *               is basically ast_rtp_quality in a string. The
00332  *               other types are RTPQOS_JITTER, RTPQOS_LOSS and
00333  *               RTPQOS_RTT which will return more specific 
00334  *               statistics.
00335  * \version 1.6.1 added qtype parameter
00336  */
00337 char *ast_rtp_get_quality(struct ast_rtp *rtp, struct ast_rtp_quality *qual, enum ast_rtp_quality_type qtype);
00338 /*! \brief Send an H.261 fast update request. Some devices need this rather than the XML message  in SIP */
00339 int ast_rtcp_send_h261fur(void *data);
00340 
00341 void ast_rtp_init(void);                                      /*! Initialize RTP subsystem */
00342 int ast_rtp_reload(void);                                     /*! reload rtp configuration */
00343 void ast_rtp_new_init(struct ast_rtp *rtp);
00344 
00345 /*! \brief Set codec preference */
00346 void ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);
00347 
00348 /*! \brief Get codec preference */
00349 struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp);
00350 
00351 /*! \brief get format from predefined dynamic payload format */
00352 int ast_rtp_codec_getformat(int pt);
00353 
00354 /*! \brief Set rtp timeout */
00355 void ast_rtp_set_rtptimeout(struct ast_rtp *rtp, int timeout);
00356 /*! \brief Set rtp hold timeout */
00357 void ast_rtp_set_rtpholdtimeout(struct ast_rtp *rtp, int timeout);
00358 /*! \brief set RTP keepalive interval */
00359 void ast_rtp_set_rtpkeepalive(struct ast_rtp *rtp, int period);
00360 /*! \brief Get RTP keepalive interval */
00361 int ast_rtp_get_rtpkeepalive(struct ast_rtp *rtp);
00362 /*! \brief Get rtp hold timeout */
00363 int ast_rtp_get_rtpholdtimeout(struct ast_rtp *rtp);
00364 /*! \brief Get rtp timeout */
00365 int ast_rtp_get_rtptimeout(struct ast_rtp *rtp);
00366 /* \brief Put RTP timeout timers on hold during another transaction, like T.38 */
00367 void ast_rtp_set_rtptimers_onhold(struct ast_rtp *rtp);
00368 
00369 /*! \brief Initalize t.140 redudancy 
00370  * \param ti time between each t140red frame is sent
00371  * \param red_pt payloadtype for RTP packet
00372  * \param pt payloadtype numbers for each generation including primary data
00373  * \param num_gen number of redundant generations, primary data excluded
00374  * \since 1.6.1
00375  */
00376 int rtp_red_init(struct ast_rtp *rtp, int ti, int *pt, int num_gen);
00377 
00378 /*! \brief Buffer t.140 data */
00379 void red_buffer_t140(struct ast_rtp *rtp, struct ast_frame *f);
00380 
00381 
00382 
00383 #if defined(__cplusplus) || defined(c_plusplus)
00384 }
00385 #endif
00386 
00387 #endif /* _ASTERISK_RTP_H */

Generated on Wed Aug 18 22:33:56 2010 for Asterisk - the Open Source PBX by  doxygen 1.4.7