/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 1999 - 2006, Digium, Inc. * * Mark Spencer * * 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 #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 */