/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 1999 - 2005, 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 "asterisk/frame.h" #include "asterisk/io.h" #include "asterisk/sched.h" #include "asterisk/channel.h" #include #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 struct ast_rtp_protocol { /* Get RTP struct, or NULL if unwilling to transfer */ struct ast_rtp *(* const get_rtp_info)(struct ast_channel *chan); /* Get RTP struct, or NULL if unwilling to transfer */ struct ast_rtp *(* const get_vrtp_info)(struct ast_channel *chan); /* 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; struct ast_rtp_protocol *next; }; /*! * \brief Structure representing a RTP session. * * RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]" * */ struct ast_rtp; typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data); /*! * \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); void 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); void ast_rtp_destroy(struct ast_rtp *rtp); void ast_rtp_reset(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(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); /* Setting RTP payload types from lines in a SDP description: */ void ast_rtp_pt_clear(struct ast_rtp* rtp); /* Set payload types to defaults */ void ast_rtp_pt_default(struct ast_rtp* rtp); 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); /* 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_offered_from_local(struct ast_rtp* rtp, int local); void ast_rtp_get_current_formats(struct ast_rtp* rtp, int* astFormats, int* nonAstFormats); /* Mapping an Asterisk code into a MIME subtype (string): */ char* ast_rtp_lookup_mime_subtype(int isAstFormat, int code); /* Build a string of MIME subtype names from a capability list */ char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, const int isAstFormat); void ast_rtp_setnat(struct ast_rtp *rtp, int nat); 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); void ast_rtp_stop(struct ast_rtp *rtp); void ast_rtp_init(void); void ast_rtp_reload(void); #if defined(__cplusplus) || defined(c_plusplus) } #endif #endif /* _ASTERISK_RTP_H */