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Diffstat (limited to '1.2-netsec/include/asterisk/rtp.h')
| -rw-r--r-- | 1.2-netsec/include/asterisk/rtp.h | 172 |
1 files changed, 0 insertions, 172 deletions
diff --git a/1.2-netsec/include/asterisk/rtp.h b/1.2-netsec/include/asterisk/rtp.h deleted file mode 100644 index f1e771e34..000000000 --- a/1.2-netsec/include/asterisk/rtp.h +++ /dev/null @@ -1,172 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 1999 - 2005, 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 "asterisk/frame.h" -#include "asterisk/io.h" -#include "asterisk/sched.h" -#include "asterisk/channel.h" - -#include <netinet/in.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 - -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); - -#ifdef MIDCOM -void ast_rtp_nat_us(struct ast_rtp *rtp, struct sockaddr_in *our_nat); -void ast_rtp_get_their_nat(struct ast_rtp *rtp, struct sockaddr_in *their_nat); -#endif - -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 */ |