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Diffstat (limited to '1.4.23-rc4/include/asterisk/rtp.h')
| -rw-r--r-- | 1.4.23-rc4/include/asterisk/rtp.h | 265 |
1 files changed, 0 insertions, 265 deletions
diff --git a/1.4.23-rc4/include/asterisk/rtp.h b/1.4.23-rc4/include/asterisk/rtp.h deleted file mode 100644 index 870529e5d..000000000 --- a/1.4.23-rc4/include/asterisk/rtp.h +++ /dev/null @@ -1,265 +0,0 @@ -/* - * 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 - -#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); - -/* 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); - -void ast_rtp_new_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 */ |