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Diffstat (limited to 'trunk/include/asterisk/rtp.h')
-rw-r--r-- | trunk/include/asterisk/rtp.h | 293 |
1 files changed, 293 insertions, 0 deletions
diff --git a/trunk/include/asterisk/rtp.h b/trunk/include/asterisk/rtp.h new file mode 100644 index 000000000..003ff268f --- /dev/null +++ b/trunk/include/asterisk/rtp.h @@ -0,0 +1,293 @@ +/* + * 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 "asterisk/network.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 + +/*! Maxmum number of payload defintions for a RTP session */ +#define MAX_RTP_PT 256 + +#define FLAG_3389_WARNING (1 << 0) + +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; + +/*! \brief This is the structure that binds a channel (SIP/Jingle/H.323) to the RTP subsystem +*/ +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); + /*! Get RTP struct, or NULL if unwilling to transfer */ + enum ast_rtp_get_result (* const get_trtp_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, struct ast_rtp *tpeer, int codecs, int nat_active); + int (* const get_codec)(struct ast_channel *chan); + const char * const type; + AST_LIST_ENTRY(ast_rtp_protocol) list; +}; + +/*! \brief RTCP quality report storage */ +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 */ +}; + +/*! RTP callback structure */ +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); + +/*! Destroy RTP session */ +void ast_rtp_destroy(struct ast_rtp *rtp); + +void ast_rtp_reset(struct ast_rtp *rtp); + +/*! Stop RTP session, do not destroy structure */ +void ast_rtp_stop(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_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_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc); + +/*! \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); + +/*! \brief Generic STUN request + * send a generic stun request to the server specified. + * \param s the socket used to send the request + * \param dst the address of the STUN server + * \param username if non null, add the username in the request + * \param answer if non null, the function waits for a response and + * puts here the externally visible address. + * \return 0 on success, other values on error. + * The interface it may change in the future. + */ +int ast_stun_request(int s, struct sockaddr_in *dst, + const char *username, struct sockaddr_in *answer); + +/*! \brief Send STUN request for an RTP socket + * Deprecated, this is just a wrapper for ast_rtp_stun_request() + */ +void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username); + +/*! \brief The RTP bridge. + \arg \ref AstRTPbridge +*/ +int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms); + +/*! \brief Register an RTP channel client */ +int ast_rtp_proto_register(struct ast_rtp_protocol *proto); + +/*! \brief Unregister an RTP channel client */ +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 *c0, struct ast_channel *c1); + +/*! \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_init(void); /*! Initialize RTP subsystem */ +int ast_rtp_reload(void); /*! reload rtp configuration */ +void ast_rtp_new_init(struct ast_rtp *rtp); + +/*! Set codec preference */ +int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs); + +/*! Get codec preference */ +struct ast_codec_pref *ast_rtp_codec_getpref(struct ast_rtp *rtp); + +/*! get format from predefined dynamic payload format */ +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 */ |