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+/*
+ * 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 */