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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, 172 insertions, 0 deletions
diff --git a/1.2-netsec/include/asterisk/rtp.h b/1.2-netsec/include/asterisk/rtp.h new file mode 100644 index 000000000..f1e771e34 --- /dev/null +++ b/1.2-netsec/include/asterisk/rtp.h @@ -0,0 +1,172 @@ +/* + * 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 */ |