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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2017, Digium, Inc.
*
* Mark Michelson <mmichelson@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.
*/
#ifndef _MAIN_SDP_PRIVATE_H
#define _MAIN_SDP_PRIVATE_H
#include "asterisk/stringfields.h"
#include "asterisk/sdp_options.h"
struct ast_sdp_options {
AST_DECLARE_STRING_FIELDS(
/*! Media address to use in SDP */
AST_STRING_FIELD(media_address);
/*! Optional address of the interface media should use. */
AST_STRING_FIELD(interface_address);
/*! SDP origin username */
AST_STRING_FIELD(sdpowner);
/*! SDP session name */
AST_STRING_FIELD(sdpsession);
/*! RTP Engine Name */
AST_STRING_FIELD(rtp_engine);
);
/*! Scheduler context for the media stream types (Mainly for RTP) */
struct ast_sched_context *sched[AST_MEDIA_TYPE_END];
struct {
unsigned int rtp_symmetric:1;
unsigned int udptl_symmetric:1;
unsigned int rtp_ipv6:1;
unsigned int g726_non_standard:1;
unsigned int locally_held:1;
unsigned int rtcp_mux:1;
unsigned int ssrc:1;
};
struct {
unsigned int tos_audio;
unsigned int cos_audio;
unsigned int tos_video;
unsigned int cos_video;
unsigned int udptl_far_max_datagram;
};
enum ast_sdp_options_dtmf dtmf;
enum ast_sdp_options_ice ice;
enum ast_sdp_options_impl impl;
enum ast_sdp_options_encryption encryption;
enum ast_t38_ec_modes udptl_error_correction;
};
#endif /* _MAIN_SDP_PRIVATE_H */
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