8470c2bdea
* Added additional fields to ast_sdp_options. * Re-organized ast_sdp. * Updated field names to correspond to RFC4566 terminology. * Created allocs/frees for SDP children. * Created getters/setters for SDP children where appropriate. * Added ast_sdp_create_from_state. * Refactored res_sdp_translator_pjmedia for changes. Change-Id: Iefbd877af7f5a4d3c74deead1bff8802661b0d48
56 lines
1.5 KiB
C
56 lines
1.5 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2017, Digium, Inc.
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*
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* Mark Michelson <mmichelson@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#ifndef _MAIN_SDP_PRIVATE_H
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#define _MAIN_SDP_PRIVATE_H
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#include "asterisk/stringfields.h"
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#include "asterisk/sdp_options.h"
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struct ast_sdp_options {
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AST_DECLARE_STRING_FIELDS(
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/*! Optional media address to use in SDP */
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AST_STRING_FIELD(media_address);
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/*! SDP origin username */
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AST_STRING_FIELD(sdpowner);
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/*! SDP session name */
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AST_STRING_FIELD(sdpsession);
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/*! RTP Engine Name */
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AST_STRING_FIELD(rtp_engine);
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);
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struct {
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unsigned int bind_rtp_to_media_address : 1;
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unsigned int rtp_symmetric : 1;
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unsigned int telephone_event : 1;
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unsigned int rtp_ipv6 : 1;
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unsigned int g726_non_standard : 1;
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unsigned int locally_held : 1;
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};
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struct {
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unsigned int tos_audio;
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unsigned int cos_audio;
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unsigned int tos_video;
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unsigned int cos_video;
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};
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enum ast_sdp_options_ice ice;
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enum ast_sdp_options_impl impl;
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enum ast_sdp_options_encryption encryption;
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};
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#endif /* _MAIN_SDP_PRIVATE_H */
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