Files
asterisk/main/sdp_private.h
Mark Michelson 32b3e36c68 SDP: Ensure SDPs "merge" properly.
The gist of this work ensures that when a remote SDP is received, it is
merged properly with the local capabilities. The remote SDP is converted
into a stream topology. That topology is then merged with the current
local topology on the SDP state. That new merged topology is then used
to create an SDP. Finally, adjustments are made to RTP instances based
on knowledge gained from the remote SDP.

There are also a battery of tests in this commit that ensure that some
basic SDP merges work as expected.

While this may not sound like a big change, it has the property that it
caused lots of ancillary changes.

* The remote SDP is no longer stored on the SDP state. Biggest reason:
  there's no need for it. The remote SDP is used at the time it is being
  set and nowhere else.

* Some new SDP APIs were added in order to find attributes and convert
  generic SDP attributes into rtpmap structures.

* Writing tests made me realize that retrieving a value from an SDP
  options structure, the SDP options needs to be made const.

* The SDP state machine was essentially gutted by a previous commit.
  Initially, I attempted to reinstate it, but I found that as it had
  been defined, it was not all that useful. What was more useful was
  knowing the role we play in SDP negotiation, so the SDP state machine
  has been transformed into an indicator of role.

* Rather than storing separate local and joint stream state
  capabilities, it makes more sense to keep track of current stream
  state and update it as things change.

Change-Id: I5938c2be3c6f0a003aa88a39a59e0880f8b2df3d
2017-04-25 13:03:33 -05:00

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1.5 KiB
C

/*
* 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(
/*! Optional media address to use in SDP */
AST_STRING_FIELD(media_address);
/*! SDP origin username */
AST_STRING_FIELD(sdpowner);
/*! SDP session name */
AST_STRING_FIELD(sdpsession);
/*! RTP Engine Name */
AST_STRING_FIELD(rtp_engine);
);
struct {
unsigned int bind_rtp_to_media_address : 1;
unsigned int rtp_symmetric : 1;
unsigned int telephone_event : 1;
unsigned int rtp_ipv6 : 1;
unsigned int g726_non_standard : 1;
unsigned int locally_held : 1;
unsigned int rtcp_mux: 1;
};
struct {
unsigned int tos_audio;
unsigned int cos_audio;
unsigned int tos_video;
unsigned int cos_video;
};
enum ast_sdp_options_ice ice;
enum ast_sdp_options_impl impl;
enum ast_sdp_options_encryption encryption;
};
#endif /* _MAIN_SDP_PRIVATE_H */