webrtc/talk/p2p/base/transportdescriptionfactory.h
henrike@webrtc.org 269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00

84 lines
3.4 KiB
C++

/*
* libjingle
* Copyright 2012 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
#define WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_
#include "webrtc/p2p/base/transportdescription.h"
namespace rtc {
class SSLIdentity;
}
namespace cricket {
struct TransportOptions {
TransportOptions() : ice_restart(false), prefer_passive_role(false) {}
bool ice_restart;
bool prefer_passive_role;
};
// Creates transport descriptions according to the supplied configuration.
// When creating answers, performs the appropriate negotiation
// of the various fields to determine the proper result.
class TransportDescriptionFactory {
public:
// Default ctor; use methods below to set configuration.
TransportDescriptionFactory();
SecurePolicy secure() const { return secure_; }
// The identity to use when setting up DTLS.
rtc::SSLIdentity* identity() const { return identity_; }
// Specifies the transport protocol to be use.
void set_protocol(TransportProtocol protocol) { protocol_ = protocol; }
// Specifies the transport security policy to use.
void set_secure(SecurePolicy s) { secure_ = s; }
// Specifies the identity to use (only used when secure is not SEC_DISABLED).
void set_identity(rtc::SSLIdentity* identity) { identity_ = identity; }
// Creates a transport description suitable for use in an offer.
TransportDescription* CreateOffer(const TransportOptions& options,
const TransportDescription* current_description) const;
// Create a transport description that is a response to an offer.
TransportDescription* CreateAnswer(
const TransportDescription* offer,
const TransportOptions& options,
const TransportDescription* current_description) const;
private:
bool SetSecurityInfo(TransportDescription* description,
ConnectionRole role) const;
TransportProtocol protocol_;
SecurePolicy secure_;
rtc::SSLIdentity* identity_;
};
} // namespace cricket
#endif // WEBRTC_P2P_BASE_TRANSPORTDESCRIPTIONFACTORY_H_