mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 22:30:40 +01:00
92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
48 lines
1.8 KiB
C++
48 lines
1.8 KiB
C++
/*
|
|
* Copyright 2017 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef API_ORTC_SRTPTRANSPORTINTERFACE_H_
|
|
#define API_ORTC_SRTPTRANSPORTINTERFACE_H_
|
|
|
|
#include "api/ortc/rtptransportinterface.h"
|
|
#include "api/rtcerror.h"
|
|
#include "media/base/cryptoparams.h"
|
|
|
|
namespace webrtc {
|
|
|
|
// The subclass of the RtpTransport which uses SRTP. The keying information
|
|
// is explicitly passed in from the application.
|
|
//
|
|
// If using SDP and SDES (RFC4568) for signaling, then after applying the
|
|
// answer, the negotiated keying information from the offer and answer would be
|
|
// set and the SRTP would be active.
|
|
//
|
|
// Note that Edge's implementation of ORTC provides a similar API point, called
|
|
// RTCSrtpSdesTransport:
|
|
// https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx
|
|
class SrtpTransportInterface : public RtpTransportInterface {
|
|
public:
|
|
virtual ~SrtpTransportInterface() {}
|
|
|
|
// There are some limitations of the current implementation:
|
|
// 1. Send and receive keys must use the same crypto suite.
|
|
// 2. The keys can't be changed after initially set.
|
|
// 3. The keys must be set before creating a sender/receiver using the SRTP
|
|
// transport.
|
|
// Set the SRTP keying material for sending RTP and RTCP.
|
|
virtual RTCError SetSrtpSendKey(const cricket::CryptoParams& params) = 0;
|
|
|
|
// Set the SRTP keying material for receiving RTP and RTCP.
|
|
virtual RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_ORTC_SRTPTRANSPORTINTERFACE_H_
|