7914b8cb41
Before this CL, we would negotiate: - No crypto suites for data m= sections. - A full set for audio m= sections. - The full set, minus SRTP_AES128_CM_SHA1_32 for video m= sections. However, this doesn't make sense with BUNDLE, since any DTLS association could end up being used for any type of media. If video is "bundled on" the audio transport (which is typical), it will actually end up using SRTP_AES128_CM_SHA1_32. So, this CL moves the responsibility of deciding SRTP crypto suites out of BaseChannel and into DtlsTransport. The only two possibilities are now "normal set" or "normal set + GCM", if enabled by the PC factory options. This fixes an issue (see linked bug) that was occurring when audio/video were "bundled onto" the data transport. Since the data transport wasn't negotiating any SRTP crypto suites, none were available to use for audio/video, so the application would get black video/no audio. This CL doesn't affect the SDES SRTP crypto suite negotiation; it only affects the negotiation in the DLTS handshake, through the use_srtp extension. BUG=chromium:711243 Review-Url: https://codereview.webrtc.org/2815513012 Cr-Commit-Position: refs/heads/master@{#17810}
109 lines
3.8 KiB
C++
109 lines
3.8 KiB
C++
/*
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* Copyright 2016 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_P2P_BASE_DTLSTRANSPORTINTERNAL_H_
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#define WEBRTC_P2P_BASE_DTLSTRANSPORTINTERNAL_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "webrtc/base/sslstreamadapter.h"
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#include "webrtc/base/stringencode.h"
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#include "webrtc/p2p/base/icetransportinternal.h"
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#include "webrtc/p2p/base/jseptransport.h"
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#include "webrtc/p2p/base/packettransportinternal.h"
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namespace cricket {
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enum PacketFlags {
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PF_NORMAL = 0x00, // A normal packet.
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PF_SRTP_BYPASS = 0x01, // An encrypted SRTP packet; bypass any additional
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// crypto provided by the transport (e.g. DTLS)
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};
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// DtlsTransportInternal is an internal interface that does DTLS, also
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// negotiating SRTP crypto suites so that it may be used for DTLS-SRTP.
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//
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// Once the public interface is supported,
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// (https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface)
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// the DtlsTransportInterface will be split from this class.
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class DtlsTransportInternal : public rtc::PacketTransportInternal {
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public:
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virtual ~DtlsTransportInternal() {}
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virtual DtlsTransportState dtls_state() const = 0;
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virtual const std::string& transport_name() const = 0;
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virtual int component() const = 0;
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virtual bool IsDtlsActive() const = 0;
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virtual bool GetSslRole(rtc::SSLRole* role) const = 0;
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virtual bool SetSslRole(rtc::SSLRole role) = 0;
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// Finds out which DTLS-SRTP cipher was negotiated.
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// TODO(zhihuang): Remove this once all dependencies implement this.
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virtual bool GetSrtpCryptoSuite(int* cipher) = 0;
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// Finds out which DTLS cipher was negotiated.
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// TODO(zhihuang): Remove this once all dependencies implement this.
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virtual bool GetSslCipherSuite(int* cipher) = 0;
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// Gets the local RTCCertificate used for DTLS.
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virtual rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate()
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const = 0;
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virtual bool SetLocalCertificate(
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const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) = 0;
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// Gets a copy of the remote side's SSL certificate.
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virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
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const = 0;
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// Allows key material to be extracted for external encryption.
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virtual bool ExportKeyingMaterial(const std::string& label,
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const uint8_t* context,
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size_t context_len,
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bool use_context,
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uint8_t* result,
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size_t result_len) = 0;
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// Set DTLS remote fingerprint. Must be after local identity set.
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virtual bool SetRemoteFingerprint(const std::string& digest_alg,
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const uint8_t* digest,
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size_t digest_len) = 0;
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// Expose the underneath IceTransport.
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virtual IceTransportInternal* ice_transport() = 0;
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sigslot::signal2<DtlsTransportInternal*, DtlsTransportState> SignalDtlsState;
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// Emitted whenever the Dtls handshake failed on some transport channel.
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sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError;
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// Debugging description of this transport.
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std::string debug_name() const override {
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return transport_name() + " " + rtc::ToString(component());
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}
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protected:
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DtlsTransportInternal() {}
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(DtlsTransportInternal);
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};
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} // namespace cricket
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#endif // WEBRTC_P2P_BASE_DTLSTRANSPORTINTERNAL_H_
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