cf990f53b0
The SrtpTransport takes the SRTP responsibilities from the BaseChannel and SrtpFilter. SrtpTransport is now responsible for setting the crypto keys, protecting and unprotecting the packets. SrtpTransport doesn't know if the keys are from SDES or DTLS handshake. BaseChannel is now only responsible setting the offer/answer for SDES or extracting the key from DtlsTransport and configuring the SrtpTransport. SrtpFilter is used by BaseChannel as a helper for SDES negotiation. BUG=webrtc:7013 Change-Id: If61489dfbdf23481a1f1831ad181fbf45eaadb3e Reviewed-on: https://webrtc-review.googlesource.com/2560 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19977}
116 lines
3.5 KiB
C++
116 lines
3.5 KiB
C++
/*
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* Copyright 2017 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 PC_RTPTRANSPORT_H_
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#define PC_RTPTRANSPORT_H_
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#include "pc/bundlefilter.h"
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#include "pc/rtptransportinternal.h"
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#include "rtc_base/sigslot.h"
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namespace rtc {
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class CopyOnWriteBuffer;
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struct PacketOptions;
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struct PacketTime;
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class PacketTransportInternal;
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} // namespace rtc
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namespace webrtc {
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class RtpTransport : public RtpTransportInternal {
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public:
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RtpTransport(const RtpTransport&) = delete;
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RtpTransport& operator=(const RtpTransport&) = delete;
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explicit RtpTransport(bool rtcp_mux_enabled)
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: rtcp_mux_enabled_(rtcp_mux_enabled) {}
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bool rtcp_mux_enabled() const { return rtcp_mux_enabled_; }
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void SetRtcpMuxEnabled(bool enable) override;
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rtc::PacketTransportInternal* rtp_packet_transport() const override {
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return rtp_packet_transport_;
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}
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void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) override;
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rtc::PacketTransportInternal* rtcp_packet_transport() const override {
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return rtcp_packet_transport_;
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}
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void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) override;
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PacketTransportInterface* GetRtpPacketTransport() const override;
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PacketTransportInterface* GetRtcpPacketTransport() const override;
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// TODO(zstein): Use these RtcpParameters for configuration elsewhere.
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RTCError SetParameters(const RtpTransportParameters& parameters) override;
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RtpTransportParameters GetParameters() const override;
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bool IsWritable(bool rtcp) const override;
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bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) override;
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bool HandlesPayloadType(int payload_type) const override;
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void AddHandledPayloadType(int payload_type) override;
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protected:
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// TODO(zstein): Remove this when we remove RtpTransportAdapter.
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RtpTransportAdapter* GetInternal() override;
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private:
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bool HandlesPacket(const uint8_t* data, size_t len);
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void OnReadyToSend(rtc::PacketTransportInternal* transport);
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// Updates "ready to send" for an individual channel and fires
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// SignalReadyToSend.
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void SetReadyToSend(bool rtcp, bool ready);
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void MaybeSignalReadyToSend();
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bool SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags);
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void OnReadPacket(rtc::PacketTransportInternal* transport,
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const char* data,
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size_t len,
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const rtc::PacketTime& packet_time,
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int flags);
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bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
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bool rtcp_mux_enabled_;
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rtc::PacketTransportInternal* rtp_packet_transport_ = nullptr;
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rtc::PacketTransportInternal* rtcp_packet_transport_ = nullptr;
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bool ready_to_send_ = false;
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bool rtp_ready_to_send_ = false;
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bool rtcp_ready_to_send_ = false;
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RtpTransportParameters parameters_;
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cricket::BundleFilter bundle_filter_;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORT_H_
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