cd3fc5d90c
The DtlsSrtpTransport takes the reponsiblity of setting up DTLS-SRTP from the BaseChannel. The BaseChannel doesn't handle the signals from the P2P layer transport anymore. The RtpTransport handles the signals from the PacketTransportInternal and the DtlsSrtpTransport handles the DTLS-specific signals and determines when to extract the keys and setting the parameters. In channel_unittests.cc, call from DTLS to SDES is expected to fail since the fallback from DTLS to SDES is not supported. Bug: webrtc:7013 Change-Id: I0a54e017986f5a8ae9710e79643a4651bef3c38f Reviewed-on: https://webrtc-review.googlesource.com/24702 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20941}
90 lines
3.4 KiB
C++
90 lines
3.4 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_RTPTRANSPORTINTERNAL_H_
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#define PC_RTPTRANSPORTINTERNAL_H_
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#include <string>
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#include "api/ortc/rtptransportinterface.h"
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#include "p2p/base/icetransportinternal.h"
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#include "rtc_base/networkroute.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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} // namespace rtc
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namespace webrtc {
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// This represents the internal interface beneath RtpTransportInterface;
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// it is not accessible to API consumers but is accessible to internal classes
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// in order to send and receive RTP and RTCP packets belonging to a single RTP
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// session. Additional convenience and configuration methods are also provided.
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class RtpTransportInternal : public RtpTransportInterface,
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public sigslot::has_slots<> {
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public:
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virtual void SetRtcpMuxEnabled(bool enable) = 0;
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// TODO(zstein): Remove PacketTransport setters. Clients should pass these
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// in to constructors instead and construct a new RtpTransportInternal instead
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// of updating them.
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virtual bool rtcp_mux_enabled() const = 0;
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virtual rtc::PacketTransportInternal* rtp_packet_transport() const = 0;
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virtual void SetRtpPacketTransport(rtc::PacketTransportInternal* rtp) = 0;
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virtual rtc::PacketTransportInternal* rtcp_packet_transport() const = 0;
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virtual void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) = 0;
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// Called whenever a transport's ready-to-send state changes. The argument
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// is true if all used transports are ready to send. This is more specific
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// than just "writable"; it means the last send didn't return ENOTCONN.
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sigslot::signal1<bool> SignalReadyToSend;
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// TODO(zstein): Consider having two signals - RtpPacketReceived and
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// RtcpPacketReceived.
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// The first argument is true for RTCP packets and false for RTP packets.
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sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
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SignalPacketReceived;
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// Called whenever the network route of the P2P layer transport changes.
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// The argument is an optional network route.
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sigslot::signal1<rtc::Optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
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// Called whenever a transport's writable state might change. The argument is
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// true if the transport is writable, otherwise it is false.
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sigslot::signal1<bool> SignalWritableState;
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sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
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virtual bool IsWritable(bool rtcp) const = 0;
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// TODO(zhihuang): Pass the |packet| by copy so that the original data
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// wouldn't be modified.
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virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) = 0;
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virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) = 0;
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virtual bool HandlesPayloadType(int payload_type) const = 0;
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virtual void AddHandledPayloadType(int payload_type) = 0;
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};
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} // namespace webrtc
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#endif // PC_RTPTRANSPORTINTERNAL_H_
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