f2d7beb1d4
The DtlsSrtpTransport is designed to take DTLS responsibilities from BaseChannel. DtlsSrtpTransport is responsible for exporting keys from DtlsTransport and setting up the wrapped SrtpTransport. The DtlsSrtpTransport is not hooked up to BaseChannel yet in this CL. Bug: webrtc:7013 Change-Id: I318c00dadf9b1e033ec842de6e1536e9227ab713 Reviewed-on: https://webrtc-review.googlesource.com/6700 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20804}
88 lines
3.3 KiB
C++
88 lines
3.3 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 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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// 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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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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