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106 lines
4.0 KiB
106 lines
4.0 KiB
4 months ago
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/*
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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_RTP_TRANSPORT_INTERNAL_H_
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#define PC_RTP_TRANSPORT_INTERNAL_H_
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#include <string>
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#include "call/rtp_demuxer.h"
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#include "p2p/base/ice_transport_internal.h"
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#include "pc/session_description.h"
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#include "rtc_base/network_route.h"
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#include "rtc_base/ssl_stream_adapter.h"
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#include "rtc_base/third_party/sigslot/sigslot.h"
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namespace rtc {
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class CopyOnWriteBuffer;
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struct PacketOptions;
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} // namespace rtc
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namespace webrtc {
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// This represents the internal interface beneath SrtpTransportInterface;
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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 sigslot::has_slots<> {
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public:
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virtual ~RtpTransportInternal() = default;
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virtual void SetRtcpMuxEnabled(bool enable) = 0;
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virtual const std::string& transport_name() const = 0;
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// Sets socket options on the underlying RTP or RTCP transports.
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virtual int SetRtpOption(rtc::Socket::Option opt, int value) = 0;
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virtual int SetRtcpOption(rtc::Socket::Option opt, int value) = 0;
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virtual bool rtcp_mux_enabled() const = 0;
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virtual bool IsReadyToSend() const = 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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// Called whenever an RTCP packet is received. There is no equivalent signal
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// for RTP packets because they would be forwarded to the BaseChannel through
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// the RtpDemuxer callback.
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sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;
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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<absl::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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// This method updates the RTP header extension map so that the RTP transport
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// can parse the received packets and identify the MID. This is called by the
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// BaseChannel when setting the content description.
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//
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// TODO(zhihuang): Merging and replacing following methods handling header
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// extensions with SetParameters:
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// UpdateRtpHeaderExtensionMap,
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// UpdateSendEncryptedHeaderExtensionIds,
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// UpdateRecvEncryptedHeaderExtensionIds,
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// CacheRtpAbsSendTimeHeaderExtension,
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virtual void UpdateRtpHeaderExtensionMap(
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const cricket::RtpHeaderExtensions& header_extensions) = 0;
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virtual bool IsSrtpActive() const = 0;
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virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
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RtpPacketSinkInterface* sink) = 0;
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virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
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};
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} // namespace webrtc
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#endif // PC_RTP_TRANSPORT_INTERNAL_H_
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