1 /*
2  *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef PC_RTP_TRANSPORT_INTERNAL_H_
12 #define PC_RTP_TRANSPORT_INTERNAL_H_
13 
14 #include <string>
15 
16 #include "call/rtp_demuxer.h"
17 #include "p2p/base/ice_transport_internal.h"
18 #include "pc/session_description.h"
19 #include "rtc_base/network_route.h"
20 #include "rtc_base/ssl_stream_adapter.h"
21 #include "rtc_base/third_party/sigslot/sigslot.h"
22 
23 namespace rtc {
24 class CopyOnWriteBuffer;
25 struct PacketOptions;
26 }  // namespace rtc
27 
28 namespace webrtc {
29 
30 // This represents the internal interface beneath SrtpTransportInterface;
31 // it is not accessible to API consumers but is accessible to internal classes
32 // in order to send and receive RTP and RTCP packets belonging to a single RTP
33 // session. Additional convenience and configuration methods are also provided.
34 class RtpTransportInternal : public sigslot::has_slots<> {
35  public:
36   virtual ~RtpTransportInternal() = default;
37 
38   virtual void SetRtcpMuxEnabled(bool enable) = 0;
39 
40   virtual const std::string& transport_name() const = 0;
41 
42   // Sets socket options on the underlying RTP or RTCP transports.
43   virtual int SetRtpOption(rtc::Socket::Option opt, int value) = 0;
44   virtual int SetRtcpOption(rtc::Socket::Option opt, int value) = 0;
45 
46   virtual bool rtcp_mux_enabled() const = 0;
47 
48   virtual bool IsReadyToSend() const = 0;
49 
50   // Called whenever a transport's ready-to-send state changes. The argument
51   // is true if all used transports are ready to send. This is more specific
52   // than just "writable"; it means the last send didn't return ENOTCONN.
53   sigslot::signal1<bool> SignalReadyToSend;
54 
55   // Called whenever an RTCP packet is received. There is no equivalent signal
56   // for RTP packets because they would be forwarded to the BaseChannel through
57   // the RtpDemuxer callback.
58   sigslot::signal2<rtc::CopyOnWriteBuffer*, int64_t> SignalRtcpPacketReceived;
59 
60   // Called whenever the network route of the P2P layer transport changes.
61   // The argument is an optional network route.
62   sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
63 
64   // Called whenever a transport's writable state might change. The argument is
65   // true if the transport is writable, otherwise it is false.
66   sigslot::signal1<bool> SignalWritableState;
67 
68   sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
69 
70   virtual bool IsWritable(bool rtcp) const = 0;
71 
72   // TODO(zhihuang): Pass the |packet| by copy so that the original data
73   // wouldn't be modified.
74   virtual bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
75                              const rtc::PacketOptions& options,
76                              int flags) = 0;
77 
78   virtual bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
79                               const rtc::PacketOptions& options,
80                               int flags) = 0;
81 
82   // This method updates the RTP header extension map so that the RTP transport
83   // can parse the received packets and identify the MID. This is called by the
84   // BaseChannel when setting the content description.
85   //
86   // TODO(zhihuang): Merging and replacing following methods handling header
87   // extensions with SetParameters:
88   //   UpdateRtpHeaderExtensionMap,
89   //   UpdateSendEncryptedHeaderExtensionIds,
90   //   UpdateRecvEncryptedHeaderExtensionIds,
91   //   CacheRtpAbsSendTimeHeaderExtension,
92   virtual void UpdateRtpHeaderExtensionMap(
93       const cricket::RtpHeaderExtensions& header_extensions) = 0;
94 
95   virtual bool IsSrtpActive() const = 0;
96 
97   virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
98                                       RtpPacketSinkInterface* sink) = 0;
99 
100   virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
101 };
102 
103 }  // namespace webrtc
104 
105 #endif  // PC_RTP_TRANSPORT_INTERNAL_H_
106