Bjorn A Mellem | 364b267 | 2019-08-20 16:58:03 -0700 | [diff] [blame] | 1 | /* |
| 2 | * Copyright 2019 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_DATAGRAM_RTP_TRANSPORT_H_ |
| 12 | #define PC_DATAGRAM_RTP_TRANSPORT_H_ |
| 13 | |
| 14 | #include <map> |
| 15 | #include <memory> |
| 16 | #include <string> |
| 17 | #include <vector> |
| 18 | |
| 19 | #include "api/crypto/crypto_options.h" |
Niels Möller | 65f17ca | 2019-09-12 13:59:36 +0200 | [diff] [blame] | 20 | #include "api/transport/datagram_transport_interface.h" |
Bjorn A Mellem | 7a9a092 | 2019-11-26 09:19:40 -0800 | [diff] [blame] | 21 | #include "api/transport/media/media_transport_interface.h" |
Bjorn A Mellem | 364b267 | 2019-08-20 16:58:03 -0700 | [diff] [blame] | 22 | #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| 23 | #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| 24 | #include "p2p/base/ice_transport_internal.h" |
| 25 | #include "p2p/base/packet_transport_internal.h" |
| 26 | #include "pc/rtp_transport_internal.h" |
| 27 | #include "rtc_base/buffer.h" |
| 28 | #include "rtc_base/buffer_queue.h" |
| 29 | #include "rtc_base/constructor_magic.h" |
| 30 | #include "rtc_base/ssl_stream_adapter.h" |
| 31 | #include "rtc_base/stream.h" |
| 32 | #include "rtc_base/strings/string_builder.h" |
| 33 | #include "rtc_base/thread_checker.h" |
| 34 | |
| 35 | namespace webrtc { |
| 36 | |
| 37 | constexpr int kDatagramDtlsAdaptorComponent = -1; |
| 38 | |
| 39 | // RTP transport which uses the DatagramTransportInterface to send and receive |
| 40 | // packets. |
| 41 | class DatagramRtpTransport : public RtpTransportInternal, |
| 42 | public webrtc::DatagramSinkInterface, |
| 43 | public webrtc::MediaTransportStateCallback { |
| 44 | public: |
| 45 | DatagramRtpTransport( |
| 46 | const std::vector<webrtc::RtpExtension>& rtp_header_extensions, |
| 47 | cricket::IceTransportInternal* ice_transport, |
| 48 | DatagramTransportInterface* datagram_transport); |
| 49 | |
| 50 | ~DatagramRtpTransport() override; |
| 51 | |
| 52 | // ===================================================== |
| 53 | // Overrides for webrtc::DatagramTransportSinkInterface |
| 54 | // and MediaTransportStateCallback |
| 55 | // ===================================================== |
| 56 | void OnDatagramReceived(rtc::ArrayView<const uint8_t> data) override; |
| 57 | |
| 58 | void OnDatagramSent(webrtc::DatagramId datagram_id) override; |
| 59 | |
| 60 | void OnDatagramAcked(const webrtc::DatagramAck& ack) override; |
| 61 | |
| 62 | void OnDatagramLost(webrtc::DatagramId datagram_id) override; |
| 63 | |
| 64 | void OnStateChanged(webrtc::MediaTransportState state) override; |
| 65 | |
| 66 | // ===================================================== |
| 67 | // RtpTransportInternal overrides |
| 68 | // ===================================================== |
| 69 | bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet, |
| 70 | const rtc::PacketOptions& options, |
| 71 | int flags) override; |
| 72 | |
| 73 | bool SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, |
| 74 | const rtc::PacketOptions& options, |
| 75 | int flags) override; |
| 76 | |
| 77 | const std::string& transport_name() const override; |
| 78 | |
| 79 | // Datagram transport always muxes RTCP. |
| 80 | bool rtcp_mux_enabled() const override { return true; } |
| 81 | void SetRtcpMuxEnabled(bool enable) override {} |
| 82 | |
| 83 | int SetRtpOption(rtc::Socket::Option opt, int value) override; |
| 84 | int SetRtcpOption(rtc::Socket::Option opt, int value) override; |
| 85 | |
| 86 | bool IsReadyToSend() const override; |
| 87 | |
| 88 | bool IsWritable(bool rtcp) const override; |
| 89 | |
| 90 | bool IsSrtpActive() const override { return false; } |
| 91 | |
| 92 | void UpdateRtpHeaderExtensionMap( |
| 93 | const cricket::RtpHeaderExtensions& header_extensions) override; |
| 94 | |
| 95 | bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria, |
| 96 | RtpPacketSinkInterface* sink) override; |
| 97 | |
| 98 | bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override; |
| 99 | |
| 100 | private: |
| 101 | // RTP/RTCP packet info stored for each sent packet. |
| 102 | struct SentPacketInfo { |
| 103 | // RTP packet info with ssrc and transport sequence number. |
| 104 | SentPacketInfo(int64_t packet_id, |
| 105 | uint32_t ssrc, |
| 106 | uint16_t transport_sequence_number) |
| 107 | : ssrc(ssrc), |
| 108 | transport_sequence_number(transport_sequence_number), |
| 109 | packet_id(packet_id) {} |
| 110 | |
| 111 | // Packet info without SSRC and transport sequence number used for RTCP |
| 112 | // packets, RTP packets when transport sequence number is not provided or |
| 113 | // when feedback translation is disabled. |
| 114 | explicit SentPacketInfo(int64_t packet_id) : packet_id(packet_id) {} |
| 115 | |
| 116 | SentPacketInfo() = default; |
| 117 | |
| 118 | absl::optional<uint32_t> ssrc; |
| 119 | |
| 120 | // Transport sequence number (if it was provided in outgoing RTP packet). |
| 121 | // It is used to re-create RTCP feedback packets from datagram ACKs. |
| 122 | absl::optional<uint16_t> transport_sequence_number; |
| 123 | |
| 124 | // Packet id from rtc::PacketOptions. It is required to propagage sent |
| 125 | // notification up the stack (SignalSentPacket). |
| 126 | int64_t packet_id = 0; |
| 127 | }; |
| 128 | |
| 129 | // Finds SentPacketInfo for given |datagram_id| and removes map entry. |
| 130 | // Returns false if entry was not found. |
| 131 | bool GetAndRemoveSentPacketInfo(webrtc::DatagramId datagram_id, |
| 132 | SentPacketInfo* sent_packet_info); |
| 133 | |
| 134 | // Sends datagram to datagram_transport. |
| 135 | bool SendDatagram(rtc::ArrayView<const uint8_t> data, |
| 136 | webrtc::DatagramId datagram_id); |
| 137 | |
| 138 | // Propagates network route changes from ICE. |
| 139 | void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route); |
| 140 | |
| 141 | rtc::ThreadChecker thread_checker_; |
| 142 | cricket::IceTransportInternal* ice_transport_; |
| 143 | webrtc::DatagramTransportInterface* datagram_transport_; |
| 144 | |
| 145 | RtpDemuxer rtp_demuxer_; |
| 146 | |
| 147 | MediaTransportState state_ = MediaTransportState::kPending; |
| 148 | |
| 149 | // Extension map for parsing transport sequence numbers. |
| 150 | webrtc::RtpHeaderExtensionMap rtp_header_extension_map_; |
| 151 | |
| 152 | // Keeps information about sent RTP packet until they are Acked or Lost. |
| 153 | std::map<webrtc::DatagramId, SentPacketInfo> sent_rtp_packet_map_; |
| 154 | |
| 155 | // Current datagram_id, incremented after each sent RTP packets. |
| 156 | // Datagram id is passed to datagram transport when we send datagram and we |
| 157 | // get it back in notifications about Sent, Acked and Lost datagrams. |
| 158 | int64_t current_datagram_id_ = 0; |
| 159 | |
| 160 | // TODO(sukhanov): Previous nonzero timestamp is required for workaround for |
| 161 | // zero timestamps received, which sometimes are received from datagram |
| 162 | // transport. Investigate if we can eliminate zero timestamps. |
| 163 | int64_t previous_nonzero_timestamp_us_ = 0; |
| 164 | |
| 165 | // Disable datagram to RTCP feedback translation and enable RTCP feedback |
| 166 | // loop (note that having both RTCP and datagram feedback loops is |
| 167 | // inefficient, but can be useful in tests and experiments). |
| 168 | const bool disable_datagram_to_rtcp_feeback_translation_; |
| 169 | }; |
| 170 | |
| 171 | } // namespace webrtc |
| 172 | |
| 173 | #endif // PC_DATAGRAM_RTP_TRANSPORT_H_ |