1 /* 2 * Copyright (c) 2016 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 MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_ 12 #define MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_ 13 14 // TODO(deadbeef): Move SCTP code out of media/, and make it not depend on 15 // anything in media/. 16 17 #include <memory> 18 #include <string> 19 #include <vector> 20 21 #include "api/transport/data_channel_transport_interface.h" 22 // For SendDataParams/ReceiveDataParams. 23 // TODO(deadbeef): Use something else for SCTP. It's confusing that we use an 24 // SSRC field for SID. 25 #include "media/base/media_channel.h" 26 #include "p2p/base/packet_transport_internal.h" 27 #include "rtc_base/copy_on_write_buffer.h" 28 #include "rtc_base/thread.h" 29 30 namespace cricket { 31 32 // Constants that are important to API users 33 // The size of the SCTP association send buffer. 256kB, the usrsctp default. 34 constexpr int kSctpSendBufferSize = 256 * 1024; 35 36 // The number of outgoing streams that we'll negotiate. Since stream IDs (SIDs) 37 // are 0-based, the highest usable SID is 1023. 38 // 39 // It's recommended to use the maximum of 65535 in: 40 // https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2 41 // However, we use 1024 in order to save memory. usrsctp allocates 104 bytes 42 // for each pair of incoming/outgoing streams (on a 64-bit system), so 65535 43 // streams would waste ~6MB. 44 // 45 // Note: "max" and "min" here are inclusive. 46 constexpr uint16_t kMaxSctpStreams = 1024; 47 constexpr uint16_t kMaxSctpSid = kMaxSctpStreams - 1; 48 constexpr uint16_t kMinSctpSid = 0; 49 50 // This is the default SCTP port to use. It is passed along the wire and the 51 // connectee and connector must be using the same port. It is not related to the 52 // ports at the IP level. (Corresponds to: sockaddr_conn.sconn_port in 53 // usrsctp.h) 54 const int kSctpDefaultPort = 5000; 55 56 // Error cause codes defined at 57 // https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24 58 enum class SctpErrorCauseCode : uint16_t { 59 kInvalidStreamIdentifier = 1, 60 kMissingMandatoryParameter = 2, 61 kStaleCookieError = 3, 62 kOutOfResource = 4, 63 kUnresolvableAddress = 5, 64 kUnrecognizedChunkType = 6, 65 kInvalidMandatoryParameter = 7, 66 kUnrecognizedParameters = 8, 67 kNoUserData = 9, 68 kCookieReceivedWhileShuttingDown = 10, 69 kRestartWithNewAddresses = 11, 70 kUserInitiatedAbort = 12, 71 kProtocolViolation = 13, 72 }; 73 74 // Abstract SctpTransport interface for use internally (by PeerConnection etc.). 75 // Exists to allow mock/fake SctpTransports to be created. 76 class SctpTransportInternal { 77 public: ~SctpTransportInternal()78 virtual ~SctpTransportInternal() {} 79 80 virtual void SetOnConnectedCallback(std::function<void()> callback) = 0; 81 virtual void SetDataChannelSink(webrtc::DataChannelSink* sink) = 0; 82 83 // Changes what underlying DTLS transport is uses. Used when switching which 84 // bundled transport the SctpTransport uses. 85 virtual void SetDtlsTransport(rtc::PacketTransportInternal* transport) = 0; 86 87 // When Start is called, connects as soon as possible; this can be called 88 // before DTLS completes, in which case the connection will begin when DTLS 89 // completes. This method can be called multiple times, though not if either 90 // of the ports are changed. 91 // 92 // `local_sctp_port` and `remote_sctp_port` are passed along the wire and the 93 // listener and connector must be using the same port. They are not related 94 // to the ports at the IP level. If set to -1, we default to 95 // kSctpDefaultPort. 96 // `max_message_size_` sets the max message size on the connection. 97 // It must be smaller than or equal to kSctpSendBufferSize. 98 // It can be changed by a secons Start() call. 99 // 100 // TODO(deadbeef): Support calling Start with different local/remote ports 101 // and create a new association? Not clear if this is something we need to 102 // support though. See: https://github.com/w3c/webrtc-pc/issues/979 103 virtual bool Start(int local_sctp_port, 104 int remote_sctp_port, 105 int max_message_size) = 0; 106 107 // NOTE: Initially there was a "Stop" method here, but it was never used, so 108 // it was removed. 109 110 // Informs SctpTransport that `sid` will start being used. Returns false if 111 // it is impossible to use `sid`, or if it's already in use. 112 // Until calling this, can't send data using `sid`. 113 // TODO(deadbeef): Actually implement the "returns false if `sid` can't be 114 // used" part. See: 115 // https://bugs.chromium.org/p/chromium/issues/detail?id=619849 116 virtual bool OpenStream(int sid) = 0; 117 // The inverse of OpenStream. Begins the closing procedure, which will 118 // eventually result in SignalClosingProcedureComplete on the side that 119 // initiates it, and both SignalClosingProcedureStartedRemotely and 120 // SignalClosingProcedureComplete on the other side. 121 virtual bool ResetStream(int sid) = 0; 122 // Send data down this channel (will be wrapped as SCTP packets then given to 123 // usrsctp that will then post the network interface). 124 // Returns true iff successful data somewhere on the send-queue/network. 125 // Uses `params.ssrc` as the SCTP sid. 126 virtual bool SendData(int sid, 127 const webrtc::SendDataParams& params, 128 const rtc::CopyOnWriteBuffer& payload, 129 SendDataResult* result = nullptr) = 0; 130 131 // Indicates when the SCTP socket is created and not blocked by congestion 132 // control. This changes to false when SDR_BLOCK is returned from SendData, 133 // and 134 // changes to true when SignalReadyToSendData is fired. The underlying DTLS/ 135 // ICE channels may be unwritable while ReadyToSendData is true, because data 136 // can still be queued in usrsctp. 137 virtual bool ReadyToSendData() = 0; 138 // Returns the current max message size, set with Start(). 139 virtual int max_message_size() const = 0; 140 // Returns the current negotiated max # of outbound streams. 141 // Will return absl::nullopt if negotiation is incomplete. 142 virtual absl::optional<int> max_outbound_streams() const = 0; 143 // Returns the current negotiated max # of inbound streams. 144 virtual absl::optional<int> max_inbound_streams() const = 0; 145 146 // Helper for debugging. 147 virtual void set_debug_name_for_testing(const char* debug_name) = 0; 148 }; 149 150 } // namespace cricket 151 152 #endif // MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_ 153