| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_ |
| #define MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_ |
| |
| // TODO(deadbeef): Move SCTP code out of media/, and make it not depend on |
| // anything in media/. |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/transport/data_channel_transport_interface.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/thread.h" |
| // For SendDataParams/ReceiveDataParams. |
| // TODO(deadbeef): Use something else for SCTP. It's confusing that we use an |
| // SSRC field for SID. |
| #include "media/base/media_channel.h" |
| #include "p2p/base/packet_transport_internal.h" |
| |
| namespace cricket { |
| |
| // Constants that are important to API users |
| // The size of the SCTP association send buffer. 256kB, the usrsctp default. |
| constexpr int kSctpSendBufferSize = 256 * 1024; |
| |
| // The number of outgoing streams that we'll negotiate. Since stream IDs (SIDs) |
| // are 0-based, the highest usable SID is 1023. |
| // |
| // It's recommended to use the maximum of 65535 in: |
| // https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2 |
| // However, we use 1024 in order to save memory. usrsctp allocates 104 bytes |
| // for each pair of incoming/outgoing streams (on a 64-bit system), so 65535 |
| // streams would waste ~6MB. |
| // |
| // Note: "max" and "min" here are inclusive. |
| constexpr uint16_t kMaxSctpStreams = 1024; |
| constexpr uint16_t kMaxSctpSid = kMaxSctpStreams - 1; |
| constexpr uint16_t kMinSctpSid = 0; |
| |
| // This is the default SCTP port to use. It is passed along the wire and the |
| // connectee and connector must be using the same port. It is not related to the |
| // ports at the IP level. (Corresponds to: sockaddr_conn.sconn_port in |
| // usrsctp.h) |
| const int kSctpDefaultPort = 5000; |
| |
| // Error cause codes defined at |
| // https://www.iana.org/assignments/sctp-parameters/sctp-parameters.xhtml#sctp-parameters-24 |
| enum class SctpErrorCauseCode : uint16_t { |
| kInvalidStreamIdentifier = 1, |
| kMissingMandatoryParameter = 2, |
| kStaleCookieError = 3, |
| kOutOfResource = 4, |
| kUnresolvableAddress = 5, |
| kUnrecognizedChunkType = 6, |
| kInvalidMandatoryParameter = 7, |
| kUnrecognizedParameters = 8, |
| kNoUserData = 9, |
| kCookieReceivedWhileShuttingDown = 10, |
| kRestartWithNewAddresses = 11, |
| kUserInitiatedAbort = 12, |
| kProtocolViolation = 13, |
| }; |
| |
| // Abstract SctpTransport interface for use internally (by PeerConnection etc.). |
| // Exists to allow mock/fake SctpTransports to be created. |
| class SctpTransportInternal { |
| public: |
| virtual ~SctpTransportInternal() {} |
| |
| // Changes what underlying DTLS transport is uses. Used when switching which |
| // bundled transport the SctpTransport uses. |
| virtual void SetDtlsTransport(rtc::PacketTransportInternal* transport) = 0; |
| |
| // When Start is called, connects as soon as possible; this can be called |
| // before DTLS completes, in which case the connection will begin when DTLS |
| // completes. This method can be called multiple times, though not if either |
| // of the ports are changed. |
| // |
| // |local_sctp_port| and |remote_sctp_port| are passed along the wire and the |
| // listener and connector must be using the same port. They are not related |
| // to the ports at the IP level. If set to -1, we default to |
| // kSctpDefaultPort. |
| // |max_message_size_| sets the max message size on the connection. |
| // It must be smaller than or equal to kSctpSendBufferSize. |
| // It can be changed by a secons Start() call. |
| // |
| // TODO(deadbeef): Support calling Start with different local/remote ports |
| // and create a new association? Not clear if this is something we need to |
| // support though. See: https://github.com/w3c/webrtc-pc/issues/979 |
| virtual bool Start(int local_sctp_port, |
| int remote_sctp_port, |
| int max_message_size) = 0; |
| |
| // NOTE: Initially there was a "Stop" method here, but it was never used, so |
| // it was removed. |
| |
| // Informs SctpTransport that |sid| will start being used. Returns false if |
| // it is impossible to use |sid|, or if it's already in use. |
| // Until calling this, can't send data using |sid|. |
| // TODO(deadbeef): Actually implement the "returns false if |sid| can't be |
| // used" part. See: |
| // https://bugs.chromium.org/p/chromium/issues/detail?id=619849 |
| virtual bool OpenStream(int sid) = 0; |
| // The inverse of OpenStream. Begins the closing procedure, which will |
| // eventually result in SignalClosingProcedureComplete on the side that |
| // initiates it, and both SignalClosingProcedureStartedRemotely and |
| // SignalClosingProcedureComplete on the other side. |
| virtual bool ResetStream(int sid) = 0; |
| // Send data down this channel (will be wrapped as SCTP packets then given to |
| // usrsctp that will then post the network interface). |
| // Returns true iff successful data somewhere on the send-queue/network. |
| // Uses |params.ssrc| as the SCTP sid. |
| virtual bool SendData(int sid, |
| const webrtc::SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| SendDataResult* result = nullptr) = 0; |
| |
| // Indicates when the SCTP socket is created and not blocked by congestion |
| // control. This changes to false when SDR_BLOCK is returned from SendData, |
| // and |
| // changes to true when SignalReadyToSendData is fired. The underlying DTLS/ |
| // ICE channels may be unwritable while ReadyToSendData is true, because data |
| // can still be queued in usrsctp. |
| virtual bool ReadyToSendData() = 0; |
| // Returns the current max message size, set with Start(). |
| virtual int max_message_size() const = 0; |
| // Returns the current negotiated max # of outbound streams. |
| // Will return absl::nullopt if negotiation is incomplete. |
| virtual absl::optional<int> max_outbound_streams() const = 0; |
| // Returns the current negotiated max # of inbound streams. |
| virtual absl::optional<int> max_inbound_streams() const = 0; |
| |
| sigslot::signal0<> SignalReadyToSendData; |
| sigslot::signal0<> SignalAssociationChangeCommunicationUp; |
| // ReceiveDataParams includes SID, seq num, timestamp, etc. CopyOnWriteBuffer |
| // contains message payload. |
| sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&> |
| SignalDataReceived; |
| // Parameter is SID; fired when we receive an incoming stream reset on an |
| // open stream, indicating that the other side started the closing procedure. |
| // After resetting the outgoing stream, SignalClosingProcedureComplete will |
| // fire too. |
| sigslot::signal1<int> SignalClosingProcedureStartedRemotely; |
| // Parameter is SID; fired when closing procedure is complete (both incoming |
| // and outgoing streams reset). |
| sigslot::signal1<int> SignalClosingProcedureComplete; |
| // Fired when the underlying DTLS transport has closed due to an error |
| // or an incoming DTLS disconnect or SCTP transport errors. |
| sigslot::signal1<webrtc::RTCError> SignalClosedAbruptly; |
| |
| // Helper for debugging. |
| virtual void set_debug_name_for_testing(const char* debug_name) = 0; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // MEDIA_SCTP_SCTP_TRANSPORT_INTERNAL_H_ |