| /* |
| * Copyright (c) 2012 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_H_ |
| #define MEDIA_SCTP_SCTP_TRANSPORT_H_ |
| |
| #include <errno.h> |
| |
| #include <cstdint> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/transport/sctp_transport_factory_interface.h" |
| #include "rtc_base/buffer.h" |
| #include "rtc_base/constructor_magic.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/task_utils/pending_task_safety_flag.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "rtc_base/thread.h" |
| // For SendDataParams/ReceiveDataParams. |
| #include "media/base/media_channel.h" |
| #include "media/sctp/sctp_transport_internal.h" |
| |
| // Defined by "usrsctplib/usrsctp.h" |
| struct sockaddr_conn; |
| struct sctp_assoc_change; |
| struct sctp_rcvinfo; |
| struct sctp_stream_reset_event; |
| struct sctp_sendv_spa; |
| |
| // Defined by <sys/socket.h> |
| struct socket; |
| namespace cricket { |
| |
| // Holds data to be passed on to a transport. |
| struct SctpInboundPacket; |
| |
| // From transport calls, data flows like this: |
| // [network thread (although it can in princple be another thread)] |
| // 1. SctpTransport::SendData(data) |
| // 2. usrsctp_sendv(data) |
| // [network thread returns; sctp thread then calls the following] |
| // 3. OnSctpOutboundPacket(wrapped_data) |
| // [sctp thread returns having async invoked on the network thread] |
| // 4. SctpTransport::OnPacketFromSctpToNetwork(wrapped_data) |
| // 5. DtlsTransport::SendPacket(wrapped_data) |
| // 6. ... across network ... a packet is sent back ... |
| // 7. SctpTransport::OnPacketReceived(wrapped_data) |
| // 8. usrsctp_conninput(wrapped_data) |
| // [network thread returns; sctp thread then calls the following] |
| // 9. OnSctpInboundData(data) |
| // 10. SctpTransport::OnDataFromSctpToTransport(data) |
| // [sctp thread returns having async invoked on the network thread] |
| // 11. SctpTransport::OnDataFromSctpToTransport(data) |
| // 12. SctpTransport::SignalDataReceived(data) |
| // [from the same thread, methods registered/connected to |
| // SctpTransport are called with the recieved data] |
| class SctpTransport : public SctpTransportInternal, |
| public sigslot::has_slots<> { |
| public: |
| // |network_thread| is where packets will be processed and callbacks from |
| // this transport will be posted, and is the only thread on which public |
| // methods can be called. |
| // |transport| is not required (can be null). |
| SctpTransport(rtc::Thread* network_thread, |
| rtc::PacketTransportInternal* transport); |
| ~SctpTransport() override; |
| |
| // SctpTransportInternal overrides (see sctptransportinternal.h for comments). |
| void SetDtlsTransport(rtc::PacketTransportInternal* transport) override; |
| bool Start(int local_port, int remote_port, int max_message_size) override; |
| bool OpenStream(int sid) override; |
| bool ResetStream(int sid) override; |
| bool SendData(const SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| SendDataResult* result = nullptr) override; |
| bool ReadyToSendData() override; |
| int max_message_size() const override { return max_message_size_; } |
| absl::optional<int> max_outbound_streams() const override { |
| return max_outbound_streams_; |
| } |
| absl::optional<int> max_inbound_streams() const override { |
| return max_inbound_streams_; |
| } |
| void set_debug_name_for_testing(const char* debug_name) override { |
| debug_name_ = debug_name; |
| } |
| int InjectDataOrNotificationFromSctpForTesting(const void* data, |
| size_t length, |
| struct sctp_rcvinfo rcv, |
| int flags); |
| |
| // Exposed to allow Post call from c-callbacks. |
| // TODO(deadbeef): Remove this or at least make it return a const pointer. |
| rtc::Thread* network_thread() const { return network_thread_; } |
| |
| private: |
| // A message to be sent by the sctp library. This class is used to track the |
| // progress of writing a single message to the sctp library in the presence of |
| // partial writes. In this case, the Advance() function is provided in order |
| // to advance over what has already been accepted by the sctp library and |
| // avoid copying the remaining partial message buffer. |
| class OutgoingMessage { |
| public: |
| OutgoingMessage(const rtc::CopyOnWriteBuffer& buffer, |
| const SendDataParams& send_params) |
| : buffer_(buffer), send_params_(send_params) {} |
| |
| // Advances the buffer by the incremented amount. Must not advance further |
| // than the current data size. |
| void Advance(size_t increment) { |
| RTC_DCHECK_LE(increment + offset_, buffer_.size()); |
| offset_ += increment; |
| } |
| |
| size_t size() const { return buffer_.size() - offset_; } |
| |
| const void* data() const { return buffer_.data() + offset_; } |
| |
| SendDataParams send_params() const { return send_params_; } |
| |
| private: |
| const rtc::CopyOnWriteBuffer buffer_; |
| const SendDataParams send_params_; |
| size_t offset_ = 0; |
| }; |
| |
| void ConnectTransportSignals(); |
| void DisconnectTransportSignals(); |
| |
| // Creates the socket and connects. |
| bool Connect(); |
| |
| // Returns false when opening the socket failed. |
| bool OpenSctpSocket(); |
| // Helpet method to set socket options. |
| bool ConfigureSctpSocket(); |
| // Sets |sock_ |to nullptr. |
| void CloseSctpSocket(); |
| |
| // Sends a SCTP_RESET_STREAM for all streams in closing_ssids_. |
| bool SendQueuedStreamResets(); |
| |
| // Sets the "ready to send" flag and fires signal if needed. |
| void SetReadyToSendData(); |
| |
| // Sends the outgoing buffered message that was only partially accepted by the |
| // sctp lib because it did not have enough space. Returns true if the entire |
| // buffered message was accepted by the sctp lib. |
| bool SendBufferedMessage(); |
| |
| // Tries to send the |payload| on the usrsctp lib. The message will be |
| // advanced by the amount that was sent. |
| SendDataResult SendMessageInternal(OutgoingMessage* message); |
| |
| // Callbacks from DTLS transport. |
| void OnWritableState(rtc::PacketTransportInternal* transport); |
| virtual void OnPacketRead(rtc::PacketTransportInternal* transport, |
| const char* data, |
| size_t len, |
| const int64_t& packet_time_us, |
| int flags); |
| void OnClosed(rtc::PacketTransportInternal* transport); |
| |
| // Methods related to usrsctp callbacks. |
| void OnSendThresholdCallback(); |
| sockaddr_conn GetSctpSockAddr(int port); |
| |
| // Called using |invoker_| to send packet on the network. |
| void OnPacketFromSctpToNetwork(const rtc::CopyOnWriteBuffer& buffer); |
| |
| // Called on the SCTP thread. |
| // Flags are standard socket API flags (RFC 6458). |
| int OnDataOrNotificationFromSctp(const void* data, |
| size_t length, |
| struct sctp_rcvinfo rcv, |
| int flags); |
| // Called using |invoker_| to decide what to do with the data. |
| void OnDataFromSctpToTransport(const ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& buffer); |
| // Called using |invoker_| to decide what to do with the notification. |
| void OnNotificationFromSctp(const rtc::CopyOnWriteBuffer& buffer); |
| void OnNotificationAssocChange(const sctp_assoc_change& change); |
| |
| void OnStreamResetEvent(const struct sctp_stream_reset_event* evt); |
| |
| // Responsible for marshalling incoming data to the transports listeners, and |
| // outgoing data to the network interface. |
| rtc::Thread* network_thread_; |
| // Helps pass inbound/outbound packets asynchronously to the network thread. |
| webrtc::ScopedTaskSafety task_safety_; |
| // Underlying DTLS transport. |
| rtc::PacketTransportInternal* transport_ = nullptr; |
| |
| // Track the data received from usrsctp between callbacks until the EOR bit |
| // arrives. |
| rtc::CopyOnWriteBuffer partial_incoming_message_; |
| ReceiveDataParams partial_params_; |
| int partial_flags_; |
| // A message that was attempted to be sent, but was only partially accepted by |
| // usrsctp lib with usrsctp_sendv() because it cannot buffer the full message. |
| // This occurs because we explicitly set the EOR bit when sending, so |
| // usrsctp_sendv() is not atomic. |
| absl::optional<OutgoingMessage> partial_outgoing_message_; |
| |
| bool was_ever_writable_ = false; |
| int local_port_ = kSctpDefaultPort; |
| int remote_port_ = kSctpDefaultPort; |
| int max_message_size_ = kSctpSendBufferSize; |
| struct socket* sock_ = nullptr; // The socket created by usrsctp_socket(...). |
| |
| // Has Start been called? Don't create SCTP socket until it has. |
| bool started_ = false; |
| // Are we ready to queue data (SCTP socket created, and not blocked due to |
| // congestion control)? Different than |transport_|'s "ready to send". |
| bool ready_to_send_data_ = false; |
| |
| // Used to keep track of the status of each stream (or rather, each pair of |
| // incoming/outgoing streams with matching IDs). It's specifically used to |
| // keep track of the status of resets, but more information could be put here |
| // later. |
| // |
| // See datachannel.h for a summary of the closing procedure. |
| struct StreamStatus { |
| // Closure initiated by application via ResetStream? Note that |
| // this may be true while outgoing_reset_initiated is false if the outgoing |
| // reset needed to be queued. |
| bool closure_initiated = false; |
| // Whether we've initiated the outgoing stream reset via |
| // SCTP_RESET_STREAMS. |
| bool outgoing_reset_initiated = false; |
| // Whether usrsctp has indicated that the incoming/outgoing streams have |
| // been reset. It's expected that the peer will reset its outgoing stream |
| // (our incoming stream) after receiving the reset for our outgoing stream, |
| // though older versions of chromium won't do this. See crbug.com/559394 |
| // for context. |
| bool outgoing_reset_complete = false; |
| bool incoming_reset_complete = false; |
| |
| // Some helper methods to improve code readability. |
| bool is_open() const { |
| return !closure_initiated && !incoming_reset_complete && |
| !outgoing_reset_complete; |
| } |
| // We need to send an outgoing reset if the application has closed the data |
| // channel, or if we received a reset of the incoming stream from the |
| // remote endpoint, indicating the data channel was closed remotely. |
| bool need_outgoing_reset() const { |
| return (incoming_reset_complete || closure_initiated) && |
| !outgoing_reset_initiated; |
| } |
| bool reset_complete() const { |
| return outgoing_reset_complete && incoming_reset_complete; |
| } |
| }; |
| |
| // Entries should only be removed from this map if |reset_complete| is |
| // true. |
| std::map<uint32_t, StreamStatus> stream_status_by_sid_; |
| |
| // A static human-readable name for debugging messages. |
| const char* debug_name_ = "SctpTransport"; |
| // Hides usrsctp interactions from this header file. |
| class UsrSctpWrapper; |
| // Number of channels negotiated. Not set before negotiation completes. |
| absl::optional<int> max_outbound_streams_; |
| absl::optional<int> max_inbound_streams_; |
| |
| // Used for associating this transport with the underlying sctp socket in |
| // various callbacks. |
| uintptr_t id_ = 0; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(SctpTransport); |
| }; |
| |
| class SctpTransportFactory : public webrtc::SctpTransportFactoryInterface { |
| public: |
| explicit SctpTransportFactory(rtc::Thread* network_thread) |
| : network_thread_(network_thread) {} |
| |
| std::unique_ptr<SctpTransportInternal> CreateSctpTransport( |
| rtc::PacketTransportInternal* transport) override { |
| return std::unique_ptr<SctpTransportInternal>( |
| new SctpTransport(network_thread_, transport)); |
| } |
| |
| private: |
| rtc::Thread* network_thread_; |
| }; |
| |
| } // namespace cricket |
| |
| #endif // MEDIA_SCTP_SCTP_TRANSPORT_H_ |