Reland of: Separating SCTP code from BaseChannel/MediaChannel.

webrtc/media/sctp/sctptransportinternal.h

+/*
+ *  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 WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_
+#define WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_
+
+#include <memory>  // for unique_ptr
+#include <string>
+#include <vector>
+
+#include "webrtc/base/copyonwritebuffer.h"
+#include "webrtc/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 "webrtc/media/base/mediachannel.h"
+#include "webrtc/p2p/base/transportchannel.h"
+
+namespace cricket {
+
+// 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;
+
+// Abstract SctpTransport interface for use internally (by
+// PeerConnection/WebRtcSession/etc.). Exists to allow mock/fake SctpTransports
+// to be created.
+class SctpTransportInternal {
+ public:
+  virtual ~SctpTransportInternal() {}
+
+  // Changes what underlying DTLS channel is uses. Used when switching which
+  // bundled transport the SctpTransport uses.
+  // Assumes |channel| is non-null.
+  virtual void SetTransportChannel(TransportChannel* channel) = 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.
+  //
+  // TODO(deadbeef): Add remote max message size as parameter to Start, once we
+  // start supporting it.
+  // 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) = 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. When this method returns, the reset process may
+  // have not finished but it will have begun.
+  // TODO(deadbeef): We need a way to tell when it's done. See:
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
+  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(const 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;
+
+  sigslot::signal0<> SignalReadyToSendData;
+  // ReceiveDataParams includes SID, seq num, timestamp, etc. CopyOnWriteBuffer
+  // contains message payload.
+  sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&>
+      SignalDataReceived;
+  // Parameter is SID of closed stream.
+  sigslot::signal1<int> SignalStreamClosedRemotely;
+
+  // Helper for debugging.
+  virtual void set_debug_name_for_testing(const char* debug_name) = 0;
+};
+
+// Factory class which can be used to allow fake SctpTransports to be injected
+// for testing. Or, theoretically, SctpTransportInternal implementations that
+// use something other than usrsctp.
+class SctpTransportInternalFactory {
+ public:
+  virtual ~SctpTransportInternalFactory() {}
+
+  // Create an SCTP transport using |channel| for the underlying transport.
+  virtual std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
+      TransportChannel* channel) = 0;
+};
+
+}  // namespace cricket
+
+#endif  // WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_