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

[pthatcher1, 2016/12/23 01:39:31]

I don't think we'll ever need that.

[Taylor Brandstetter, 2016/12/23 06:29:05]

Can you comment on the github issue?

[pthatcher1, 2017/01/03 18:31:26]

I commented on github.  It seems like Bernard agrees that we won't need this.

+  virtual bool Start(int local_sctp_port, int remote_sctp_port) = 0;

[pthatcher1, 2016/12/23 01:39:31]

We could model this as the ORTC object:

- A port is passed into the constructor (with default of 5000)
- The thing passed into Start is the remote capabilities (max message size)

Do we even need the remote port?  I don't see it in the ORTC API.

[Taylor Brandstetter, 2016/12/23 06:29:05]

There's an issue filed for ORTC for that. There was discussion about it this
week, coincidentally; I'd recommend joining the mailing list if you haven't.

I agree that we may end up putting the local port in the constructor, but that
would require more changes in WebRtcSession than I wanted to do now. But we
should do that by the time we have an ORTC version of SctpTransport in our
public API, assuming that's how the ORTC API settles out.

[pthatcher1, 2017/01/03 18:31:26]

Sure, we can always refactor more later.  

I just responded to the mailing list.  JSEP (via draft-ietf-mmusic-sctp-sdp)
requires the ports to be the same, so could simplify this start method to only
taking one port value.

[pthatcher1, 2017/01/03 20:45:00]

Nevermind, I was wrong about that :).

+
+  // 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;

[pthatcher1, 2016/12/23 01:39:31]

It would be nice if we instead had a CreateDataChannel method here that returned
a data channel that's tied to a SID.  But that can come later, I guess. 

[Taylor Brandstetter, 2016/12/23 06:29:05]

I disagree; that would create a circular dependency between DataChannel and
SctpTransport that we don't really need.

[pthatcher1, 2017/01/03 18:31:26]

Sorry, I meant return a DataChannelInterface, which would avoid a circular
dependency.

+  // 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;

[pthatcher1, 2016/12/23 01:39:31]

Why not CloseStream (to go along with DataChannel.close() and the opposite of
Open()).

[Taylor Brandstetter, 2016/12/23 06:29:05]

Because streams aren't closed, they're reset; that's just how SCTP works. See:
https://tools.ietf.org/html/rfc6525

If anything, OpenStream should be removed; calling OpenStream doesn't actually
do anything.

+  // 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_