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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 WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_ | |
12 #define WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_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> // for unique_ptr | |
18 #include <string> | |
19 #include <vector> | |
20 | |
21 #include "webrtc/base/copyonwritebuffer.h" | |
22 #include "webrtc/base/thread.h" | |
23 // For SendDataParams/ReceiveDataParams. | |
24 // TODO(deadbeef): Use something else for SCTP. It's confusing that we use an | |
25 // SSRC field for SID. | |
26 #include "webrtc/media/base/mediachannel.h" | |
27 #include "webrtc/p2p/base/transportchannel.h" | |
28 | |
29 namespace cricket { | |
30 | |
31 // The number of outgoing streams that we'll negotiate. Since stream IDs (SIDs) | |
32 // are 0-based, the highest usable SID is 1023. | |
33 // | |
34 // It's recommended to use the maximum of 65535 in: | |
35 // https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13#section-6.2 | |
36 // However, we use 1024 in order to save memory. usrsctp allocates 104 bytes | |
37 // for each pair of incoming/outgoing streams (on a 64-bit system), so 65535 | |
38 // streams would waste ~6MB. | |
39 // | |
40 // Note: "max" and "min" here are inclusive. | |
41 constexpr uint16_t kMaxSctpStreams = 1024; | |
42 constexpr uint16_t kMaxSctpSid = kMaxSctpStreams - 1; | |
43 constexpr uint16_t kMinSctpSid = 0; | |
44 | |
45 // This is the default SCTP port to use. It is passed along the wire and the | |
46 // connectee and connector must be using the same port. It is not related to the | |
47 // ports at the IP level. (Corresponds to: sockaddr_conn.sconn_port in | |
48 // usrsctp.h) | |
49 const int kSctpDefaultPort = 5000; | |
50 | |
51 // Abstract SctpTransport interface for use internally (by | |
52 // PeerConnection/WebRtcSession/etc.). Exists to allow mock/fake SctpTransports | |
53 // to be created. | |
54 class SctpTransportInternal { | |
55 public: | |
56 virtual ~SctpTransportInternal() {} | |
57 | |
58 // Changes what underlying DTLS channel is uses. Used when switching which | |
59 // bundled transport the SctpTransport uses. | |
60 // Assumes |channel| is non-null. | |
61 virtual void SetTransportChannel(TransportChannel* channel) = 0; | |
62 | |
63 // When Start is called, connects as soon as possible; this can be called | |
64 // before DTLS completes, in which case the connection will begin when DTLS | |
65 // completes. This method can be called multiple times, though not if either | |
66 // of the ports are changed. | |
67 // | |
68 // |local_sctp_port| and |remote_sctp_port| are passed along the wire and the | |
69 // listener and connector must be using the same port. They are not related | |
70 // to the ports at the IP level. If set to -1, we default to | |
71 // kSctpDefaultPort. | |
72 // | |
73 // TODO(deadbeef): Add remote max message size as parameter to Start, once we | |
74 // start supporting it. | |
75 // TODO(deadbeef): Support calling Start with different local/remote ports | |
76 // and create a new association? Not clear if this is something we need to | |
77 // support though. See: https://github.com/w3c/webrtc-pc/issues/979 | |
78 virtual bool Start(int local_sctp_port, int remote_sctp_port) = 0; | |
79 | |
80 // NOTE: Initially there was a "Stop" method here, but it was never used, so | |
81 // it was removed. | |
82 | |
83 // Informs SctpTransport that |sid| will start being used. Returns false if | |
84 // it is impossible to use |sid|, or if it's already in use. | |
85 // Until calling this, can't send data using |sid|. | |
86 // TODO(deadbeef): Actually implement the "returns false if |sid| can't be | |
87 // used" part. See: | |
88 // https://bugs.chromium.org/p/chromium/issues/detail?id=619849 | |
89 virtual bool OpenStream(int sid) = 0; | |
90 // The inverse of OpenStream. When this method returns, the reset process may | |
91 // have not finished but it will have begun. | |
92 // TODO(deadbeef): We need a way to tell when it's done. See: | |
93 // https://bugs.chromium.org/p/webrtc/issues/detail?id=4453 | |
94 virtual bool ResetStream(int sid) = 0; | |
95 // Send data down this channel (will be wrapped as SCTP packets then given to | |
96 // usrsctp that will then post the network interface). | |
97 // Returns true iff successful data somewhere on the send-queue/network. | |
98 // Uses |params.ssrc| as the SCTP sid. | |
99 virtual bool SendData(const SendDataParams& params, | |
100 const rtc::CopyOnWriteBuffer& payload, | |
101 SendDataResult* result = nullptr) = 0; | |
102 | |
103 // Indicates when the SCTP socket is created and not blocked by congestion | |
104 // control. This changes to false when SDR_BLOCK is returned from SendData, | |
105 // and | |
106 // changes to true when SignalReadyToSendData is fired. The underlying DTLS/ | |
107 // ICE channels may be unwritable while ReadyToSendData is true, because data | |
108 // can still be queued in usrsctp. | |
109 virtual bool ReadyToSendData() = 0; | |
110 | |
111 sigslot::signal0<> SignalReadyToSendData; | |
112 // ReceiveDataParams includes SID, seq num, timestamp, etc. CopyOnWriteBuffer | |
113 // contains message payload. | |
114 sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&> | |
115 SignalDataReceived; | |
116 // Parameter is SID of closed stream. | |
117 sigslot::signal1<int> SignalStreamClosedRemotely; | |
118 | |
119 // Helper for debugging. | |
120 virtual void set_debug_name_for_testing(const char* debug_name) = 0; | |
121 }; | |
122 | |
123 // Factory class which can be used to allow fake SctpTransports to be injected | |
124 // for testing. Or, theoretically, SctpTransportInternal implementations that | |
125 // use something other than usrsctp. | |
126 class SctpTransportInternalFactory { | |
127 public: | |
128 virtual ~SctpTransportInternalFactory() {} | |
129 | |
130 // Create an SCTP transport using |channel| for the underlying transport. | |
131 virtual std::unique_ptr<SctpTransportInternal> CreateSctpTransport( | |
132 TransportChannel* channel) = 0; | |
133 }; | |
134 | |
135 } // namespace cricket | |
136 | |
137 #endif // WEBRTC_MEDIA_SCTP_SCTPTRANSPORTINTERNAL_H_ | |
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