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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle SCTP
3 * Copyright 2012 Google Inc, and Robin Seggelmann
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_MEDIA_SCTP_SCTPDATAENGINE_H_
29#define TALK_MEDIA_SCTP_SCTPDATAENGINE_H_
30
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000031#include <errno.h>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000032#include <string>
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +000033#include <vector>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000034
henrike@webrtc.org28654cb2013-07-22 21:07:49 +000035namespace cricket {
36// Some ERRNO values get re-#defined to WSA* equivalents in some talk/
37// headers. We save the original ones in an enum.
38enum PreservedErrno {
39 SCTP_EINPROGRESS = EINPROGRESS,
40 SCTP_EWOULDBLOCK = EWOULDBLOCK
41};
42} // namespace cricket
43
henrike@webrtc.org28e20752013-07-10 00:45:36 +000044#include "talk/base/buffer.h"
45#include "talk/base/scoped_ptr.h"
46#include "talk/media/base/codec.h"
47#include "talk/media/base/mediachannel.h"
48#include "talk/media/base/mediaengine.h"
49
50// Defined by "usrsctplib/usrsctp.h"
51struct sockaddr_conn;
52struct sctp_assoc_change;
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +000053struct sctp_stream_reset_event;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000054// Defined by <sys/socket.h>
55struct socket;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000056namespace cricket {
mallinath@webrtc.org1112c302013-09-23 20:34:45 +000057// The highest stream ID (Sid) that SCTP allows, and the number of streams we
58// tell SCTP we're going to use.
wu@webrtc.org97077a32013-10-25 21:18:33 +000059const uint32 kMaxSctpSid = 1023;
mallinath@webrtc.org1112c302013-09-23 20:34:45 +000060
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061// A DataEngine that interacts with usrsctp.
62//
63// From channel calls, data flows like this:
64// [worker thread (although it can in princple be another thread)]
65// 1. SctpDataMediaChannel::SendData(data)
66// 2. usrsctp_sendv(data)
67// [worker thread returns; sctp thread then calls the following]
68// 3. OnSctpOutboundPacket(wrapped_data)
69// [sctp thread returns having posted a message for the worker thread]
70// 4. SctpDataMediaChannel::OnMessage(wrapped_data)
71// 5. SctpDataMediaChannel::OnPacketFromSctpToNetwork(wrapped_data)
72// 6. NetworkInterface::SendPacket(wrapped_data)
73// 7. ... across network ... a packet is sent back ...
74// 8. SctpDataMediaChannel::OnPacketReceived(wrapped_data)
75// 9. usrsctp_conninput(wrapped_data)
76// [worker thread returns; sctp thread then calls the following]
77// 10. OnSctpInboundData(data)
78// [sctp thread returns having posted a message fot the worker thread]
79// 11. SctpDataMediaChannel::OnMessage(inboundpacket)
80// 12. SctpDataMediaChannel::OnInboundPacketFromSctpToChannel(inboundpacket)
81// 13. SctpDataMediaChannel::OnDataFromSctpToChannel(data)
82// 14. SctpDataMediaChannel::SignalDataReceived(data)
83// [from the same thread, methods registered/connected to
84// SctpDataMediaChannel are called with the recieved data]
85class SctpDataEngine : public DataEngineInterface {
86 public:
87 SctpDataEngine();
88 virtual ~SctpDataEngine();
89
90 virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type);
91
92 virtual const std::vector<DataCodec>& data_codecs() { return codecs_; }
93
94 private:
95 static int usrsctp_engines_count;
96 std::vector<DataCodec> codecs_;
97};
98
99// TODO(ldixon): Make into a special type of TypedMessageData.
100// Holds data to be passed on to a channel.
101struct SctpInboundPacket;
102
103class SctpDataMediaChannel : public DataMediaChannel,
104 public talk_base::MessageHandler {
105 public:
106 // DataMessageType is used for the SCTP "Payload Protocol Identifier", as
107 // defined in http://tools.ietf.org/html/rfc4960#section-14.4
108 //
109 // For the list of IANA approved values see:
110 // http://www.iana.org/assignments/sctp-parameters/sctp-parameters.xml
111 // The value is not used by SCTP itself. It indicates the protocol running
112 // on top of SCTP.
113 enum PayloadProtocolIdentifier {
114 PPID_NONE = 0, // No protocol is specified.
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000115 // Matches the PPIDs in mozilla source and
116 // https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol Sec. 9
117 // They're not yet assigned by IANA.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118 PPID_CONTROL = 50,
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000119 PPID_BINARY_PARTIAL = 52,
120 PPID_BINARY_LAST = 53,
121 PPID_TEXT_PARTIAL = 54,
122 PPID_TEXT_LAST = 51
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000123 };
124
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000125 typedef std::set<uint32> StreamSet;
126
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000127 // Given a thread which will be used to post messages (received data) to this
128 // SctpDataMediaChannel instance.
129 explicit SctpDataMediaChannel(talk_base::Thread* thread);
130 virtual ~SctpDataMediaChannel();
131
132 // When SetSend is set to true, connects. When set to false, disconnects.
133 // Calling: "SetSend(true); SetSend(false); SetSend(true);" will connect,
134 // disconnect, and reconnect.
135 virtual bool SetSend(bool send);
136 // Unless SetReceive(true) is called, received packets will be discarded.
137 virtual bool SetReceive(bool receive);
138
139 virtual bool AddSendStream(const StreamParams& sp);
140 virtual bool RemoveSendStream(uint32 ssrc);
141 virtual bool AddRecvStream(const StreamParams& sp);
142 virtual bool RemoveRecvStream(uint32 ssrc);
143
144 // Called when Sctp gets data. The data may be a notification or data for
145 // OnSctpInboundData. Called from the worker thread.
146 virtual void OnMessage(talk_base::Message* msg);
147 // Send data down this channel (will be wrapped as SCTP packets then given to
148 // sctp that will then post the network interface by OnMessage).
149 // Returns true iff successful data somewhere on the send-queue/network.
150 virtual bool SendData(const SendDataParams& params,
151 const talk_base::Buffer& payload,
152 SendDataResult* result = NULL);
153 // A packet is received from the network interface. Posted to OnMessage.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000154 virtual void OnPacketReceived(talk_base::Buffer* packet,
155 const talk_base::PacketTime& packet_time);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000156
157 // Exposed to allow Post call from c-callbacks.
158 talk_base::Thread* worker_thread() const { return worker_thread_; }
159
160 // TODO(ldixon): add a DataOptions class to mediachannel.h
161 virtual bool SetOptions(int options) { return false; }
162 virtual int GetOptions() const { return 0; }
163
164 // Many of these things are unused by SCTP, but are needed to fulfill
165 // the MediaChannel interface.
166 // TODO(pthatcher): Cleanup MediaChannel interface, or at least
167 // don't try calling these and return false. Right now, things
168 // don't work if we return false.
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000169 virtual bool SetStartSendBandwidth(int bps) { return true; }
170 virtual bool SetMaxSendBandwidth(int bps) { return true; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000171 virtual bool SetRecvRtpHeaderExtensions(
172 const std::vector<RtpHeaderExtension>& extensions) { return true; }
173 virtual bool SetSendRtpHeaderExtensions(
174 const std::vector<RtpHeaderExtension>& extensions) { return true; }
wu@webrtc.org78187522013-10-07 23:32:02 +0000175 virtual bool SetSendCodecs(const std::vector<DataCodec>& codecs);
176 virtual bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
wu@webrtc.orga9890802013-12-13 00:21:03 +0000177 virtual void OnRtcpReceived(talk_base::Buffer* packet,
178 const talk_base::PacketTime& packet_time) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000179 virtual void OnReadyToSend(bool ready) {}
180
181 // Helper for debugging.
182 void set_debug_name(const std::string& debug_name) {
183 debug_name_ = debug_name;
184 }
185 const std::string& debug_name() const { return debug_name_; }
186
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000187 // Called with the SSID of a remote stream that's been closed.
188 sigslot::signal1<int> SignalStreamClosed;
189
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000190 private:
191 sockaddr_conn GetSctpSockAddr(int port);
192
193 // Creates the socket and connects. Sets sending_ to true.
194 bool Connect();
195 // Closes the socket. Sets sending_ to false.
196 void Disconnect();
197
198 // Returns false when openning the socket failed; when successfull sets
199 // sending_ to true
200 bool OpenSctpSocket();
201 // Sets sending_ to false and sock_ to NULL.
202 void CloseSctpSocket();
203
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000204 // Sends a SCTP_RESET_STREAM for all streams in closing_ssids_.
205 bool SendQueuedStreamResets();
206
207 // Adds a stream.
208 bool AddStream(const StreamParams &sp);
209 // Queues a stream for reset.
210 bool ResetStream(uint32 ssrc);
211
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000212 // Called by OnMessage to send packet on the network.
213 void OnPacketFromSctpToNetwork(talk_base::Buffer* buffer);
214 // Called by OnMessage to decide what to do with the packet.
215 void OnInboundPacketFromSctpToChannel(SctpInboundPacket* packet);
216 void OnDataFromSctpToChannel(const ReceiveDataParams& params,
217 talk_base::Buffer* buffer);
218 void OnNotificationFromSctp(talk_base::Buffer* buffer);
219 void OnNotificationAssocChange(const sctp_assoc_change& change);
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000220
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000221 void OnStreamResetEvent(const struct sctp_stream_reset_event* evt);
222
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 // Responsible for marshalling incoming data to the channels listeners, and
224 // outgoing data to the network interface.
225 talk_base::Thread* worker_thread_;
226 // The local and remote SCTP port to use. These are passed along the wire
227 // and the listener and connector must be using the same port. It is not
wu@webrtc.org78187522013-10-07 23:32:02 +0000228 // related to the ports at the IP level. If set to -1, we default to
229 // kSctpDefaultPort.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000230 int local_port_;
231 int remote_port_;
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000232 struct socket* sock_; // The socket created by usrsctp_socket(...).
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000233
234 // sending_ is true iff there is a connected socket.
235 bool sending_;
236 // receiving_ controls whether inbound packets are thrown away.
237 bool receiving_;
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000238
239 // When a data channel opens a stream, it goes into open_streams_. When we
240 // want to close it, the stream's ID goes into queued_reset_streams_. When
241 // we actually transmit a RE-CONFIG chunk with that stream ID, the ID goes
242 // into sent_reset_streams_. When we get a response RE-CONFIG chunk back
243 // acknowledging the reset, we remove the stream ID from
244 // sent_reset_streams_. We use sent_reset_streams_ to differentiate
245 // between acknowledgment RE-CONFIG and peer-initiated RE-CONFIGs.
246 StreamSet open_streams_;
247 StreamSet queued_reset_streams_;
248 StreamSet sent_reset_streams_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000249
250 // A human-readable name for debugging messages.
251 std::string debug_name_;
252};
253
254} // namespace cricket
255
256#endif // TALK_MEDIA_SCTP_SCTPDATAENGINE_H_