henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1 | /* |
| 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.org | 28654cb | 2013-07-22 21:07:49 +0000 | [diff] [blame] | 31 | #include <errno.h> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 32 | #include <string> |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 33 | #include <vector> |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 34 | |
henrike@webrtc.org | 28654cb | 2013-07-22 21:07:49 +0000 | [diff] [blame] | 35 | namespace cricket { |
| 36 | // Some ERRNO values get re-#defined to WSA* equivalents in some talk/ |
| 37 | // headers. We save the original ones in an enum. |
| 38 | enum PreservedErrno { |
| 39 | SCTP_EINPROGRESS = EINPROGRESS, |
| 40 | SCTP_EWOULDBLOCK = EWOULDBLOCK |
| 41 | }; |
| 42 | } // namespace cricket |
| 43 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 44 | #include "webrtc/base/buffer.h" |
| 45 | #include "webrtc/base/scoped_ptr.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 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" |
| 51 | struct sockaddr_conn; |
| 52 | struct sctp_assoc_change; |
wu@webrtc.org | f6d6ed0 | 2014-01-03 22:08:47 +0000 | [diff] [blame] | 53 | struct sctp_stream_reset_event; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 54 | // Defined by <sys/socket.h> |
| 55 | struct socket; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 56 | namespace cricket { |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 57 | // The highest stream ID (Sid) that SCTP allows, and the number of streams we |
| 58 | // tell SCTP we're going to use. |
wu@webrtc.org | 97077a3 | 2013-10-25 21:18:33 +0000 | [diff] [blame] | 59 | const uint32 kMaxSctpSid = 1023; |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 60 | |
jiayl@webrtc.org | 9c16c39 | 2014-05-01 18:30:30 +0000 | [diff] [blame] | 61 | // This is the default SCTP port to use. It is passed along the wire and the |
| 62 | // connectee and connector must be using the same port. It is not related to the |
| 63 | // ports at the IP level. (Corresponds to: sockaddr_conn.sconn_port in |
| 64 | // usrsctp.h) |
| 65 | const int kSctpDefaultPort = 5000; |
| 66 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 67 | // A DataEngine that interacts with usrsctp. |
| 68 | // |
| 69 | // From channel calls, data flows like this: |
| 70 | // [worker thread (although it can in princple be another thread)] |
| 71 | // 1. SctpDataMediaChannel::SendData(data) |
| 72 | // 2. usrsctp_sendv(data) |
| 73 | // [worker thread returns; sctp thread then calls the following] |
| 74 | // 3. OnSctpOutboundPacket(wrapped_data) |
| 75 | // [sctp thread returns having posted a message for the worker thread] |
| 76 | // 4. SctpDataMediaChannel::OnMessage(wrapped_data) |
| 77 | // 5. SctpDataMediaChannel::OnPacketFromSctpToNetwork(wrapped_data) |
| 78 | // 6. NetworkInterface::SendPacket(wrapped_data) |
| 79 | // 7. ... across network ... a packet is sent back ... |
| 80 | // 8. SctpDataMediaChannel::OnPacketReceived(wrapped_data) |
| 81 | // 9. usrsctp_conninput(wrapped_data) |
| 82 | // [worker thread returns; sctp thread then calls the following] |
| 83 | // 10. OnSctpInboundData(data) |
| 84 | // [sctp thread returns having posted a message fot the worker thread] |
| 85 | // 11. SctpDataMediaChannel::OnMessage(inboundpacket) |
| 86 | // 12. SctpDataMediaChannel::OnInboundPacketFromSctpToChannel(inboundpacket) |
| 87 | // 13. SctpDataMediaChannel::OnDataFromSctpToChannel(data) |
| 88 | // 14. SctpDataMediaChannel::SignalDataReceived(data) |
| 89 | // [from the same thread, methods registered/connected to |
| 90 | // SctpDataMediaChannel are called with the recieved data] |
| 91 | class SctpDataEngine : public DataEngineInterface { |
| 92 | public: |
| 93 | SctpDataEngine(); |
| 94 | virtual ~SctpDataEngine(); |
| 95 | |
| 96 | virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type); |
| 97 | |
| 98 | virtual const std::vector<DataCodec>& data_codecs() { return codecs_; } |
| 99 | |
| 100 | private: |
| 101 | static int usrsctp_engines_count; |
| 102 | std::vector<DataCodec> codecs_; |
| 103 | }; |
| 104 | |
| 105 | // TODO(ldixon): Make into a special type of TypedMessageData. |
| 106 | // Holds data to be passed on to a channel. |
| 107 | struct SctpInboundPacket; |
| 108 | |
| 109 | class SctpDataMediaChannel : public DataMediaChannel, |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 110 | public rtc::MessageHandler { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 111 | public: |
| 112 | // DataMessageType is used for the SCTP "Payload Protocol Identifier", as |
| 113 | // defined in http://tools.ietf.org/html/rfc4960#section-14.4 |
| 114 | // |
| 115 | // For the list of IANA approved values see: |
| 116 | // http://www.iana.org/assignments/sctp-parameters/sctp-parameters.xml |
| 117 | // The value is not used by SCTP itself. It indicates the protocol running |
| 118 | // on top of SCTP. |
| 119 | enum PayloadProtocolIdentifier { |
| 120 | PPID_NONE = 0, // No protocol is specified. |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 121 | // Matches the PPIDs in mozilla source and |
| 122 | // https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol Sec. 9 |
| 123 | // They're not yet assigned by IANA. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 124 | PPID_CONTROL = 50, |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 125 | PPID_BINARY_PARTIAL = 52, |
| 126 | PPID_BINARY_LAST = 53, |
| 127 | PPID_TEXT_PARTIAL = 54, |
| 128 | PPID_TEXT_LAST = 51 |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 129 | }; |
| 130 | |
wu@webrtc.org | f6d6ed0 | 2014-01-03 22:08:47 +0000 | [diff] [blame] | 131 | typedef std::set<uint32> StreamSet; |
| 132 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 133 | // Given a thread which will be used to post messages (received data) to this |
| 134 | // SctpDataMediaChannel instance. |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 135 | explicit SctpDataMediaChannel(rtc::Thread* thread); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 136 | virtual ~SctpDataMediaChannel(); |
| 137 | |
| 138 | // When SetSend is set to true, connects. When set to false, disconnects. |
| 139 | // Calling: "SetSend(true); SetSend(false); SetSend(true);" will connect, |
| 140 | // disconnect, and reconnect. |
| 141 | virtual bool SetSend(bool send); |
| 142 | // Unless SetReceive(true) is called, received packets will be discarded. |
| 143 | virtual bool SetReceive(bool receive); |
| 144 | |
| 145 | virtual bool AddSendStream(const StreamParams& sp); |
| 146 | virtual bool RemoveSendStream(uint32 ssrc); |
| 147 | virtual bool AddRecvStream(const StreamParams& sp); |
| 148 | virtual bool RemoveRecvStream(uint32 ssrc); |
| 149 | |
| 150 | // Called when Sctp gets data. The data may be a notification or data for |
| 151 | // OnSctpInboundData. Called from the worker thread. |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 152 | virtual void OnMessage(rtc::Message* msg); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 153 | // Send data down this channel (will be wrapped as SCTP packets then given to |
| 154 | // sctp that will then post the network interface by OnMessage). |
| 155 | // Returns true iff successful data somewhere on the send-queue/network. |
| 156 | virtual bool SendData(const SendDataParams& params, |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 157 | const rtc::Buffer& payload, |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 158 | SendDataResult* result = NULL); |
| 159 | // A packet is received from the network interface. Posted to OnMessage. |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 160 | virtual void OnPacketReceived(rtc::Buffer* packet, |
| 161 | const rtc::PacketTime& packet_time); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 162 | |
| 163 | // Exposed to allow Post call from c-callbacks. |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 164 | rtc::Thread* worker_thread() const { return worker_thread_; } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 165 | |
| 166 | // TODO(ldixon): add a DataOptions class to mediachannel.h |
| 167 | virtual bool SetOptions(int options) { return false; } |
| 168 | virtual int GetOptions() const { return 0; } |
| 169 | |
| 170 | // Many of these things are unused by SCTP, but are needed to fulfill |
| 171 | // the MediaChannel interface. |
| 172 | // TODO(pthatcher): Cleanup MediaChannel interface, or at least |
| 173 | // don't try calling these and return false. Right now, things |
| 174 | // don't work if we return false. |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 175 | virtual bool SetStartSendBandwidth(int bps) { return true; } |
| 176 | virtual bool SetMaxSendBandwidth(int bps) { return true; } |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 177 | virtual bool SetRecvRtpHeaderExtensions( |
| 178 | const std::vector<RtpHeaderExtension>& extensions) { return true; } |
| 179 | virtual bool SetSendRtpHeaderExtensions( |
| 180 | const std::vector<RtpHeaderExtension>& extensions) { return true; } |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 181 | virtual bool SetSendCodecs(const std::vector<DataCodec>& codecs); |
| 182 | virtual bool SetRecvCodecs(const std::vector<DataCodec>& codecs); |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 183 | virtual void OnRtcpReceived(rtc::Buffer* packet, |
| 184 | const rtc::PacketTime& packet_time) {} |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 185 | virtual void OnReadyToSend(bool ready) {} |
| 186 | |
| 187 | // Helper for debugging. |
| 188 | void set_debug_name(const std::string& debug_name) { |
| 189 | debug_name_ = debug_name; |
| 190 | } |
| 191 | const std::string& debug_name() const { return debug_name_; } |
| 192 | |
| 193 | private: |
| 194 | sockaddr_conn GetSctpSockAddr(int port); |
| 195 | |
| 196 | // Creates the socket and connects. Sets sending_ to true. |
| 197 | bool Connect(); |
| 198 | // Closes the socket. Sets sending_ to false. |
| 199 | void Disconnect(); |
| 200 | |
| 201 | // Returns false when openning the socket failed; when successfull sets |
| 202 | // sending_ to true |
| 203 | bool OpenSctpSocket(); |
| 204 | // Sets sending_ to false and sock_ to NULL. |
| 205 | void CloseSctpSocket(); |
| 206 | |
wu@webrtc.org | f6d6ed0 | 2014-01-03 22:08:47 +0000 | [diff] [blame] | 207 | // Sends a SCTP_RESET_STREAM for all streams in closing_ssids_. |
| 208 | bool SendQueuedStreamResets(); |
| 209 | |
| 210 | // Adds a stream. |
| 211 | bool AddStream(const StreamParams &sp); |
| 212 | // Queues a stream for reset. |
| 213 | bool ResetStream(uint32 ssrc); |
| 214 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 215 | // Called by OnMessage to send packet on the network. |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 216 | void OnPacketFromSctpToNetwork(rtc::Buffer* buffer); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 217 | // Called by OnMessage to decide what to do with the packet. |
| 218 | void OnInboundPacketFromSctpToChannel(SctpInboundPacket* packet); |
| 219 | void OnDataFromSctpToChannel(const ReceiveDataParams& params, |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 220 | rtc::Buffer* buffer); |
| 221 | void OnNotificationFromSctp(rtc::Buffer* buffer); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 222 | void OnNotificationAssocChange(const sctp_assoc_change& change); |
sergeyu@chromium.org | 4b26e2e | 2014-01-15 23:15:54 +0000 | [diff] [blame] | 223 | |
wu@webrtc.org | f6d6ed0 | 2014-01-03 22:08:47 +0000 | [diff] [blame] | 224 | void OnStreamResetEvent(const struct sctp_stream_reset_event* evt); |
| 225 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 226 | // Responsible for marshalling incoming data to the channels listeners, and |
| 227 | // outgoing data to the network interface. |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame^] | 228 | rtc::Thread* worker_thread_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 229 | // The local and remote SCTP port to use. These are passed along the wire |
| 230 | // and the listener and connector must be using the same port. It is not |
wu@webrtc.org | 7818752 | 2013-10-07 23:32:02 +0000 | [diff] [blame] | 231 | // related to the ports at the IP level. If set to -1, we default to |
| 232 | // kSctpDefaultPort. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 233 | int local_port_; |
| 234 | int remote_port_; |
mallinath@webrtc.org | 1112c30 | 2013-09-23 20:34:45 +0000 | [diff] [blame] | 235 | struct socket* sock_; // The socket created by usrsctp_socket(...). |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 236 | |
| 237 | // sending_ is true iff there is a connected socket. |
| 238 | bool sending_; |
| 239 | // receiving_ controls whether inbound packets are thrown away. |
| 240 | bool receiving_; |
wu@webrtc.org | f6d6ed0 | 2014-01-03 22:08:47 +0000 | [diff] [blame] | 241 | |
| 242 | // When a data channel opens a stream, it goes into open_streams_. When we |
| 243 | // want to close it, the stream's ID goes into queued_reset_streams_. When |
| 244 | // we actually transmit a RE-CONFIG chunk with that stream ID, the ID goes |
| 245 | // into sent_reset_streams_. When we get a response RE-CONFIG chunk back |
| 246 | // acknowledging the reset, we remove the stream ID from |
| 247 | // sent_reset_streams_. We use sent_reset_streams_ to differentiate |
| 248 | // between acknowledgment RE-CONFIG and peer-initiated RE-CONFIGs. |
| 249 | StreamSet open_streams_; |
| 250 | StreamSet queued_reset_streams_; |
| 251 | StreamSet sent_reset_streams_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 252 | |
| 253 | // A human-readable name for debugging messages. |
| 254 | std::string debug_name_; |
| 255 | }; |
| 256 | |
| 257 | } // namespace cricket |
| 258 | |
| 259 | #endif // TALK_MEDIA_SCTP_SCTPDATAENGINE_H_ |