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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#include "talk/media/sctp/sctpdataengine.h"
29
henrike@webrtc.org28e20752013-07-10 00:45:36 +000030#include <stdarg.h>
31#include <stdio.h>
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +000032#include <sstream>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000033#include <vector>
34
35#include "talk/base/buffer.h"
36#include "talk/base/helpers.h"
37#include "talk/base/logging.h"
jiayl@webrtc.orga576faf2014-01-29 17:45:53 +000038#include "talk/base/safe_conversions.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039#include "talk/media/base/codec.h"
40#include "talk/media/base/constants.h"
41#include "talk/media/base/streamparams.h"
42#include "usrsctplib/usrsctp.h"
43
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +000044namespace {
45typedef cricket::SctpDataMediaChannel::StreamSet StreamSet;
46// Returns a comma-separated, human-readable list of the stream IDs in 's'
47std::string ListStreams(const StreamSet& s) {
48 std::stringstream result;
49 bool first = true;
wu@webrtc.orge00265e2014-01-07 19:32:40 +000050 for (StreamSet::const_iterator it = s.begin(); it != s.end(); ++it) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +000051 if (!first) {
52 result << ", " << *it;
53 } else {
54 result << *it;
55 first = false;
56 }
57 }
58 return result.str();
59}
60
61// Returns a pipe-separated, human-readable list of the SCTP_STREAM_RESET
62// flags in 'flags'
63std::string ListFlags(int flags) {
64 std::stringstream result;
65 bool first = true;
66 // Skip past the first 12 chars (strlen("SCTP_STREAM_"))
67#define MAKEFLAG(X) { X, #X + 12}
68 struct flaginfo_t {
69 int value;
70 const char* name;
71 } flaginfo[] = {
72 MAKEFLAG(SCTP_STREAM_RESET_INCOMING_SSN),
73 MAKEFLAG(SCTP_STREAM_RESET_OUTGOING_SSN),
74 MAKEFLAG(SCTP_STREAM_RESET_DENIED),
75 MAKEFLAG(SCTP_STREAM_RESET_FAILED),
76 MAKEFLAG(SCTP_STREAM_CHANGE_DENIED)
77 };
78#undef MAKEFLAG
79 for (int i = 0; i < ARRAY_SIZE(flaginfo); ++i) {
80 if (flags & flaginfo[i].value) {
81 if (!first) result << " | ";
82 result << flaginfo[i].name;
83 first = false;
84 }
85 }
86 return result.str();
87}
88
89// Returns a comma-separated, human-readable list of the integers in 'array'.
90// All 'num_elems' of them.
91std::string ListArray(const uint16* array, int num_elems) {
92 std::stringstream result;
93 for (int i = 0; i < num_elems; ++i) {
94 if (i) {
95 result << ", " << array[i];
96 } else {
97 result << array[i];
98 }
99 }
100 return result.str();
101}
102} // namespace
103
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000104namespace cricket {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000105typedef talk_base::ScopedMessageData<SctpInboundPacket> InboundPacketMessage;
106typedef talk_base::ScopedMessageData<talk_base::Buffer> OutboundPacketMessage;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000108// TODO(ldixon): Find where this is defined, and also check is Sctp really
109// respects this.
110static const size_t kSctpMtu = 1280;
111
112enum {
113 MSG_SCTPINBOUNDPACKET = 1, // MessageData is SctpInboundPacket
114 MSG_SCTPOUTBOUNDPACKET = 2, // MessageData is talk_base:Buffer
115};
116
117struct SctpInboundPacket {
118 talk_base::Buffer buffer;
119 ReceiveDataParams params;
120 // The |flags| parameter is used by SCTP to distinguish notification packets
121 // from other types of packets.
122 int flags;
123};
124
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000125// Helper for logging SCTP messages.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000126static void debug_sctp_printf(const char *format, ...) {
127 char s[255];
128 va_list ap;
129 va_start(ap, format);
130 vsnprintf(s, sizeof(s), format, ap);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000131 LOG(LS_INFO) << "SCTP: " << s;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000132 va_end(ap);
133}
134
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000135// Get the PPID to use for the terminating fragment of this type.
136static SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid(
137 cricket::DataMessageType type) {
138 switch (type) {
139 default:
140 case cricket::DMT_NONE:
141 return SctpDataMediaChannel::PPID_NONE;
142 case cricket::DMT_CONTROL:
143 return SctpDataMediaChannel::PPID_CONTROL;
144 case cricket::DMT_BINARY:
145 return SctpDataMediaChannel::PPID_BINARY_LAST;
146 case cricket::DMT_TEXT:
147 return SctpDataMediaChannel::PPID_TEXT_LAST;
148 };
149}
150
151static bool GetDataMediaType(
152 SctpDataMediaChannel::PayloadProtocolIdentifier ppid,
153 cricket::DataMessageType *dest) {
154 ASSERT(dest != NULL);
155 switch (ppid) {
156 case SctpDataMediaChannel::PPID_BINARY_PARTIAL:
157 case SctpDataMediaChannel::PPID_BINARY_LAST:
158 *dest = cricket::DMT_BINARY;
159 return true;
160
161 case SctpDataMediaChannel::PPID_TEXT_PARTIAL:
162 case SctpDataMediaChannel::PPID_TEXT_LAST:
163 *dest = cricket::DMT_TEXT;
164 return true;
165
166 case SctpDataMediaChannel::PPID_CONTROL:
167 *dest = cricket::DMT_CONTROL;
168 return true;
169
170 case SctpDataMediaChannel::PPID_NONE:
171 *dest = cricket::DMT_NONE;
172 return true;
173
174 default:
175 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000176 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177}
178
179// This is the callback usrsctp uses when there's data to send on the network
180// that has been wrapped appropriatly for the SCTP protocol.
181static int OnSctpOutboundPacket(void* addr, void* data, size_t length,
182 uint8_t tos, uint8_t set_df) {
183 SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(addr);
184 LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
185 << "addr: " << addr << "; length: " << length
186 << "; tos: " << std::hex << static_cast<int>(tos)
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000187 << "; set_df: " << std::hex << static_cast<int>(set_df);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000188 // Note: We have to copy the data; the caller will delete it.
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000189 OutboundPacketMessage* msg =
190 new OutboundPacketMessage(new talk_base::Buffer(data, length));
191 channel->worker_thread()->Post(channel, MSG_SCTPOUTBOUNDPACKET, msg);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000192 return 0;
193}
194
195// This is the callback called from usrsctp when data has been received, after
196// a packet has been interpreted and parsed by usrsctp and found to contain
197// payload data. It is called by a usrsctp thread. It is assumed this function
198// will free the memory used by 'data'.
199static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr,
200 void* data, size_t length,
201 struct sctp_rcvinfo rcv, int flags,
202 void* ulp_info) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000203 SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(ulp_info);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204 // Post data to the channel's receiver thread (copying it).
205 // TODO(ldixon): Unclear if copy is needed as this method is responsible for
206 // memory cleanup. But this does simplify code.
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000207 const SctpDataMediaChannel::PayloadProtocolIdentifier ppid =
208 static_cast<SctpDataMediaChannel::PayloadProtocolIdentifier>(
209 talk_base::HostToNetwork32(rcv.rcv_ppid));
210 cricket::DataMessageType type = cricket::DMT_NONE;
211 if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) {
212 // It's neither a notification nor a recognized data packet. Drop it.
213 LOG(LS_ERROR) << "Received an unknown PPID " << ppid
214 << " on an SCTP packet. Dropping.";
215 } else {
216 SctpInboundPacket* packet = new SctpInboundPacket;
217 packet->buffer.SetData(data, length);
218 packet->params.ssrc = rcv.rcv_sid;
219 packet->params.seq_num = rcv.rcv_ssn;
220 packet->params.timestamp = rcv.rcv_tsn;
221 packet->params.type = type;
222 packet->flags = flags;
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000223 // The ownership of |packet| transfers to |msg|.
224 InboundPacketMessage* msg = new InboundPacketMessage(packet);
225 channel->worker_thread()->Post(channel, MSG_SCTPINBOUNDPACKET, msg);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000226 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000227 free(data);
228 return 1;
229}
230
231// Set the initial value of the static SCTP Data Engines reference count.
232int SctpDataEngine::usrsctp_engines_count = 0;
233
wu@webrtc.org0de29502014-02-13 19:54:28 +0000234SctpDataEngine::SctpDataEngine() {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000235 if (usrsctp_engines_count == 0) {
236 // First argument is udp_encapsulation_port, which is not releveant for our
237 // AF_CONN use of sctp.
238 usrsctp_init(0, cricket::OnSctpOutboundPacket, debug_sctp_printf);
239
240 // To turn on/off detailed SCTP debugging. You will also need to have the
241 // SCTP_DEBUG cpp defines flag.
242 // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
243
244 // TODO(ldixon): Consider turning this on/off.
245 usrsctp_sysctl_set_sctp_ecn_enable(0);
246
247 // TODO(ldixon): Consider turning this on/off.
248 // This is not needed right now (we don't do dynamic address changes):
249 // If SCTP Auto-ASCONF is enabled, the peer is informed automatically
250 // when a new address is added or removed. This feature is enabled by
251 // default.
252 // usrsctp_sysctl_set_sctp_auto_asconf(0);
253
254 // TODO(ldixon): Consider turning this on/off.
255 // Add a blackhole sysctl. Setting it to 1 results in no ABORTs
256 // being sent in response to INITs, setting it to 2 results
257 // in no ABORTs being sent for received OOTB packets.
258 // This is similar to the TCP sysctl.
259 //
260 // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
261 // See: http://svnweb.freebsd.org/base?view=revision&revision=229805
262 // usrsctp_sysctl_set_sctp_blackhole(2);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000263
264 // Set the number of default outgoing streams. This is the number we'll
265 // send in the SCTP INIT message. The 'appropriate default' in the
266 // second paragraph of
267 // http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2
268 // is cricket::kMaxSctpSid.
269 usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(
270 cricket::kMaxSctpSid);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000271 }
272 usrsctp_engines_count++;
273
jiayl@webrtc.org9c16c392014-05-01 18:30:30 +0000274 cricket::DataCodec codec(kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, 0);
275 codec.SetParam(kCodecParamPort, kSctpDefaultPort);
276 codecs_.push_back(codec);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000277}
278
279SctpDataEngine::~SctpDataEngine() {
wu@webrtc.org0de29502014-02-13 19:54:28 +0000280 // TODO(ldixon): There is currently a bug in teardown of usrsctp that blocks
281 // indefintely if a finish call made too soon after close calls. So teardown
282 // has been skipped. Once the bug is fixed, retest and enable teardown.
283 // Tracked in webrtc issue 2749.
284 //
285 // usrsctp_engines_count--;
286 // LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
287 // if (usrsctp_engines_count == 0) {
288 // if (usrsctp_finish() != 0) {
289 // LOG(LS_WARNING) << "usrsctp_finish.";
290 // }
291 // }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000292}
293
294DataMediaChannel* SctpDataEngine::CreateChannel(
295 DataChannelType data_channel_type) {
296 if (data_channel_type != DCT_SCTP) {
297 return NULL;
298 }
299 return new SctpDataMediaChannel(talk_base::Thread::Current());
300}
301
302SctpDataMediaChannel::SctpDataMediaChannel(talk_base::Thread* thread)
303 : worker_thread_(thread),
jiayl@webrtc.org9c16c392014-05-01 18:30:30 +0000304 local_port_(kSctpDefaultPort),
305 remote_port_(kSctpDefaultPort),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000306 sock_(NULL),
307 sending_(false),
308 receiving_(false),
309 debug_name_("SctpDataMediaChannel") {
310}
311
312SctpDataMediaChannel::~SctpDataMediaChannel() {
313 CloseSctpSocket();
314}
315
316sockaddr_conn SctpDataMediaChannel::GetSctpSockAddr(int port) {
317 sockaddr_conn sconn = {0};
318 sconn.sconn_family = AF_CONN;
319#ifdef HAVE_SCONN_LEN
320 sconn.sconn_len = sizeof(sockaddr_conn);
321#endif
322 // Note: conversion from int to uint16_t happens here.
323 sconn.sconn_port = talk_base::HostToNetwork16(port);
324 sconn.sconn_addr = this;
325 return sconn;
326}
327
328bool SctpDataMediaChannel::OpenSctpSocket() {
329 if (sock_) {
330 LOG(LS_VERBOSE) << debug_name_
331 << "->Ignoring attempt to re-create existing socket.";
332 return false;
333 }
334 sock_ = usrsctp_socket(AF_CONN, SOCK_STREAM, IPPROTO_SCTP,
335 cricket::OnSctpInboundPacket, NULL, 0, this);
336 if (!sock_) {
337 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to create SCTP socket.";
338 return false;
339 }
340
341 // Make the socket non-blocking. Connect, close, shutdown etc will not block
342 // the thread waiting for the socket operation to complete.
343 if (usrsctp_set_non_blocking(sock_, 1) < 0) {
344 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP to non blocking.";
345 return false;
346 }
347
348 // This ensures that the usrsctp close call deletes the association. This
349 // prevents usrsctp from calling OnSctpOutboundPacket with references to
350 // this class as the address.
351 linger linger_opt;
352 linger_opt.l_onoff = 1;
353 linger_opt.l_linger = 0;
354 if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt,
355 sizeof(linger_opt))) {
356 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SO_LINGER.";
357 return false;
358 }
359
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000360 // Enable stream ID resets.
361 struct sctp_assoc_value stream_rst;
362 stream_rst.assoc_id = SCTP_ALL_ASSOC;
363 stream_rst.assoc_value = 1;
364 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET,
365 &stream_rst, sizeof(stream_rst))) {
366 LOG_ERRNO(LS_ERROR) << debug_name_
367 << "Failed to set SCTP_ENABLE_STREAM_RESET.";
368 return false;
369 }
370
371 // Nagle.
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +0000372 uint32_t nodelay = 1;
373 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay,
374 sizeof(nodelay))) {
375 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_NODELAY.";
376 return false;
377 }
378
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000379 // Subscribe to SCTP event notifications.
380 int event_types[] = {SCTP_ASSOC_CHANGE,
381 SCTP_PEER_ADDR_CHANGE,
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000382 SCTP_SEND_FAILED_EVENT,
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000383 SCTP_SENDER_DRY_EVENT,
384 SCTP_STREAM_RESET_EVENT};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000385 struct sctp_event event = {0};
386 event.se_assoc_id = SCTP_ALL_ASSOC;
387 event.se_on = 1;
388 for (size_t i = 0; i < ARRAY_SIZE(event_types); i++) {
389 event.se_type = event_types[i];
390 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event,
391 sizeof(event)) < 0) {
392 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_EVENT type: "
393 << event.se_type;
394 return false;
395 }
396 }
397
398 // Register this class as an address for usrsctp. This is used by SCTP to
399 // direct the packets received (by the created socket) to this class.
400 usrsctp_register_address(this);
401 sending_ = true;
402 return true;
403}
404
405void SctpDataMediaChannel::CloseSctpSocket() {
406 sending_ = false;
407 if (sock_) {
408 // We assume that SO_LINGER option is set to close the association when
409 // close is called. This means that any pending packets in usrsctp will be
410 // discarded instead of being sent.
411 usrsctp_close(sock_);
412 sock_ = NULL;
413 usrsctp_deregister_address(this);
414 }
415}
416
417bool SctpDataMediaChannel::Connect() {
418 LOG(LS_VERBOSE) << debug_name_ << "->Connect().";
419
420 // If we already have a socket connection, just return.
421 if (sock_) {
422 LOG(LS_WARNING) << debug_name_ << "->Connect(): Ignored as socket "
423 "is already established.";
424 return true;
425 }
426
427 // If no socket (it was closed) try to start it again. This can happen when
428 // the socket we are connecting to closes, does an sctp shutdown handshake,
429 // or behaves unexpectedly causing us to perform a CloseSctpSocket.
430 if (!sock_ && !OpenSctpSocket()) {
431 return false;
432 }
433
434 // Note: conversion from int to uint16_t happens on assignment.
435 sockaddr_conn local_sconn = GetSctpSockAddr(local_port_);
436 if (usrsctp_bind(sock_, reinterpret_cast<sockaddr *>(&local_sconn),
437 sizeof(local_sconn)) < 0) {
438 LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
439 << ("Failed usrsctp_bind");
440 CloseSctpSocket();
441 return false;
442 }
443
444 // Note: conversion from int to uint16_t happens on assignment.
445 sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_);
446 int connect_result = usrsctp_connect(
447 sock_, reinterpret_cast<sockaddr *>(&remote_sconn), sizeof(remote_sconn));
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000448 if (connect_result < 0 && errno != SCTP_EINPROGRESS) {
449 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed usrsctp_connect. got errno="
450 << errno << ", but wanted " << SCTP_EINPROGRESS;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000451 CloseSctpSocket();
452 return false;
453 }
454 return true;
455}
456
457void SctpDataMediaChannel::Disconnect() {
458 // TODO(ldixon): Consider calling |usrsctp_shutdown(sock_, ...)| to do a
459 // shutdown handshake and remove the association.
460 CloseSctpSocket();
461}
462
463bool SctpDataMediaChannel::SetSend(bool send) {
464 if (!sending_ && send) {
465 return Connect();
466 }
467 if (sending_ && !send) {
468 Disconnect();
469 }
470 return true;
471}
472
473bool SctpDataMediaChannel::SetReceive(bool receive) {
474 receiving_ = receive;
475 return true;
476}
477
478bool SctpDataMediaChannel::AddSendStream(const StreamParams& stream) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000479 return AddStream(stream);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000480}
481
482bool SctpDataMediaChannel::RemoveSendStream(uint32 ssrc) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000483 return ResetStream(ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000484}
485
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000486bool SctpDataMediaChannel::AddRecvStream(const StreamParams& stream) {
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000487 // SCTP DataChannels are always bi-directional and calling AddSendStream will
488 // enable both sending and receiving on the stream. So AddRecvStream is a
489 // no-op.
490 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000491}
492
493bool SctpDataMediaChannel::RemoveRecvStream(uint32 ssrc) {
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000494 // SCTP DataChannels are always bi-directional and calling RemoveSendStream
495 // will disable both sending and receiving on the stream. So RemoveRecvStream
496 // is a no-op.
497 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000498}
499
500bool SctpDataMediaChannel::SendData(
501 const SendDataParams& params,
502 const talk_base::Buffer& payload,
503 SendDataResult* result) {
504 if (result) {
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000505 // Preset |result| to assume an error. If SendData succeeds, we'll
506 // overwrite |*result| once more at the end.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000507 *result = SDR_ERROR;
508 }
509
510 if (!sending_) {
511 LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
512 << "Not sending packet with ssrc=" << params.ssrc
513 << " len=" << payload.length() << " before SetSend(true).";
514 return false;
515 }
516
wu@webrtc.org91053e72013-08-10 07:18:04 +0000517 if (params.type != cricket::DMT_CONTROL &&
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000518 open_streams_.find(params.ssrc) == open_streams_.end()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000519 LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
520 << "Not sending data because ssrc is unknown: "
521 << params.ssrc;
522 return false;
523 }
524
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000525 //
526 // Send data using SCTP.
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000527 ssize_t send_res = 0; // result from usrsctp_sendv.
528 struct sctp_sendv_spa spa = {0};
529 spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID;
530 spa.sendv_sndinfo.snd_sid = params.ssrc;
531 spa.sendv_sndinfo.snd_ppid = talk_base::HostToNetwork32(
532 GetPpid(params.type));
533
534 // Ordered implies reliable.
535 if (!params.ordered) {
536 spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED;
537 if (params.max_rtx_count >= 0 || params.max_rtx_ms == 0) {
538 spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
539 spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX;
540 spa.sendv_prinfo.pr_value = params.max_rtx_count;
541 } else {
542 spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
543 spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL;
544 spa.sendv_prinfo.pr_value = params.max_rtx_ms;
545 }
546 }
547
548 // We don't fragment.
549 send_res = usrsctp_sendv(sock_, payload.data(),
550 static_cast<size_t>(payload.length()),
551 NULL, 0, &spa,
jiayl@webrtc.orga576faf2014-01-29 17:45:53 +0000552 talk_base::checked_cast<socklen_t>(sizeof(spa)),
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000553 SCTP_SENDV_SPA, 0);
554 if (send_res < 0) {
jiayl@webrtc.orgf7026cd2014-05-08 16:02:23 +0000555 if (errno == SCTP_EWOULDBLOCK) {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000556 *result = SDR_BLOCK;
557 LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned";
558 } else {
559 LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_
560 << "->SendData(...): "
561 << " usrsctp_sendv: ";
562 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000563 return false;
564 }
565 if (result) {
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000566 // Only way out now is success.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000567 *result = SDR_SUCCESS;
568 }
569 return true;
570}
571
572// Called by network interface when a packet has been received.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000573void SctpDataMediaChannel::OnPacketReceived(
574 talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000575 LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...): " << " length="
576 << packet->length() << ", sending: " << sending_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000577 // Only give receiving packets to usrsctp after if connected. This enables two
578 // peers to each make a connect call, but for them not to receive an INIT
579 // packet before they have called connect; least the last receiver of the INIT
580 // packet will have called connect, and a connection will be established.
581 if (sending_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000582 // Pass received packet to SCTP stack. Once processed by usrsctp, the data
583 // will be will be given to the global OnSctpInboundData, and then,
584 // marshalled by a Post and handled with OnMessage.
585 usrsctp_conninput(this, packet->data(), packet->length(), 0);
586 } else {
587 // TODO(ldixon): Consider caching the packet for very slightly better
588 // reliability.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000589 }
590}
591
592void SctpDataMediaChannel::OnInboundPacketFromSctpToChannel(
593 SctpInboundPacket* packet) {
594 LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
595 << "Received SCTP data:"
596 << " ssrc=" << packet->params.ssrc
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000597 << " notification: " << (packet->flags & MSG_NOTIFICATION)
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000598 << " length=" << packet->buffer.length();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000599 // Sending a packet with data == NULL (no data) is SCTPs "close the
600 // connection" message. This sets sock_ = NULL;
601 if (!packet->buffer.length() || !packet->buffer.data()) {
602 LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
603 "No data, closing.";
604 return;
605 }
606 if (packet->flags & MSG_NOTIFICATION) {
607 OnNotificationFromSctp(&packet->buffer);
608 } else {
609 OnDataFromSctpToChannel(packet->params, &packet->buffer);
610 }
611}
612
613void SctpDataMediaChannel::OnDataFromSctpToChannel(
614 const ReceiveDataParams& params, talk_base::Buffer* buffer) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000615 if (receiving_) {
616 LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): "
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000617 << "Posting with length: " << buffer->length()
618 << " on stream " << params.ssrc;
619 // Reports all received messages to upper layers, no matter whether the sid
620 // is known.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000621 SignalDataReceived(params, buffer->data(), buffer->length());
622 } else {
623 LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): "
624 << "Not receiving packet with sid=" << params.ssrc
625 << " len=" << buffer->length()
626 << " before SetReceive(true).";
627 }
628}
629
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000630bool SctpDataMediaChannel::AddStream(const StreamParams& stream) {
631 if (!stream.has_ssrcs()) {
632 return false;
633 }
634
635 const uint32 ssrc = stream.first_ssrc();
636 if (open_streams_.find(ssrc) != open_streams_.end()) {
henrika@webrtc.orgaebb1ad2014-01-14 10:00:58 +0000637 LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000638 << "Not adding data stream '" << stream.id
639 << "' with ssrc=" << ssrc
640 << " because stream is already open.";
641 return false;
642 } else if (queued_reset_streams_.find(ssrc) != queued_reset_streams_.end()
643 || sent_reset_streams_.find(ssrc) != sent_reset_streams_.end()) {
644 LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
645 << "Not adding data stream '" << stream.id
646 << "' with ssrc=" << ssrc
647 << " because stream is still closing.";
648 return false;
649 }
650
651 open_streams_.insert(ssrc);
652 return true;
653}
654
655bool SctpDataMediaChannel::ResetStream(uint32 ssrc) {
656 // We typically get this called twice for the same stream, once each for
657 // Send and Recv.
658 StreamSet::iterator found = open_streams_.find(ssrc);
659
660 if (found == open_streams_.end()) {
661 LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): "
662 << "stream not found.";
663 return false;
664 } else {
665 LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): "
666 << "Removing and queuing RE-CONFIG chunk.";
667 open_streams_.erase(found);
668 }
669
670 // SCTP won't let you have more than one stream reset pending at a time, but
671 // you can close multiple streams in a single reset. So, we keep an internal
672 // queue of streams-to-reset, and send them as one reset message in
673 // SendQueuedStreamResets().
674 queued_reset_streams_.insert(ssrc);
675
676 // Signal our stream-reset logic that it should try to send now, if it can.
677 SendQueuedStreamResets();
678
679 // The stream will actually get removed when we get the acknowledgment.
680 return true;
681}
682
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000683void SctpDataMediaChannel::OnNotificationFromSctp(talk_base::Buffer* buffer) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000684 const sctp_notification& notification =
685 reinterpret_cast<const sctp_notification&>(*buffer->data());
686 ASSERT(notification.sn_header.sn_length == buffer->length());
687
688 // TODO(ldixon): handle notifications appropriately.
689 switch (notification.sn_header.sn_type) {
690 case SCTP_ASSOC_CHANGE:
691 LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE";
692 OnNotificationAssocChange(notification.sn_assoc_change);
693 break;
694 case SCTP_REMOTE_ERROR:
695 LOG(LS_INFO) << "SCTP_REMOTE_ERROR";
696 break;
697 case SCTP_SHUTDOWN_EVENT:
698 LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT";
699 break;
700 case SCTP_ADAPTATION_INDICATION:
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000701 LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION";
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000702 break;
703 case SCTP_PARTIAL_DELIVERY_EVENT:
704 LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT";
705 break;
706 case SCTP_AUTHENTICATION_EVENT:
707 LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT";
708 break;
709 case SCTP_SENDER_DRY_EVENT:
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000710 LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT";
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000711 SignalReadyToSend(true);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000712 break;
713 // TODO(ldixon): Unblock after congestion.
714 case SCTP_NOTIFICATIONS_STOPPED_EVENT:
715 LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT";
716 break;
717 case SCTP_SEND_FAILED_EVENT:
718 LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT";
719 break;
720 case SCTP_STREAM_RESET_EVENT:
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000721 OnStreamResetEvent(&notification.sn_strreset_event);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000722 break;
723 case SCTP_ASSOC_RESET_EVENT:
724 LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT";
725 break;
726 case SCTP_STREAM_CHANGE_EVENT:
727 LOG(LS_INFO) << "SCTP_STREAM_CHANGE_EVENT";
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000728 // An acknowledgment we get after our stream resets have gone through,
729 // if they've failed. We log the message, but don't react -- we don't
730 // keep around the last-transmitted set of SSIDs we wanted to close for
731 // error recovery. It doesn't seem likely to occur, and if so, likely
732 // harmless within the lifetime of a single SCTP association.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000733 break;
734 default:
735 LOG(LS_WARNING) << "Unknown SCTP event: "
736 << notification.sn_header.sn_type;
737 break;
738 }
739}
740
741void SctpDataMediaChannel::OnNotificationAssocChange(
742 const sctp_assoc_change& change) {
743 switch (change.sac_state) {
744 case SCTP_COMM_UP:
745 LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP";
746 break;
747 case SCTP_COMM_LOST:
748 LOG(LS_INFO) << "Association change SCTP_COMM_LOST";
749 break;
750 case SCTP_RESTART:
751 LOG(LS_INFO) << "Association change SCTP_RESTART";
752 break;
753 case SCTP_SHUTDOWN_COMP:
754 LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP";
755 break;
756 case SCTP_CANT_STR_ASSOC:
757 LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC";
758 break;
759 default:
760 LOG(LS_INFO) << "Association change UNKNOWN";
761 break;
762 }
763}
764
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000765void SctpDataMediaChannel::OnStreamResetEvent(
766 const struct sctp_stream_reset_event* evt) {
767 // A stream reset always involves two RE-CONFIG chunks for us -- we always
768 // simultaneously reset a sid's sequence number in both directions. The
769 // requesting side transmits a RE-CONFIG chunk and waits for the peer to send
770 // one back. Both sides get this SCTP_STREAM_RESET_EVENT when they receive
771 // RE-CONFIGs.
772 const int num_ssrcs = (evt->strreset_length - sizeof(*evt)) /
773 sizeof(evt->strreset_stream_list[0]);
774 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
775 << "): Flags = 0x"
776 << std::hex << evt->strreset_flags << " ("
777 << ListFlags(evt->strreset_flags) << ")";
778 LOG(LS_VERBOSE) << "Assoc = " << evt->strreset_assoc_id << ", Streams = ["
779 << ListArray(evt->strreset_stream_list, num_ssrcs)
780 << "], Open: ["
781 << ListStreams(open_streams_) << "], Q'd: ["
782 << ListStreams(queued_reset_streams_) << "], Sent: ["
783 << ListStreams(sent_reset_streams_) << "]";
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000784
785 // If both sides try to reset some streams at the same time (even if they're
786 // disjoint sets), we can get reset failures.
787 if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) {
788 // OK, just try again. The stream IDs sent over when the RESET_FAILED flag
789 // is set seem to be garbage values. Ignore them.
790 queued_reset_streams_.insert(
791 sent_reset_streams_.begin(),
792 sent_reset_streams_.end());
793 sent_reset_streams_.clear();
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000794
795 } else if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) {
796 // Each side gets an event for each direction of a stream. That is,
797 // closing sid k will make each side receive INCOMING and OUTGOING reset
798 // events for k. As per RFC6525, Section 5, paragraph 2, each side will
799 // get an INCOMING event first.
800 for (int i = 0; i < num_ssrcs; i++) {
801 const int stream_id = evt->strreset_stream_list[i];
802
803 // See if this stream ID was closed by our peer or ourselves.
804 StreamSet::iterator it = sent_reset_streams_.find(stream_id);
805
806 // The reset was requested locally.
807 if (it != sent_reset_streams_.end()) {
808 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
809 << "): local sid " << stream_id << " acknowledged.";
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000810 sent_reset_streams_.erase(it);
811
812 } else if ((it = open_streams_.find(stream_id))
813 != open_streams_.end()) {
814 // The peer requested the reset.
815 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
816 << "): closing sid " << stream_id;
817 open_streams_.erase(it);
buildbot@webrtc.org1d66be22014-05-29 22:54:24 +0000818 SignalStreamClosedRemotely(stream_id);
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000819
820 } else if ((it = queued_reset_streams_.find(stream_id))
821 != queued_reset_streams_.end()) {
822 // The peer requested the reset, but there was a local reset
823 // queued.
824 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
825 << "): double-sided close for sid " << stream_id;
826 // Both sides want the stream closed, and the peer got to send the
827 // RE-CONFIG first. Treat it like the local Remove(Send|Recv)Stream
828 // finished quickly.
829 queued_reset_streams_.erase(it);
830
831 } else {
832 // This stream is unknown. Sometimes this can be from an
833 // RESET_FAILED-related retransmit.
834 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
835 << "): Unknown sid " << stream_id;
836 }
837 }
838 }
839
jiayl@webrtc.org1a6c6282014-06-12 21:59:29 +0000840 // Always try to send the queued RESET because this call indicates that the
841 // last local RESET or remote RESET has made some progress.
842 SendQueuedStreamResets();
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000843}
844
wu@webrtc.org78187522013-10-07 23:32:02 +0000845// Puts the specified |param| from the codec identified by |id| into |dest|
846// and returns true. Or returns false if it wasn't there, leaving |dest|
847// untouched.
848static bool GetCodecIntParameter(const std::vector<DataCodec>& codecs,
849 int id, const std::string& name,
850 const std::string& param, int* dest) {
851 std::string value;
852 Codec match_pattern;
853 match_pattern.id = id;
854 match_pattern.name = name;
855 for (size_t i = 0; i < codecs.size(); ++i) {
856 if (codecs[i].Matches(match_pattern)) {
857 if (codecs[i].GetParam(param, &value)) {
858 *dest = talk_base::FromString<int>(value);
859 return true;
860 }
861 }
862 }
863 return false;
864}
865
866bool SctpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
867 return GetCodecIntParameter(
868 codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort,
869 &remote_port_);
870}
871
872bool SctpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
873 return GetCodecIntParameter(
874 codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort,
875 &local_port_);
876}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000877
878void SctpDataMediaChannel::OnPacketFromSctpToNetwork(
879 talk_base::Buffer* buffer) {
880 if (buffer->length() > kSctpMtu) {
881 LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): "
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000882 << "SCTP seems to have made a packet that is bigger "
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000883 "than its official MTU.";
884 }
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000885 MediaChannel::SendPacket(buffer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000886}
887
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000888bool SctpDataMediaChannel::SendQueuedStreamResets() {
889 if (!sent_reset_streams_.empty() || queued_reset_streams_.empty())
890 return true;
891
892 LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ << "]: Sending ["
893 << ListStreams(queued_reset_streams_) << "], Open: ["
894 << ListStreams(open_streams_) << "], Sent: ["
895 << ListStreams(sent_reset_streams_) << "]";
896
897 const size_t num_streams = queued_reset_streams_.size();
898 const size_t num_bytes = sizeof(struct sctp_reset_streams)
899 + (num_streams * sizeof(uint16));
900
901 std::vector<uint8> reset_stream_buf(num_bytes, 0);
902 struct sctp_reset_streams* resetp = reinterpret_cast<sctp_reset_streams*>(
903 &reset_stream_buf[0]);
904 resetp->srs_assoc_id = SCTP_ALL_ASSOC;
905 resetp->srs_flags = SCTP_STREAM_RESET_INCOMING | SCTP_STREAM_RESET_OUTGOING;
jiayl@webrtc.orga576faf2014-01-29 17:45:53 +0000906 resetp->srs_number_streams = talk_base::checked_cast<uint16_t>(num_streams);
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000907 int result_idx = 0;
908 for (StreamSet::iterator it = queued_reset_streams_.begin();
909 it != queued_reset_streams_.end(); ++it) {
910 resetp->srs_stream_list[result_idx++] = *it;
911 }
912
jiayl@webrtc.orga576faf2014-01-29 17:45:53 +0000913 int ret = usrsctp_setsockopt(
914 sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp,
915 talk_base::checked_cast<socklen_t>(reset_stream_buf.size()));
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000916 if (ret < 0) {
917 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to send a stream reset for "
918 << num_streams << " streams";
919 return false;
920 }
921
922 // sent_reset_streams_ is empty, and all the queued_reset_streams_ go into
923 // it now.
924 queued_reset_streams_.swap(sent_reset_streams_);
925 return true;
926}
927
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000928void SctpDataMediaChannel::OnMessage(talk_base::Message* msg) {
929 switch (msg->message_id) {
930 case MSG_SCTPINBOUNDPACKET: {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000931 talk_base::scoped_ptr<InboundPacketMessage> pdata(
932 static_cast<InboundPacketMessage*>(msg->pdata));
933 OnInboundPacketFromSctpToChannel(pdata->data().get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000934 break;
935 }
936 case MSG_SCTPOUTBOUNDPACKET: {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000937 talk_base::scoped_ptr<OutboundPacketMessage> pdata(
938 static_cast<OutboundPacketMessage*>(msg->pdata));
939 OnPacketFromSctpToNetwork(pdata->data().get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000940 break;
941 }
942 }
943}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000944} // namespace cricket