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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
wu@webrtc.org1d1ffc92013-10-16 18:12:02 +000035#include "talk/app/webrtc/datachannelinterface.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000036#include "talk/base/buffer.h"
37#include "talk/base/helpers.h"
38#include "talk/base/logging.h"
39#include "talk/media/base/codec.h"
40#include "talk/media/base/constants.h"
41#include "talk/media/base/streamparams.h"
wu@webrtc.org1d1ffc92013-10-16 18:12:02 +000042#include "talk/media/sctp/sctputils.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000043#include "usrsctplib/usrsctp.h"
44
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +000045namespace {
46typedef cricket::SctpDataMediaChannel::StreamSet StreamSet;
47// Returns a comma-separated, human-readable list of the stream IDs in 's'
48std::string ListStreams(const StreamSet& s) {
49 std::stringstream result;
50 bool first = true;
51 for (StreamSet::iterator it = s.begin(); it != s.end(); ++it) {
52 if (!first) {
53 result << ", " << *it;
54 } else {
55 result << *it;
56 first = false;
57 }
58 }
59 return result.str();
60}
61
62// Returns a pipe-separated, human-readable list of the SCTP_STREAM_RESET
63// flags in 'flags'
64std::string ListFlags(int flags) {
65 std::stringstream result;
66 bool first = true;
67 // Skip past the first 12 chars (strlen("SCTP_STREAM_"))
68#define MAKEFLAG(X) { X, #X + 12}
69 struct flaginfo_t {
70 int value;
71 const char* name;
72 } flaginfo[] = {
73 MAKEFLAG(SCTP_STREAM_RESET_INCOMING_SSN),
74 MAKEFLAG(SCTP_STREAM_RESET_OUTGOING_SSN),
75 MAKEFLAG(SCTP_STREAM_RESET_DENIED),
76 MAKEFLAG(SCTP_STREAM_RESET_FAILED),
77 MAKEFLAG(SCTP_STREAM_CHANGE_DENIED)
78 };
79#undef MAKEFLAG
80 for (int i = 0; i < ARRAY_SIZE(flaginfo); ++i) {
81 if (flags & flaginfo[i].value) {
82 if (!first) result << " | ";
83 result << flaginfo[i].name;
84 first = false;
85 }
86 }
87 return result.str();
88}
89
90// Returns a comma-separated, human-readable list of the integers in 'array'.
91// All 'num_elems' of them.
92std::string ListArray(const uint16* array, int num_elems) {
93 std::stringstream result;
94 for (int i = 0; i < num_elems; ++i) {
95 if (i) {
96 result << ", " << array[i];
97 } else {
98 result << array[i];
99 }
100 }
101 return result.str();
102}
103} // namespace
104
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000105namespace cricket {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000106typedef talk_base::ScopedMessageData<SctpInboundPacket> InboundPacketMessage;
107typedef talk_base::ScopedMessageData<talk_base::Buffer> OutboundPacketMessage;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000108
109// This is the SCTP port to use. It is passed along the wire and the listener
110// and connector must be using the same port. It is not related to the ports at
111// the IP level. (Corresponds to: sockaddr_conn.sconn_port in usrsctp.h)
112//
113// TODO(ldixon): Allow port to be set from higher level code.
114static const int kSctpDefaultPort = 5001;
115// TODO(ldixon): Find where this is defined, and also check is Sctp really
116// respects this.
117static const size_t kSctpMtu = 1280;
118
119enum {
120 MSG_SCTPINBOUNDPACKET = 1, // MessageData is SctpInboundPacket
121 MSG_SCTPOUTBOUNDPACKET = 2, // MessageData is talk_base:Buffer
122};
123
124struct SctpInboundPacket {
125 talk_base::Buffer buffer;
126 ReceiveDataParams params;
127 // The |flags| parameter is used by SCTP to distinguish notification packets
128 // from other types of packets.
129 int flags;
130};
131
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000132// Helper for logging SCTP messages.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000133static void debug_sctp_printf(const char *format, ...) {
134 char s[255];
135 va_list ap;
136 va_start(ap, format);
137 vsnprintf(s, sizeof(s), format, ap);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000138 LOG(LS_INFO) << "SCTP: " << s;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000139 va_end(ap);
140}
141
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000142// Get the PPID to use for the terminating fragment of this type.
143static SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid(
144 cricket::DataMessageType type) {
145 switch (type) {
146 default:
147 case cricket::DMT_NONE:
148 return SctpDataMediaChannel::PPID_NONE;
149 case cricket::DMT_CONTROL:
150 return SctpDataMediaChannel::PPID_CONTROL;
151 case cricket::DMT_BINARY:
152 return SctpDataMediaChannel::PPID_BINARY_LAST;
153 case cricket::DMT_TEXT:
154 return SctpDataMediaChannel::PPID_TEXT_LAST;
155 };
156}
157
158static bool GetDataMediaType(
159 SctpDataMediaChannel::PayloadProtocolIdentifier ppid,
160 cricket::DataMessageType *dest) {
161 ASSERT(dest != NULL);
162 switch (ppid) {
163 case SctpDataMediaChannel::PPID_BINARY_PARTIAL:
164 case SctpDataMediaChannel::PPID_BINARY_LAST:
165 *dest = cricket::DMT_BINARY;
166 return true;
167
168 case SctpDataMediaChannel::PPID_TEXT_PARTIAL:
169 case SctpDataMediaChannel::PPID_TEXT_LAST:
170 *dest = cricket::DMT_TEXT;
171 return true;
172
173 case SctpDataMediaChannel::PPID_CONTROL:
174 *dest = cricket::DMT_CONTROL;
175 return true;
176
177 case SctpDataMediaChannel::PPID_NONE:
178 *dest = cricket::DMT_NONE;
179 return true;
180
181 default:
182 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000183 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000184}
185
186// This is the callback usrsctp uses when there's data to send on the network
187// that has been wrapped appropriatly for the SCTP protocol.
188static int OnSctpOutboundPacket(void* addr, void* data, size_t length,
189 uint8_t tos, uint8_t set_df) {
190 SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(addr);
191 LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
192 << "addr: " << addr << "; length: " << length
193 << "; tos: " << std::hex << static_cast<int>(tos)
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000194 << "; set_df: " << std::hex << static_cast<int>(set_df);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000195 // Note: We have to copy the data; the caller will delete it.
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000196 OutboundPacketMessage* msg =
197 new OutboundPacketMessage(new talk_base::Buffer(data, length));
198 channel->worker_thread()->Post(channel, MSG_SCTPOUTBOUNDPACKET, msg);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000199 return 0;
200}
201
202// This is the callback called from usrsctp when data has been received, after
203// a packet has been interpreted and parsed by usrsctp and found to contain
204// payload data. It is called by a usrsctp thread. It is assumed this function
205// will free the memory used by 'data'.
206static int OnSctpInboundPacket(struct socket* sock, union sctp_sockstore addr,
207 void* data, size_t length,
208 struct sctp_rcvinfo rcv, int flags,
209 void* ulp_info) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000210 SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(ulp_info);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000211 // Post data to the channel's receiver thread (copying it).
212 // TODO(ldixon): Unclear if copy is needed as this method is responsible for
213 // memory cleanup. But this does simplify code.
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000214 const SctpDataMediaChannel::PayloadProtocolIdentifier ppid =
215 static_cast<SctpDataMediaChannel::PayloadProtocolIdentifier>(
216 talk_base::HostToNetwork32(rcv.rcv_ppid));
217 cricket::DataMessageType type = cricket::DMT_NONE;
218 if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) {
219 // It's neither a notification nor a recognized data packet. Drop it.
220 LOG(LS_ERROR) << "Received an unknown PPID " << ppid
221 << " on an SCTP packet. Dropping.";
222 } else {
223 SctpInboundPacket* packet = new SctpInboundPacket;
224 packet->buffer.SetData(data, length);
225 packet->params.ssrc = rcv.rcv_sid;
226 packet->params.seq_num = rcv.rcv_ssn;
227 packet->params.timestamp = rcv.rcv_tsn;
228 packet->params.type = type;
229 packet->flags = flags;
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000230 // The ownership of |packet| transfers to |msg|.
231 InboundPacketMessage* msg = new InboundPacketMessage(packet);
232 channel->worker_thread()->Post(channel, MSG_SCTPINBOUNDPACKET, msg);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000233 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000234 free(data);
235 return 1;
236}
237
238// Set the initial value of the static SCTP Data Engines reference count.
239int SctpDataEngine::usrsctp_engines_count = 0;
240
241SctpDataEngine::SctpDataEngine() {
242 if (usrsctp_engines_count == 0) {
243 // First argument is udp_encapsulation_port, which is not releveant for our
244 // AF_CONN use of sctp.
245 usrsctp_init(0, cricket::OnSctpOutboundPacket, debug_sctp_printf);
246
247 // To turn on/off detailed SCTP debugging. You will also need to have the
248 // SCTP_DEBUG cpp defines flag.
249 // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
250
251 // TODO(ldixon): Consider turning this on/off.
252 usrsctp_sysctl_set_sctp_ecn_enable(0);
253
254 // TODO(ldixon): Consider turning this on/off.
255 // This is not needed right now (we don't do dynamic address changes):
256 // If SCTP Auto-ASCONF is enabled, the peer is informed automatically
257 // when a new address is added or removed. This feature is enabled by
258 // default.
259 // usrsctp_sysctl_set_sctp_auto_asconf(0);
260
261 // TODO(ldixon): Consider turning this on/off.
262 // Add a blackhole sysctl. Setting it to 1 results in no ABORTs
263 // being sent in response to INITs, setting it to 2 results
264 // in no ABORTs being sent for received OOTB packets.
265 // This is similar to the TCP sysctl.
266 //
267 // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
268 // See: http://svnweb.freebsd.org/base?view=revision&revision=229805
269 // usrsctp_sysctl_set_sctp_blackhole(2);
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000270
271 // Set the number of default outgoing streams. This is the number we'll
272 // send in the SCTP INIT message. The 'appropriate default' in the
273 // second paragraph of
274 // http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2
275 // is cricket::kMaxSctpSid.
276 usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(
277 cricket::kMaxSctpSid);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000278 }
279 usrsctp_engines_count++;
280
281 // We don't put in a codec because we don't want one offered when we
282 // use the hybrid data engine.
283 // codecs_.push_back(cricket::DataCodec( kGoogleSctpDataCodecId,
284 // kGoogleSctpDataCodecName, 0));
285}
286
287SctpDataEngine::~SctpDataEngine() {
288 // TODO(ldixon): There is currently a bug in teardown of usrsctp that blocks
289 // indefintely if a finish call made too soon after close calls. So teardown
290 // has been skipped. Once the bug is fixed, retest and enable teardown.
291 //
292 // usrsctp_engines_count--;
293 // LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
294 // if (usrsctp_engines_count == 0) {
295 // if (usrsctp_finish() != 0) {
296 // LOG(LS_WARNING) << "usrsctp_finish.";
297 // }
298 // }
299}
300
301DataMediaChannel* SctpDataEngine::CreateChannel(
302 DataChannelType data_channel_type) {
303 if (data_channel_type != DCT_SCTP) {
304 return NULL;
305 }
306 return new SctpDataMediaChannel(talk_base::Thread::Current());
307}
308
309SctpDataMediaChannel::SctpDataMediaChannel(talk_base::Thread* thread)
310 : worker_thread_(thread),
wu@webrtc.org78187522013-10-07 23:32:02 +0000311 local_port_(-1),
312 remote_port_(-1),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000313 sock_(NULL),
314 sending_(false),
315 receiving_(false),
316 debug_name_("SctpDataMediaChannel") {
317}
318
319SctpDataMediaChannel::~SctpDataMediaChannel() {
320 CloseSctpSocket();
321}
322
323sockaddr_conn SctpDataMediaChannel::GetSctpSockAddr(int port) {
324 sockaddr_conn sconn = {0};
325 sconn.sconn_family = AF_CONN;
326#ifdef HAVE_SCONN_LEN
327 sconn.sconn_len = sizeof(sockaddr_conn);
328#endif
329 // Note: conversion from int to uint16_t happens here.
330 sconn.sconn_port = talk_base::HostToNetwork16(port);
331 sconn.sconn_addr = this;
332 return sconn;
333}
334
335bool SctpDataMediaChannel::OpenSctpSocket() {
336 if (sock_) {
337 LOG(LS_VERBOSE) << debug_name_
338 << "->Ignoring attempt to re-create existing socket.";
339 return false;
340 }
341 sock_ = usrsctp_socket(AF_CONN, SOCK_STREAM, IPPROTO_SCTP,
342 cricket::OnSctpInboundPacket, NULL, 0, this);
343 if (!sock_) {
344 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to create SCTP socket.";
345 return false;
346 }
347
348 // Make the socket non-blocking. Connect, close, shutdown etc will not block
349 // the thread waiting for the socket operation to complete.
350 if (usrsctp_set_non_blocking(sock_, 1) < 0) {
351 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP to non blocking.";
352 return false;
353 }
354
355 // This ensures that the usrsctp close call deletes the association. This
356 // prevents usrsctp from calling OnSctpOutboundPacket with references to
357 // this class as the address.
358 linger linger_opt;
359 linger_opt.l_onoff = 1;
360 linger_opt.l_linger = 0;
361 if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt,
362 sizeof(linger_opt))) {
363 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SO_LINGER.";
364 return false;
365 }
366
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000367 // Enable stream ID resets.
368 struct sctp_assoc_value stream_rst;
369 stream_rst.assoc_id = SCTP_ALL_ASSOC;
370 stream_rst.assoc_value = 1;
371 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET,
372 &stream_rst, sizeof(stream_rst))) {
373 LOG_ERRNO(LS_ERROR) << debug_name_
374 << "Failed to set SCTP_ENABLE_STREAM_RESET.";
375 return false;
376 }
377
378 // Nagle.
sergeyu@chromium.orga59696b2013-09-13 23:48:58 +0000379 uint32_t nodelay = 1;
380 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay,
381 sizeof(nodelay))) {
382 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_NODELAY.";
383 return false;
384 }
385
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000386 // Subscribe to SCTP event notifications.
387 int event_types[] = {SCTP_ASSOC_CHANGE,
388 SCTP_PEER_ADDR_CHANGE,
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000389 SCTP_SEND_FAILED_EVENT,
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000390 SCTP_SENDER_DRY_EVENT,
391 SCTP_STREAM_RESET_EVENT};
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000392 struct sctp_event event = {0};
393 event.se_assoc_id = SCTP_ALL_ASSOC;
394 event.se_on = 1;
395 for (size_t i = 0; i < ARRAY_SIZE(event_types); i++) {
396 event.se_type = event_types[i];
397 if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event,
398 sizeof(event)) < 0) {
399 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to set SCTP_EVENT type: "
400 << event.se_type;
401 return false;
402 }
403 }
404
405 // Register this class as an address for usrsctp. This is used by SCTP to
406 // direct the packets received (by the created socket) to this class.
407 usrsctp_register_address(this);
408 sending_ = true;
409 return true;
410}
411
412void SctpDataMediaChannel::CloseSctpSocket() {
413 sending_ = false;
414 if (sock_) {
415 // We assume that SO_LINGER option is set to close the association when
416 // close is called. This means that any pending packets in usrsctp will be
417 // discarded instead of being sent.
418 usrsctp_close(sock_);
419 sock_ = NULL;
420 usrsctp_deregister_address(this);
421 }
422}
423
424bool SctpDataMediaChannel::Connect() {
425 LOG(LS_VERBOSE) << debug_name_ << "->Connect().";
wu@webrtc.org78187522013-10-07 23:32:02 +0000426 if (remote_port_ < 0) {
427 remote_port_ = kSctpDefaultPort;
428 }
429 if (local_port_ < 0) {
430 local_port_ = kSctpDefaultPort;
431 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000432
433 // If we already have a socket connection, just return.
434 if (sock_) {
435 LOG(LS_WARNING) << debug_name_ << "->Connect(): Ignored as socket "
436 "is already established.";
437 return true;
438 }
439
440 // If no socket (it was closed) try to start it again. This can happen when
441 // the socket we are connecting to closes, does an sctp shutdown handshake,
442 // or behaves unexpectedly causing us to perform a CloseSctpSocket.
443 if (!sock_ && !OpenSctpSocket()) {
444 return false;
445 }
446
447 // Note: conversion from int to uint16_t happens on assignment.
448 sockaddr_conn local_sconn = GetSctpSockAddr(local_port_);
449 if (usrsctp_bind(sock_, reinterpret_cast<sockaddr *>(&local_sconn),
450 sizeof(local_sconn)) < 0) {
451 LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
452 << ("Failed usrsctp_bind");
453 CloseSctpSocket();
454 return false;
455 }
456
457 // Note: conversion from int to uint16_t happens on assignment.
458 sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_);
459 int connect_result = usrsctp_connect(
460 sock_, reinterpret_cast<sockaddr *>(&remote_sconn), sizeof(remote_sconn));
henrike@webrtc.org28654cb2013-07-22 21:07:49 +0000461 if (connect_result < 0 && errno != SCTP_EINPROGRESS) {
462 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed usrsctp_connect. got errno="
463 << errno << ", but wanted " << SCTP_EINPROGRESS;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000464 CloseSctpSocket();
465 return false;
466 }
467 return true;
468}
469
470void SctpDataMediaChannel::Disconnect() {
471 // TODO(ldixon): Consider calling |usrsctp_shutdown(sock_, ...)| to do a
472 // shutdown handshake and remove the association.
473 CloseSctpSocket();
474}
475
476bool SctpDataMediaChannel::SetSend(bool send) {
477 if (!sending_ && send) {
478 return Connect();
479 }
480 if (sending_ && !send) {
481 Disconnect();
482 }
483 return true;
484}
485
486bool SctpDataMediaChannel::SetReceive(bool receive) {
487 receiving_ = receive;
488 return true;
489}
490
491bool SctpDataMediaChannel::AddSendStream(const StreamParams& stream) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000492 return AddStream(stream);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000493}
494
495bool SctpDataMediaChannel::RemoveSendStream(uint32 ssrc) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000496 return ResetStream(ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000497}
498
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000499bool SctpDataMediaChannel::AddRecvStream(const StreamParams& stream) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000500 return AddStream(stream);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000501}
502
503bool SctpDataMediaChannel::RemoveRecvStream(uint32 ssrc) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000504 return ResetStream(ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000505}
506
507bool SctpDataMediaChannel::SendData(
508 const SendDataParams& params,
509 const talk_base::Buffer& payload,
510 SendDataResult* result) {
511 if (result) {
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000512 // Preset |result| to assume an error. If SendData succeeds, we'll
513 // overwrite |*result| once more at the end.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000514 *result = SDR_ERROR;
515 }
516
517 if (!sending_) {
518 LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
519 << "Not sending packet with ssrc=" << params.ssrc
520 << " len=" << payload.length() << " before SetSend(true).";
521 return false;
522 }
523
wu@webrtc.org91053e72013-08-10 07:18:04 +0000524 if (params.type != cricket::DMT_CONTROL &&
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000525 open_streams_.find(params.ssrc) == open_streams_.end()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000526 LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
527 << "Not sending data because ssrc is unknown: "
528 << params.ssrc;
529 return false;
530 }
531
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000532 //
533 // Send data using SCTP.
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000534 ssize_t send_res = 0; // result from usrsctp_sendv.
535 struct sctp_sendv_spa spa = {0};
536 spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID;
537 spa.sendv_sndinfo.snd_sid = params.ssrc;
538 spa.sendv_sndinfo.snd_ppid = talk_base::HostToNetwork32(
539 GetPpid(params.type));
540
541 // Ordered implies reliable.
542 if (!params.ordered) {
543 spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED;
544 if (params.max_rtx_count >= 0 || params.max_rtx_ms == 0) {
545 spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
546 spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX;
547 spa.sendv_prinfo.pr_value = params.max_rtx_count;
548 } else {
549 spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
550 spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL;
551 spa.sendv_prinfo.pr_value = params.max_rtx_ms;
552 }
553 }
554
555 // We don't fragment.
556 send_res = usrsctp_sendv(sock_, payload.data(),
557 static_cast<size_t>(payload.length()),
558 NULL, 0, &spa,
559 static_cast<socklen_t>(sizeof(spa)),
560 SCTP_SENDV_SPA, 0);
561 if (send_res < 0) {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000562 if (errno == EWOULDBLOCK) {
563 *result = SDR_BLOCK;
564 LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned";
565 } else {
566 LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_
567 << "->SendData(...): "
568 << " usrsctp_sendv: ";
569 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000570 return false;
571 }
572 if (result) {
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000573 // Only way out now is success.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000574 *result = SDR_SUCCESS;
575 }
576 return true;
577}
578
579// Called by network interface when a packet has been received.
wu@webrtc.orga9890802013-12-13 00:21:03 +0000580void SctpDataMediaChannel::OnPacketReceived(
581 talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000582 LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...): " << " length="
583 << packet->length() << ", sending: " << sending_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000584 // Only give receiving packets to usrsctp after if connected. This enables two
585 // peers to each make a connect call, but for them not to receive an INIT
586 // packet before they have called connect; least the last receiver of the INIT
587 // packet will have called connect, and a connection will be established.
588 if (sending_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000589 // Pass received packet to SCTP stack. Once processed by usrsctp, the data
590 // will be will be given to the global OnSctpInboundData, and then,
591 // marshalled by a Post and handled with OnMessage.
592 usrsctp_conninput(this, packet->data(), packet->length(), 0);
593 } else {
594 // TODO(ldixon): Consider caching the packet for very slightly better
595 // reliability.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000596 }
597}
598
599void SctpDataMediaChannel::OnInboundPacketFromSctpToChannel(
600 SctpInboundPacket* packet) {
601 LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
602 << "Received SCTP data:"
603 << " ssrc=" << packet->params.ssrc
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000604 << " notification: " << (packet->flags & MSG_NOTIFICATION)
mallinath@webrtc.org1112c302013-09-23 20:34:45 +0000605 << " length=" << packet->buffer.length();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000606 // Sending a packet with data == NULL (no data) is SCTPs "close the
607 // connection" message. This sets sock_ = NULL;
608 if (!packet->buffer.length() || !packet->buffer.data()) {
609 LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
610 "No data, closing.";
611 return;
612 }
613 if (packet->flags & MSG_NOTIFICATION) {
614 OnNotificationFromSctp(&packet->buffer);
615 } else {
616 OnDataFromSctpToChannel(packet->params, &packet->buffer);
617 }
618}
619
620void SctpDataMediaChannel::OnDataFromSctpToChannel(
621 const ReceiveDataParams& params, talk_base::Buffer* buffer) {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000622 if (open_streams_.find(params.ssrc) == open_streams_.end()) {
wu@webrtc.org91053e72013-08-10 07:18:04 +0000623 if (params.type == DMT_CONTROL) {
wu@webrtc.org1d1ffc92013-10-16 18:12:02 +0000624 std::string label;
625 webrtc::DataChannelInit config;
626 if (ParseDataChannelOpenMessage(*buffer, &label, &config)) {
627 config.id = params.ssrc;
628 // Do not send the OPEN message for this data channel.
629 config.negotiated = true;
630 SignalNewStreamReceived(label, config);
631
632 // Add the stream immediately.
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000633 StreamParams sparams = StreamParams::CreateLegacy(params.ssrc);
wu@webrtc.org1d1ffc92013-10-16 18:12:02 +0000634 AddSendStream(sparams);
635 AddRecvStream(sparams);
636 } else {
637 LOG(LS_ERROR) << debug_name_ << "->OnDataFromSctpToChannel(...): "
638 << "Received malformed control message";
639 }
wu@webrtc.org91053e72013-08-10 07:18:04 +0000640 } else {
641 LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): "
642 << "Received packet for unknown ssrc: " << params.ssrc;
643 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000644 return;
645 }
646
647 if (receiving_) {
648 LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): "
649 << "Posting with length: " << buffer->length();
650 SignalDataReceived(params, buffer->data(), buffer->length());
651 } else {
652 LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): "
653 << "Not receiving packet with sid=" << params.ssrc
654 << " len=" << buffer->length()
655 << " before SetReceive(true).";
656 }
657}
658
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000659bool SctpDataMediaChannel::AddStream(const StreamParams& stream) {
660 if (!stream.has_ssrcs()) {
661 return false;
662 }
663
664 const uint32 ssrc = stream.first_ssrc();
665 if (open_streams_.find(ssrc) != open_streams_.end()) {
666 // We usually get an AddSendStream and an AddRecvStream for each stream, so
667 // this is really unlikely to be a useful warning message.
668 LOG(LS_VERBOSE) << debug_name_ << "->Add(Send|Recv)Stream(...): "
669 << "Not adding data stream '" << stream.id
670 << "' with ssrc=" << ssrc
671 << " because stream is already open.";
672 return false;
673 } else if (queued_reset_streams_.find(ssrc) != queued_reset_streams_.end()
674 || sent_reset_streams_.find(ssrc) != sent_reset_streams_.end()) {
675 LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
676 << "Not adding data stream '" << stream.id
677 << "' with ssrc=" << ssrc
678 << " because stream is still closing.";
679 return false;
680 }
681
682 open_streams_.insert(ssrc);
683 return true;
684}
685
686bool SctpDataMediaChannel::ResetStream(uint32 ssrc) {
687 // We typically get this called twice for the same stream, once each for
688 // Send and Recv.
689 StreamSet::iterator found = open_streams_.find(ssrc);
690
691 if (found == open_streams_.end()) {
692 LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): "
693 << "stream not found.";
694 return false;
695 } else {
696 LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): "
697 << "Removing and queuing RE-CONFIG chunk.";
698 open_streams_.erase(found);
699 }
700
701 // SCTP won't let you have more than one stream reset pending at a time, but
702 // you can close multiple streams in a single reset. So, we keep an internal
703 // queue of streams-to-reset, and send them as one reset message in
704 // SendQueuedStreamResets().
705 queued_reset_streams_.insert(ssrc);
706
707 // Signal our stream-reset logic that it should try to send now, if it can.
708 SendQueuedStreamResets();
709
710 // The stream will actually get removed when we get the acknowledgment.
711 return true;
712}
713
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000714void SctpDataMediaChannel::OnNotificationFromSctp(talk_base::Buffer* buffer) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000715 const sctp_notification& notification =
716 reinterpret_cast<const sctp_notification&>(*buffer->data());
717 ASSERT(notification.sn_header.sn_length == buffer->length());
718
719 // TODO(ldixon): handle notifications appropriately.
720 switch (notification.sn_header.sn_type) {
721 case SCTP_ASSOC_CHANGE:
722 LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE";
723 OnNotificationAssocChange(notification.sn_assoc_change);
724 break;
725 case SCTP_REMOTE_ERROR:
726 LOG(LS_INFO) << "SCTP_REMOTE_ERROR";
727 break;
728 case SCTP_SHUTDOWN_EVENT:
729 LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT";
730 break;
731 case SCTP_ADAPTATION_INDICATION:
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000732 LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION";
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000733 break;
734 case SCTP_PARTIAL_DELIVERY_EVENT:
735 LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT";
736 break;
737 case SCTP_AUTHENTICATION_EVENT:
738 LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT";
739 break;
740 case SCTP_SENDER_DRY_EVENT:
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000741 LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT";
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000742 SignalReadyToSend(true);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000743 break;
744 // TODO(ldixon): Unblock after congestion.
745 case SCTP_NOTIFICATIONS_STOPPED_EVENT:
746 LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT";
747 break;
748 case SCTP_SEND_FAILED_EVENT:
749 LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT";
750 break;
751 case SCTP_STREAM_RESET_EVENT:
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000752 OnStreamResetEvent(&notification.sn_strreset_event);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000753 break;
754 case SCTP_ASSOC_RESET_EVENT:
755 LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT";
756 break;
757 case SCTP_STREAM_CHANGE_EVENT:
758 LOG(LS_INFO) << "SCTP_STREAM_CHANGE_EVENT";
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000759 // An acknowledgment we get after our stream resets have gone through,
760 // if they've failed. We log the message, but don't react -- we don't
761 // keep around the last-transmitted set of SSIDs we wanted to close for
762 // error recovery. It doesn't seem likely to occur, and if so, likely
763 // harmless within the lifetime of a single SCTP association.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000764 break;
765 default:
766 LOG(LS_WARNING) << "Unknown SCTP event: "
767 << notification.sn_header.sn_type;
768 break;
769 }
770}
771
772void SctpDataMediaChannel::OnNotificationAssocChange(
773 const sctp_assoc_change& change) {
774 switch (change.sac_state) {
775 case SCTP_COMM_UP:
776 LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP";
777 break;
778 case SCTP_COMM_LOST:
779 LOG(LS_INFO) << "Association change SCTP_COMM_LOST";
780 break;
781 case SCTP_RESTART:
782 LOG(LS_INFO) << "Association change SCTP_RESTART";
783 break;
784 case SCTP_SHUTDOWN_COMP:
785 LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP";
786 break;
787 case SCTP_CANT_STR_ASSOC:
788 LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC";
789 break;
790 default:
791 LOG(LS_INFO) << "Association change UNKNOWN";
792 break;
793 }
794}
795
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000796void SctpDataMediaChannel::OnStreamResetEvent(
797 const struct sctp_stream_reset_event* evt) {
798 // A stream reset always involves two RE-CONFIG chunks for us -- we always
799 // simultaneously reset a sid's sequence number in both directions. The
800 // requesting side transmits a RE-CONFIG chunk and waits for the peer to send
801 // one back. Both sides get this SCTP_STREAM_RESET_EVENT when they receive
802 // RE-CONFIGs.
803 const int num_ssrcs = (evt->strreset_length - sizeof(*evt)) /
804 sizeof(evt->strreset_stream_list[0]);
805 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
806 << "): Flags = 0x"
807 << std::hex << evt->strreset_flags << " ("
808 << ListFlags(evt->strreset_flags) << ")";
809 LOG(LS_VERBOSE) << "Assoc = " << evt->strreset_assoc_id << ", Streams = ["
810 << ListArray(evt->strreset_stream_list, num_ssrcs)
811 << "], Open: ["
812 << ListStreams(open_streams_) << "], Q'd: ["
813 << ListStreams(queued_reset_streams_) << "], Sent: ["
814 << ListStreams(sent_reset_streams_) << "]";
815 bool local_stream_reset_acknowledged = false;
816
817 // If both sides try to reset some streams at the same time (even if they're
818 // disjoint sets), we can get reset failures.
819 if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) {
820 // OK, just try again. The stream IDs sent over when the RESET_FAILED flag
821 // is set seem to be garbage values. Ignore them.
822 queued_reset_streams_.insert(
823 sent_reset_streams_.begin(),
824 sent_reset_streams_.end());
825 sent_reset_streams_.clear();
826 local_stream_reset_acknowledged = true;
827
828 } else if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) {
829 // Each side gets an event for each direction of a stream. That is,
830 // closing sid k will make each side receive INCOMING and OUTGOING reset
831 // events for k. As per RFC6525, Section 5, paragraph 2, each side will
832 // get an INCOMING event first.
833 for (int i = 0; i < num_ssrcs; i++) {
834 const int stream_id = evt->strreset_stream_list[i];
835
836 // See if this stream ID was closed by our peer or ourselves.
837 StreamSet::iterator it = sent_reset_streams_.find(stream_id);
838
839 // The reset was requested locally.
840 if (it != sent_reset_streams_.end()) {
841 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
842 << "): local sid " << stream_id << " acknowledged.";
843 local_stream_reset_acknowledged = true;
844 sent_reset_streams_.erase(it);
845
846 } else if ((it = open_streams_.find(stream_id))
847 != open_streams_.end()) {
848 // The peer requested the reset.
849 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
850 << "): closing sid " << stream_id;
851 open_streams_.erase(it);
852 SignalStreamClosed(stream_id);
853
854 } else if ((it = queued_reset_streams_.find(stream_id))
855 != queued_reset_streams_.end()) {
856 // The peer requested the reset, but there was a local reset
857 // queued.
858 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
859 << "): double-sided close for sid " << stream_id;
860 // Both sides want the stream closed, and the peer got to send the
861 // RE-CONFIG first. Treat it like the local Remove(Send|Recv)Stream
862 // finished quickly.
863 queued_reset_streams_.erase(it);
864
865 } else {
866 // This stream is unknown. Sometimes this can be from an
867 // RESET_FAILED-related retransmit.
868 LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
869 << "): Unknown sid " << stream_id;
870 }
871 }
872 }
873
874 if (local_stream_reset_acknowledged) {
875 // This message acknowledges the last stream-reset request we sent out
876 // (only one can be outstanding at a time). Send out the next one.
877 SendQueuedStreamResets();
878 }
879}
880
wu@webrtc.org78187522013-10-07 23:32:02 +0000881// Puts the specified |param| from the codec identified by |id| into |dest|
882// and returns true. Or returns false if it wasn't there, leaving |dest|
883// untouched.
884static bool GetCodecIntParameter(const std::vector<DataCodec>& codecs,
885 int id, const std::string& name,
886 const std::string& param, int* dest) {
887 std::string value;
888 Codec match_pattern;
889 match_pattern.id = id;
890 match_pattern.name = name;
891 for (size_t i = 0; i < codecs.size(); ++i) {
892 if (codecs[i].Matches(match_pattern)) {
893 if (codecs[i].GetParam(param, &value)) {
894 *dest = talk_base::FromString<int>(value);
895 return true;
896 }
897 }
898 }
899 return false;
900}
901
902bool SctpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
903 return GetCodecIntParameter(
904 codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort,
905 &remote_port_);
906}
907
908bool SctpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
909 return GetCodecIntParameter(
910 codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort,
911 &local_port_);
912}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000913
914void SctpDataMediaChannel::OnPacketFromSctpToNetwork(
915 talk_base::Buffer* buffer) {
916 if (buffer->length() > kSctpMtu) {
917 LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): "
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000918 << "SCTP seems to have made a packet that is bigger "
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000919 "than its official MTU.";
920 }
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000921 MediaChannel::SendPacket(buffer);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000922}
923
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000924bool SctpDataMediaChannel::SendQueuedStreamResets() {
925 if (!sent_reset_streams_.empty() || queued_reset_streams_.empty())
926 return true;
927
928 LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ << "]: Sending ["
929 << ListStreams(queued_reset_streams_) << "], Open: ["
930 << ListStreams(open_streams_) << "], Sent: ["
931 << ListStreams(sent_reset_streams_) << "]";
932
933 const size_t num_streams = queued_reset_streams_.size();
934 const size_t num_bytes = sizeof(struct sctp_reset_streams)
935 + (num_streams * sizeof(uint16));
936
937 std::vector<uint8> reset_stream_buf(num_bytes, 0);
938 struct sctp_reset_streams* resetp = reinterpret_cast<sctp_reset_streams*>(
939 &reset_stream_buf[0]);
940 resetp->srs_assoc_id = SCTP_ALL_ASSOC;
941 resetp->srs_flags = SCTP_STREAM_RESET_INCOMING | SCTP_STREAM_RESET_OUTGOING;
942 resetp->srs_number_streams = num_streams;
943 int result_idx = 0;
944 for (StreamSet::iterator it = queued_reset_streams_.begin();
945 it != queued_reset_streams_.end(); ++it) {
946 resetp->srs_stream_list[result_idx++] = *it;
947 }
948
949 int ret = usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp,
950 reset_stream_buf.size());
951 if (ret < 0) {
952 LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to send a stream reset for "
953 << num_streams << " streams";
954 return false;
955 }
956
957 // sent_reset_streams_ is empty, and all the queued_reset_streams_ go into
958 // it now.
959 queued_reset_streams_.swap(sent_reset_streams_);
960 return true;
961}
962
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000963void SctpDataMediaChannel::OnMessage(talk_base::Message* msg) {
964 switch (msg->message_id) {
965 case MSG_SCTPINBOUNDPACKET: {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000966 talk_base::scoped_ptr<InboundPacketMessage> pdata(
967 static_cast<InboundPacketMessage*>(msg->pdata));
968 OnInboundPacketFromSctpToChannel(pdata->data().get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000969 break;
970 }
971 case MSG_SCTPOUTBOUNDPACKET: {
wu@webrtc.orgf6d6ed02014-01-03 22:08:47 +0000972 talk_base::scoped_ptr<OutboundPacketMessage> pdata(
973 static_cast<OutboundPacketMessage*>(msg->pdata));
974 OnPacketFromSctpToNetwork(pdata->data().get());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000975 break;
976 }
977 }
978}
979
980} // namespace cricket