| Index: webrtc/media/sctp/sctptransport.cc
|
| diff --git a/webrtc/media/sctp/sctptransport.cc b/webrtc/media/sctp/sctptransport.cc
|
| deleted file mode 100644
|
| index b95cf8a4baf444bc3f14aee79e47e9a9e2983d0a..0000000000000000000000000000000000000000
|
| --- a/webrtc/media/sctp/sctptransport.cc
|
| +++ /dev/null
|
| @@ -1,1090 +0,0 @@
|
| -/*
|
| - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
|
| - *
|
| - * Use of this source code is governed by a BSD-style license
|
| - * that can be found in the LICENSE file in the root of the source
|
| - * tree. An additional intellectual property rights grant can be found
|
| - * in the file PATENTS. All contributing project authors may
|
| - * be found in the AUTHORS file in the root of the source tree.
|
| - */
|
| -
|
| -#include <errno.h>
|
| -namespace {
|
| -// Some ERRNO values get re-#defined to WSA* equivalents in some talk/
|
| -// headers. We save the original ones in an enum.
|
| -enum PreservedErrno {
|
| - SCTP_EINPROGRESS = EINPROGRESS,
|
| - SCTP_EWOULDBLOCK = EWOULDBLOCK
|
| -};
|
| -}
|
| -
|
| -#include "webrtc/media/sctp/sctptransport.h"
|
| -
|
| -#include <stdarg.h>
|
| -#include <stdio.h>
|
| -
|
| -#include <memory>
|
| -#include <sstream>
|
| -
|
| -#include "usrsctplib/usrsctp.h"
|
| -#include "webrtc/base/arraysize.h"
|
| -#include "webrtc/base/copyonwritebuffer.h"
|
| -#include "webrtc/base/criticalsection.h"
|
| -#include "webrtc/base/helpers.h"
|
| -#include "webrtc/base/logging.h"
|
| -#include "webrtc/base/safe_conversions.h"
|
| -#include "webrtc/base/thread_checker.h"
|
| -#include "webrtc/base/trace_event.h"
|
| -#include "webrtc/media/base/codec.h"
|
| -#include "webrtc/media/base/mediaconstants.h"
|
| -#include "webrtc/media/base/rtputils.h" // For IsRtpPacket
|
| -#include "webrtc/media/base/streamparams.h"
|
| -
|
| -namespace {
|
| -
|
| -// The biggest SCTP packet. Starting from a 'safe' wire MTU value of 1280,
|
| -// take off 80 bytes for DTLS/TURN/TCP/IP overhead.
|
| -static constexpr size_t kSctpMtu = 1200;
|
| -
|
| -// The size of the SCTP association send buffer. 256kB, the usrsctp default.
|
| -static constexpr int kSendBufferSize = 262144;
|
| -
|
| -// Set the initial value of the static SCTP Data Engines reference count.
|
| -int g_usrsctp_usage_count = 0;
|
| -rtc::GlobalLockPod g_usrsctp_lock_;
|
| -
|
| -// DataMessageType is used for the SCTP "Payload Protocol Identifier", as
|
| -// defined in http://tools.ietf.org/html/rfc4960#section-14.4
|
| -//
|
| -// For the list of IANA approved values see:
|
| -// http://www.iana.org/assignments/sctp-parameters/sctp-parameters.xml
|
| -// The value is not used by SCTP itself. It indicates the protocol running
|
| -// on top of SCTP.
|
| -enum PayloadProtocolIdentifier {
|
| - PPID_NONE = 0, // No protocol is specified.
|
| - // Matches the PPIDs in mozilla source and
|
| - // https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol Sec. 9
|
| - // They're not yet assigned by IANA.
|
| - PPID_CONTROL = 50,
|
| - PPID_BINARY_PARTIAL = 52,
|
| - PPID_BINARY_LAST = 53,
|
| - PPID_TEXT_PARTIAL = 54,
|
| - PPID_TEXT_LAST = 51
|
| -};
|
| -
|
| -typedef std::set<uint32_t> StreamSet;
|
| -
|
| -// Returns a comma-separated, human-readable list of the stream IDs in 's'
|
| -std::string ListStreams(const StreamSet& s) {
|
| - std::stringstream result;
|
| - bool first = true;
|
| - for (StreamSet::const_iterator it = s.begin(); it != s.end(); ++it) {
|
| - if (!first) {
|
| - result << ", " << *it;
|
| - } else {
|
| - result << *it;
|
| - first = false;
|
| - }
|
| - }
|
| - return result.str();
|
| -}
|
| -
|
| -// Returns a pipe-separated, human-readable list of the SCTP_STREAM_RESET
|
| -// flags in 'flags'
|
| -std::string ListFlags(int flags) {
|
| - std::stringstream result;
|
| - bool first = true;
|
| -// Skip past the first 12 chars (strlen("SCTP_STREAM_"))
|
| -#define MAKEFLAG(X) \
|
| - { X, #X + 12 }
|
| - struct flaginfo_t {
|
| - int value;
|
| - const char* name;
|
| - } flaginfo[] = {MAKEFLAG(SCTP_STREAM_RESET_INCOMING_SSN),
|
| - MAKEFLAG(SCTP_STREAM_RESET_OUTGOING_SSN),
|
| - MAKEFLAG(SCTP_STREAM_RESET_DENIED),
|
| - MAKEFLAG(SCTP_STREAM_RESET_FAILED),
|
| - MAKEFLAG(SCTP_STREAM_CHANGE_DENIED)};
|
| -#undef MAKEFLAG
|
| - for (uint32_t i = 0; i < arraysize(flaginfo); ++i) {
|
| - if (flags & flaginfo[i].value) {
|
| - if (!first)
|
| - result << " | ";
|
| - result << flaginfo[i].name;
|
| - first = false;
|
| - }
|
| - }
|
| - return result.str();
|
| -}
|
| -
|
| -// Returns a comma-separated, human-readable list of the integers in 'array'.
|
| -// All 'num_elems' of them.
|
| -std::string ListArray(const uint16_t* array, int num_elems) {
|
| - std::stringstream result;
|
| - for (int i = 0; i < num_elems; ++i) {
|
| - if (i) {
|
| - result << ", " << array[i];
|
| - } else {
|
| - result << array[i];
|
| - }
|
| - }
|
| - return result.str();
|
| -}
|
| -
|
| -// Helper for logging SCTP messages.
|
| -void DebugSctpPrintf(const char* format, ...) {
|
| -#if RTC_DCHECK_IS_ON
|
| - char s[255];
|
| - va_list ap;
|
| - va_start(ap, format);
|
| - vsnprintf(s, sizeof(s), format, ap);
|
| - LOG(LS_INFO) << "SCTP: " << s;
|
| - va_end(ap);
|
| -#endif
|
| -}
|
| -
|
| -// Get the PPID to use for the terminating fragment of this type.
|
| -PayloadProtocolIdentifier GetPpid(cricket::DataMessageType type) {
|
| - switch (type) {
|
| - default:
|
| - case cricket::DMT_NONE:
|
| - return PPID_NONE;
|
| - case cricket::DMT_CONTROL:
|
| - return PPID_CONTROL;
|
| - case cricket::DMT_BINARY:
|
| - return PPID_BINARY_LAST;
|
| - case cricket::DMT_TEXT:
|
| - return PPID_TEXT_LAST;
|
| - }
|
| -}
|
| -
|
| -bool GetDataMediaType(PayloadProtocolIdentifier ppid,
|
| - cricket::DataMessageType* dest) {
|
| - RTC_DCHECK(dest != NULL);
|
| - switch (ppid) {
|
| - case PPID_BINARY_PARTIAL:
|
| - case PPID_BINARY_LAST:
|
| - *dest = cricket::DMT_BINARY;
|
| - return true;
|
| -
|
| - case PPID_TEXT_PARTIAL:
|
| - case PPID_TEXT_LAST:
|
| - *dest = cricket::DMT_TEXT;
|
| - return true;
|
| -
|
| - case PPID_CONTROL:
|
| - *dest = cricket::DMT_CONTROL;
|
| - return true;
|
| -
|
| - case PPID_NONE:
|
| - *dest = cricket::DMT_NONE;
|
| - return true;
|
| -
|
| - default:
|
| - return false;
|
| - }
|
| -}
|
| -
|
| -// Log the packet in text2pcap format, if log level is at LS_VERBOSE.
|
| -void VerboseLogPacket(const void* data, size_t length, int direction) {
|
| - if (LOG_CHECK_LEVEL(LS_VERBOSE) && length > 0) {
|
| - char* dump_buf;
|
| - // Some downstream project uses an older version of usrsctp that expects
|
| - // a non-const "void*" as first parameter when dumping the packet, so we
|
| - // need to cast the const away here to avoid a compiler error.
|
| - if ((dump_buf = usrsctp_dumppacket(const_cast<void*>(data), length,
|
| - direction)) != NULL) {
|
| - LOG(LS_VERBOSE) << dump_buf;
|
| - usrsctp_freedumpbuffer(dump_buf);
|
| - }
|
| - }
|
| -}
|
| -
|
| -} // namespace
|
| -
|
| -namespace cricket {
|
| -
|
| -// Handles global init/deinit, and mapping from usrsctp callbacks to
|
| -// SctpTransport calls.
|
| -class SctpTransport::UsrSctpWrapper {
|
| - public:
|
| - static void InitializeUsrSctp() {
|
| - LOG(LS_INFO) << __FUNCTION__;
|
| - // First argument is udp_encapsulation_port, which is not releveant for our
|
| - // AF_CONN use of sctp.
|
| - usrsctp_init(0, &UsrSctpWrapper::OnSctpOutboundPacket, &DebugSctpPrintf);
|
| -
|
| - // To turn on/off detailed SCTP debugging. You will also need to have the
|
| - // SCTP_DEBUG cpp defines flag.
|
| - // usrsctp_sysctl_set_sctp_debug_on(SCTP_DEBUG_ALL);
|
| -
|
| - // TODO(ldixon): Consider turning this on/off.
|
| - usrsctp_sysctl_set_sctp_ecn_enable(0);
|
| -
|
| - // This is harmless, but we should find out when the library default
|
| - // changes.
|
| - int send_size = usrsctp_sysctl_get_sctp_sendspace();
|
| - if (send_size != kSendBufferSize) {
|
| - LOG(LS_ERROR) << "Got different send size than expected: " << send_size;
|
| - }
|
| -
|
| - // TODO(ldixon): Consider turning this on/off.
|
| - // This is not needed right now (we don't do dynamic address changes):
|
| - // If SCTP Auto-ASCONF is enabled, the peer is informed automatically
|
| - // when a new address is added or removed. This feature is enabled by
|
| - // default.
|
| - // usrsctp_sysctl_set_sctp_auto_asconf(0);
|
| -
|
| - // TODO(ldixon): Consider turning this on/off.
|
| - // Add a blackhole sysctl. Setting it to 1 results in no ABORTs
|
| - // being sent in response to INITs, setting it to 2 results
|
| - // in no ABORTs being sent for received OOTB packets.
|
| - // This is similar to the TCP sysctl.
|
| - //
|
| - // See: http://lakerest.net/pipermail/sctp-coders/2012-January/009438.html
|
| - // See: http://svnweb.freebsd.org/base?view=revision&revision=229805
|
| - // usrsctp_sysctl_set_sctp_blackhole(2);
|
| -
|
| - // Set the number of default outgoing streams. This is the number we'll
|
| - // send in the SCTP INIT message.
|
| - usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(kMaxSctpStreams);
|
| - }
|
| -
|
| - static void UninitializeUsrSctp() {
|
| - LOG(LS_INFO) << __FUNCTION__;
|
| - // usrsctp_finish() may fail if it's called too soon after the transports
|
| - // are
|
| - // closed. Wait and try again until it succeeds for up to 3 seconds.
|
| - for (size_t i = 0; i < 300; ++i) {
|
| - if (usrsctp_finish() == 0) {
|
| - return;
|
| - }
|
| -
|
| - rtc::Thread::SleepMs(10);
|
| - }
|
| - LOG(LS_ERROR) << "Failed to shutdown usrsctp.";
|
| - }
|
| -
|
| - static void IncrementUsrSctpUsageCount() {
|
| - rtc::GlobalLockScope lock(&g_usrsctp_lock_);
|
| - if (!g_usrsctp_usage_count) {
|
| - InitializeUsrSctp();
|
| - }
|
| - ++g_usrsctp_usage_count;
|
| - }
|
| -
|
| - static void DecrementUsrSctpUsageCount() {
|
| - rtc::GlobalLockScope lock(&g_usrsctp_lock_);
|
| - --g_usrsctp_usage_count;
|
| - if (!g_usrsctp_usage_count) {
|
| - UninitializeUsrSctp();
|
| - }
|
| - }
|
| -
|
| - // This is the callback usrsctp uses when there's data to send on the network
|
| - // that has been wrapped appropriatly for the SCTP protocol.
|
| - static int OnSctpOutboundPacket(void* addr,
|
| - void* data,
|
| - size_t length,
|
| - uint8_t tos,
|
| - uint8_t set_df) {
|
| - SctpTransport* transport = static_cast<SctpTransport*>(addr);
|
| - LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
|
| - << "addr: " << addr << "; length: " << length
|
| - << "; tos: " << std::hex << static_cast<int>(tos)
|
| - << "; set_df: " << std::hex << static_cast<int>(set_df);
|
| -
|
| - VerboseLogPacket(data, length, SCTP_DUMP_OUTBOUND);
|
| - // Note: We have to copy the data; the caller will delete it.
|
| - rtc::CopyOnWriteBuffer buf(reinterpret_cast<uint8_t*>(data), length);
|
| - // TODO(deadbeef): Why do we need an AsyncInvoke here? We're already on the
|
| - // right thread and don't need to unwind the stack.
|
| - transport->invoker_.AsyncInvoke<void>(
|
| - RTC_FROM_HERE, transport->network_thread_,
|
| - rtc::Bind(&SctpTransport::OnPacketFromSctpToNetwork, transport, buf));
|
| - return 0;
|
| - }
|
| -
|
| - // This is the callback called from usrsctp when data has been received, after
|
| - // a packet has been interpreted and parsed by usrsctp and found to contain
|
| - // payload data. It is called by a usrsctp thread. It is assumed this function
|
| - // will free the memory used by 'data'.
|
| - static int OnSctpInboundPacket(struct socket* sock,
|
| - union sctp_sockstore addr,
|
| - void* data,
|
| - size_t length,
|
| - struct sctp_rcvinfo rcv,
|
| - int flags,
|
| - void* ulp_info) {
|
| - SctpTransport* transport = static_cast<SctpTransport*>(ulp_info);
|
| - // Post data to the transport's receiver thread (copying it).
|
| - // TODO(ldixon): Unclear if copy is needed as this method is responsible for
|
| - // memory cleanup. But this does simplify code.
|
| - const PayloadProtocolIdentifier ppid =
|
| - static_cast<PayloadProtocolIdentifier>(
|
| - rtc::HostToNetwork32(rcv.rcv_ppid));
|
| - DataMessageType type = DMT_NONE;
|
| - if (!GetDataMediaType(ppid, &type) && !(flags & MSG_NOTIFICATION)) {
|
| - // It's neither a notification nor a recognized data packet. Drop it.
|
| - LOG(LS_ERROR) << "Received an unknown PPID " << ppid
|
| - << " on an SCTP packet. Dropping.";
|
| - } else {
|
| - rtc::CopyOnWriteBuffer buffer;
|
| - ReceiveDataParams params;
|
| - buffer.SetData(reinterpret_cast<uint8_t*>(data), length);
|
| - params.sid = rcv.rcv_sid;
|
| - params.seq_num = rcv.rcv_ssn;
|
| - params.timestamp = rcv.rcv_tsn;
|
| - params.type = type;
|
| - // The ownership of the packet transfers to |invoker_|. Using
|
| - // CopyOnWriteBuffer is the most convenient way to do this.
|
| - transport->invoker_.AsyncInvoke<void>(
|
| - RTC_FROM_HERE, transport->network_thread_,
|
| - rtc::Bind(&SctpTransport::OnInboundPacketFromSctpToChannel, transport,
|
| - buffer, params, flags));
|
| - }
|
| - free(data);
|
| - return 1;
|
| - }
|
| -
|
| - static SctpTransport* GetTransportFromSocket(struct socket* sock) {
|
| - struct sockaddr* addrs = nullptr;
|
| - int naddrs = usrsctp_getladdrs(sock, 0, &addrs);
|
| - if (naddrs <= 0 || addrs[0].sa_family != AF_CONN) {
|
| - return nullptr;
|
| - }
|
| - // usrsctp_getladdrs() returns the addresses bound to this socket, which
|
| - // contains the SctpTransport* as sconn_addr. Read the pointer,
|
| - // then free the list of addresses once we have the pointer. We only open
|
| - // AF_CONN sockets, and they should all have the sconn_addr set to the
|
| - // pointer that created them, so [0] is as good as any other.
|
| - struct sockaddr_conn* sconn =
|
| - reinterpret_cast<struct sockaddr_conn*>(&addrs[0]);
|
| - SctpTransport* transport =
|
| - reinterpret_cast<SctpTransport*>(sconn->sconn_addr);
|
| - usrsctp_freeladdrs(addrs);
|
| -
|
| - return transport;
|
| - }
|
| -
|
| - static int SendThresholdCallback(struct socket* sock, uint32_t sb_free) {
|
| - // Fired on our I/O thread. SctpTransport::OnPacketReceived() gets
|
| - // a packet containing acknowledgments, which goes into usrsctp_conninput,
|
| - // and then back here.
|
| - SctpTransport* transport = GetTransportFromSocket(sock);
|
| - if (!transport) {
|
| - LOG(LS_ERROR)
|
| - << "SendThresholdCallback: Failed to get transport for socket "
|
| - << sock;
|
| - return 0;
|
| - }
|
| - transport->OnSendThresholdCallback();
|
| - return 0;
|
| - }
|
| -};
|
| -
|
| -SctpTransport::SctpTransport(rtc::Thread* network_thread,
|
| - TransportChannel* channel)
|
| - : network_thread_(network_thread),
|
| - transport_channel_(channel),
|
| - was_ever_writable_(channel->writable()) {
|
| - RTC_DCHECK(network_thread_);
|
| - RTC_DCHECK(transport_channel_);
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - ConnectTransportChannelSignals();
|
| -}
|
| -
|
| -SctpTransport::~SctpTransport() {
|
| - // Close abruptly; no reset procedure.
|
| - CloseSctpSocket();
|
| -}
|
| -
|
| -void SctpTransport::SetTransportChannel(cricket::TransportChannel* channel) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - RTC_DCHECK(channel);
|
| - DisconnectTransportChannelSignals();
|
| - transport_channel_ = channel;
|
| - ConnectTransportChannelSignals();
|
| - if (!was_ever_writable_ && channel->writable()) {
|
| - was_ever_writable_ = true;
|
| - // New channel is writable, now we can start the SCTP connection if Start
|
| - // was called already.
|
| - if (started_) {
|
| - RTC_DCHECK(!sock_);
|
| - Connect();
|
| - }
|
| - }
|
| -}
|
| -
|
| -bool SctpTransport::Start(int local_sctp_port, int remote_sctp_port) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (local_sctp_port == -1) {
|
| - local_sctp_port = kSctpDefaultPort;
|
| - }
|
| - if (remote_sctp_port == -1) {
|
| - remote_sctp_port = kSctpDefaultPort;
|
| - }
|
| - if (started_) {
|
| - if (local_sctp_port != local_port_ || remote_sctp_port != remote_port_) {
|
| - LOG(LS_ERROR) << "Can't change SCTP port after SCTP association formed.";
|
| - return false;
|
| - }
|
| - return true;
|
| - }
|
| - local_port_ = local_sctp_port;
|
| - remote_port_ = remote_sctp_port;
|
| - started_ = true;
|
| - RTC_DCHECK(!sock_);
|
| - // Only try to connect if the DTLS channel has been writable before
|
| - // (indicating that the DTLS handshake is complete).
|
| - if (was_ever_writable_) {
|
| - return Connect();
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool SctpTransport::OpenStream(int sid) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (sid > kMaxSctpSid) {
|
| - LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
|
| - << "Not adding data stream "
|
| - << "with sid=" << sid << " because sid is too high.";
|
| - return false;
|
| - } else if (open_streams_.find(sid) != open_streams_.end()) {
|
| - LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
|
| - << "Not adding data stream "
|
| - << "with sid=" << sid << " because stream is already open.";
|
| - return false;
|
| - } else if (queued_reset_streams_.find(sid) != queued_reset_streams_.end() ||
|
| - sent_reset_streams_.find(sid) != sent_reset_streams_.end()) {
|
| - LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
|
| - << "Not adding data stream "
|
| - << " with sid=" << sid
|
| - << " because stream is still closing.";
|
| - return false;
|
| - }
|
| -
|
| - open_streams_.insert(sid);
|
| - return true;
|
| -}
|
| -
|
| -bool SctpTransport::ResetStream(int sid) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - StreamSet::iterator found = open_streams_.find(sid);
|
| - if (found == open_streams_.end()) {
|
| - LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid << "): "
|
| - << "stream not found.";
|
| - return false;
|
| - } else {
|
| - LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid << "): "
|
| - << "Removing and queuing RE-CONFIG chunk.";
|
| - open_streams_.erase(found);
|
| - }
|
| -
|
| - // SCTP won't let you have more than one stream reset pending at a time, but
|
| - // you can close multiple streams in a single reset. So, we keep an internal
|
| - // queue of streams-to-reset, and send them as one reset message in
|
| - // SendQueuedStreamResets().
|
| - queued_reset_streams_.insert(sid);
|
| -
|
| - // Signal our stream-reset logic that it should try to send now, if it can.
|
| - SendQueuedStreamResets();
|
| -
|
| - // The stream will actually get removed when we get the acknowledgment.
|
| - return true;
|
| -}
|
| -
|
| -bool SctpTransport::SendData(const SendDataParams& params,
|
| - const rtc::CopyOnWriteBuffer& payload,
|
| - SendDataResult* result) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (result) {
|
| - // Preset |result| to assume an error. If SendData succeeds, we'll
|
| - // overwrite |*result| once more at the end.
|
| - *result = SDR_ERROR;
|
| - }
|
| -
|
| - if (!sock_) {
|
| - LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
|
| - << "Not sending packet with sid=" << params.sid
|
| - << " len=" << payload.size() << " before Start().";
|
| - return false;
|
| - }
|
| -
|
| - if (params.type != DMT_CONTROL &&
|
| - open_streams_.find(params.sid) == open_streams_.end()) {
|
| - LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
|
| - << "Not sending data because sid is unknown: "
|
| - << params.sid;
|
| - return false;
|
| - }
|
| -
|
| - // Send data using SCTP.
|
| - ssize_t send_res = 0; // result from usrsctp_sendv.
|
| - struct sctp_sendv_spa spa = {0};
|
| - spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID;
|
| - spa.sendv_sndinfo.snd_sid = params.sid;
|
| - spa.sendv_sndinfo.snd_ppid = rtc::HostToNetwork32(GetPpid(params.type));
|
| -
|
| - // Ordered implies reliable.
|
| - if (!params.ordered) {
|
| - spa.sendv_sndinfo.snd_flags |= SCTP_UNORDERED;
|
| - if (params.max_rtx_count >= 0 || params.max_rtx_ms == 0) {
|
| - spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
|
| - spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_RTX;
|
| - spa.sendv_prinfo.pr_value = params.max_rtx_count;
|
| - } else {
|
| - spa.sendv_flags |= SCTP_SEND_PRINFO_VALID;
|
| - spa.sendv_prinfo.pr_policy = SCTP_PR_SCTP_TTL;
|
| - spa.sendv_prinfo.pr_value = params.max_rtx_ms;
|
| - }
|
| - }
|
| -
|
| - // We don't fragment.
|
| - send_res = usrsctp_sendv(
|
| - sock_, payload.data(), static_cast<size_t>(payload.size()), NULL, 0, &spa,
|
| - rtc::checked_cast<socklen_t>(sizeof(spa)), SCTP_SENDV_SPA, 0);
|
| - if (send_res < 0) {
|
| - if (errno == SCTP_EWOULDBLOCK) {
|
| - *result = SDR_BLOCK;
|
| - ready_to_send_data_ = false;
|
| - LOG(LS_INFO) << debug_name_ << "->SendData(...): EWOULDBLOCK returned";
|
| - } else {
|
| - LOG_ERRNO(LS_ERROR) << "ERROR:" << debug_name_ << "->SendData(...): "
|
| - << " usrsctp_sendv: ";
|
| - }
|
| - return false;
|
| - }
|
| - if (result) {
|
| - // Only way out now is success.
|
| - *result = SDR_SUCCESS;
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -bool SctpTransport::ReadyToSendData() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - return ready_to_send_data_;
|
| -}
|
| -
|
| -void SctpTransport::ConnectTransportChannelSignals() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - transport_channel_->SignalWritableState.connect(
|
| - this, &SctpTransport::OnWritableState);
|
| - transport_channel_->SignalReadPacket.connect(this,
|
| - &SctpTransport::OnPacketRead);
|
| -}
|
| -
|
| -void SctpTransport::DisconnectTransportChannelSignals() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - transport_channel_->SignalWritableState.disconnect(this);
|
| - transport_channel_->SignalReadPacket.disconnect(this);
|
| -}
|
| -
|
| -bool SctpTransport::Connect() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - LOG(LS_VERBOSE) << debug_name_ << "->Connect().";
|
| -
|
| - // If we already have a socket connection (which shouldn't ever happen), just
|
| - // return.
|
| - RTC_DCHECK(!sock_);
|
| - if (sock_) {
|
| - LOG(LS_ERROR) << debug_name_ << "->Connect(): Ignored as socket "
|
| - "is already established.";
|
| - return true;
|
| - }
|
| -
|
| - // If no socket (it was closed) try to start it again. This can happen when
|
| - // the socket we are connecting to closes, does an sctp shutdown handshake,
|
| - // or behaves unexpectedly causing us to perform a CloseSctpSocket.
|
| - if (!OpenSctpSocket()) {
|
| - return false;
|
| - }
|
| -
|
| - // Note: conversion from int to uint16_t happens on assignment.
|
| - sockaddr_conn local_sconn = GetSctpSockAddr(local_port_);
|
| - if (usrsctp_bind(sock_, reinterpret_cast<sockaddr*>(&local_sconn),
|
| - sizeof(local_sconn)) < 0) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_
|
| - << "->Connect(): " << ("Failed usrsctp_bind");
|
| - CloseSctpSocket();
|
| - return false;
|
| - }
|
| -
|
| - // Note: conversion from int to uint16_t happens on assignment.
|
| - sockaddr_conn remote_sconn = GetSctpSockAddr(remote_port_);
|
| - int connect_result = usrsctp_connect(
|
| - sock_, reinterpret_cast<sockaddr*>(&remote_sconn), sizeof(remote_sconn));
|
| - if (connect_result < 0 && errno != SCTP_EINPROGRESS) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
|
| - << "Failed usrsctp_connect. got errno=" << errno
|
| - << ", but wanted " << SCTP_EINPROGRESS;
|
| - CloseSctpSocket();
|
| - return false;
|
| - }
|
| - // Set the MTU and disable MTU discovery.
|
| - // We can only do this after usrsctp_connect or it has no effect.
|
| - sctp_paddrparams params = {{0}};
|
| - memcpy(¶ms.spp_address, &remote_sconn, sizeof(remote_sconn));
|
| - params.spp_flags = SPP_PMTUD_DISABLE;
|
| - params.spp_pathmtu = kSctpMtu;
|
| - if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, ¶ms,
|
| - sizeof(params))) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
|
| - << "Failed to set SCTP_PEER_ADDR_PARAMS.";
|
| - }
|
| - // Since this is a fresh SCTP association, we'll always start out with empty
|
| - // queues, so "ReadyToSendData" should be true.
|
| - SetReadyToSendData();
|
| - return true;
|
| -}
|
| -
|
| -bool SctpTransport::OpenSctpSocket() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (sock_) {
|
| - LOG(LS_WARNING) << debug_name_ << "->OpenSctpSocket(): "
|
| - << "Ignoring attempt to re-create existing socket.";
|
| - return false;
|
| - }
|
| -
|
| - UsrSctpWrapper::IncrementUsrSctpUsageCount();
|
| -
|
| - // If kSendBufferSize isn't reflective of reality, we log an error, but we
|
| - // still have to do something reasonable here. Look up what the buffer's
|
| - // real size is and set our threshold to something reasonable.
|
| - static const int kSendThreshold = usrsctp_sysctl_get_sctp_sendspace() / 2;
|
| -
|
| - sock_ = usrsctp_socket(
|
| - AF_CONN, SOCK_STREAM, IPPROTO_SCTP, &UsrSctpWrapper::OnSctpInboundPacket,
|
| - &UsrSctpWrapper::SendThresholdCallback, kSendThreshold, this);
|
| - if (!sock_) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->OpenSctpSocket(): "
|
| - << "Failed to create SCTP socket.";
|
| - UsrSctpWrapper::DecrementUsrSctpUsageCount();
|
| - return false;
|
| - }
|
| -
|
| - if (!ConfigureSctpSocket()) {
|
| - usrsctp_close(sock_);
|
| - sock_ = nullptr;
|
| - UsrSctpWrapper::DecrementUsrSctpUsageCount();
|
| - return false;
|
| - }
|
| - // Register this class as an address for usrsctp. This is used by SCTP to
|
| - // direct the packets received (by the created socket) to this class.
|
| - usrsctp_register_address(this);
|
| - return true;
|
| -}
|
| -
|
| -bool SctpTransport::ConfigureSctpSocket() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - RTC_DCHECK(sock_);
|
| - // Make the socket non-blocking. Connect, close, shutdown etc will not block
|
| - // the thread waiting for the socket operation to complete.
|
| - if (usrsctp_set_non_blocking(sock_, 1) < 0) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
|
| - << "Failed to set SCTP to non blocking.";
|
| - return false;
|
| - }
|
| -
|
| - // This ensures that the usrsctp close call deletes the association. This
|
| - // prevents usrsctp from calling OnSctpOutboundPacket with references to
|
| - // this class as the address.
|
| - linger linger_opt;
|
| - linger_opt.l_onoff = 1;
|
| - linger_opt.l_linger = 0;
|
| - if (usrsctp_setsockopt(sock_, SOL_SOCKET, SO_LINGER, &linger_opt,
|
| - sizeof(linger_opt))) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
|
| - << "Failed to set SO_LINGER.";
|
| - return false;
|
| - }
|
| -
|
| - // Enable stream ID resets.
|
| - struct sctp_assoc_value stream_rst;
|
| - stream_rst.assoc_id = SCTP_ALL_ASSOC;
|
| - stream_rst.assoc_value = 1;
|
| - if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_ENABLE_STREAM_RESET,
|
| - &stream_rst, sizeof(stream_rst))) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
|
| -
|
| - << "Failed to set SCTP_ENABLE_STREAM_RESET.";
|
| - return false;
|
| - }
|
| -
|
| - // Nagle.
|
| - uint32_t nodelay = 1;
|
| - if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_NODELAY, &nodelay,
|
| - sizeof(nodelay))) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
|
| - << "Failed to set SCTP_NODELAY.";
|
| - return false;
|
| - }
|
| -
|
| - // Subscribe to SCTP event notifications.
|
| - int event_types[] = {SCTP_ASSOC_CHANGE, SCTP_PEER_ADDR_CHANGE,
|
| - SCTP_SEND_FAILED_EVENT, SCTP_SENDER_DRY_EVENT,
|
| - SCTP_STREAM_RESET_EVENT};
|
| - struct sctp_event event = {0};
|
| - event.se_assoc_id = SCTP_ALL_ASSOC;
|
| - event.se_on = 1;
|
| - for (size_t i = 0; i < arraysize(event_types); i++) {
|
| - event.se_type = event_types[i];
|
| - if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EVENT, &event,
|
| - sizeof(event)) < 0) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
|
| -
|
| - << "Failed to set SCTP_EVENT type: " << event.se_type;
|
| - return false;
|
| - }
|
| - }
|
| - return true;
|
| -}
|
| -
|
| -void SctpTransport::CloseSctpSocket() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (sock_) {
|
| - // We assume that SO_LINGER option is set to close the association when
|
| - // close is called. This means that any pending packets in usrsctp will be
|
| - // discarded instead of being sent.
|
| - usrsctp_close(sock_);
|
| - sock_ = nullptr;
|
| - usrsctp_deregister_address(this);
|
| - UsrSctpWrapper::DecrementUsrSctpUsageCount();
|
| - ready_to_send_data_ = false;
|
| - }
|
| -}
|
| -
|
| -bool SctpTransport::SendQueuedStreamResets() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (!sent_reset_streams_.empty() || queued_reset_streams_.empty()) {
|
| - return true;
|
| - }
|
| -
|
| - LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_ << "]: Sending ["
|
| - << ListStreams(queued_reset_streams_) << "], Open: ["
|
| - << ListStreams(open_streams_) << "], Sent: ["
|
| - << ListStreams(sent_reset_streams_) << "]";
|
| -
|
| - const size_t num_streams = queued_reset_streams_.size();
|
| - const size_t num_bytes =
|
| - sizeof(struct sctp_reset_streams) + (num_streams * sizeof(uint16_t));
|
| -
|
| - std::vector<uint8_t> reset_stream_buf(num_bytes, 0);
|
| - struct sctp_reset_streams* resetp =
|
| - reinterpret_cast<sctp_reset_streams*>(&reset_stream_buf[0]);
|
| - resetp->srs_assoc_id = SCTP_ALL_ASSOC;
|
| - resetp->srs_flags = SCTP_STREAM_RESET_INCOMING | SCTP_STREAM_RESET_OUTGOING;
|
| - resetp->srs_number_streams = rtc::checked_cast<uint16_t>(num_streams);
|
| - int result_idx = 0;
|
| - for (StreamSet::iterator it = queued_reset_streams_.begin();
|
| - it != queued_reset_streams_.end(); ++it) {
|
| - resetp->srs_stream_list[result_idx++] = *it;
|
| - }
|
| -
|
| - int ret =
|
| - usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp,
|
| - rtc::checked_cast<socklen_t>(reset_stream_buf.size()));
|
| - if (ret < 0) {
|
| - LOG_ERRNO(LS_ERROR) << debug_name_ << "->SendQueuedStreamResets(): "
|
| - "Failed to send a stream reset for "
|
| - << num_streams << " streams";
|
| - return false;
|
| - }
|
| -
|
| - // sent_reset_streams_ is empty, and all the queued_reset_streams_ go into
|
| - // it now.
|
| - queued_reset_streams_.swap(sent_reset_streams_);
|
| - return true;
|
| -}
|
| -
|
| -void SctpTransport::SetReadyToSendData() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (!ready_to_send_data_) {
|
| - ready_to_send_data_ = true;
|
| - SignalReadyToSendData();
|
| - }
|
| -}
|
| -
|
| -void SctpTransport::OnWritableState(rtc::PacketTransportInterface* transport) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - RTC_DCHECK_EQ(transport_channel_, transport);
|
| - if (!was_ever_writable_ && transport->writable()) {
|
| - was_ever_writable_ = true;
|
| - if (started_) {
|
| - Connect();
|
| - }
|
| - }
|
| -}
|
| -
|
| -// Called by network interface when a packet has been received.
|
| -void SctpTransport::OnPacketRead(rtc::PacketTransportInterface* transport,
|
| - const char* data,
|
| - size_t len,
|
| - const rtc::PacketTime& packet_time,
|
| - int flags) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - RTC_DCHECK_EQ(transport_channel_, transport);
|
| - TRACE_EVENT0("webrtc", "SctpTransport::OnPacketRead");
|
| -
|
| - // TODO(pthatcher): Do this in a more robust way by checking for
|
| - // SCTP or DTLS.
|
| - if (IsRtpPacket(data, len)) {
|
| - return;
|
| - }
|
| -
|
| - LOG(LS_VERBOSE) << debug_name_ << "->OnPacketRead(...): "
|
| - << " length=" << len << ", started: " << started_;
|
| - // Only give receiving packets to usrsctp after if connected. This enables two
|
| - // peers to each make a connect call, but for them not to receive an INIT
|
| - // packet before they have called connect; least the last receiver of the INIT
|
| - // packet will have called connect, and a connection will be established.
|
| - if (sock_) {
|
| - // Pass received packet to SCTP stack. Once processed by usrsctp, the data
|
| - // will be will be given to the global OnSctpInboundData, and then,
|
| - // marshalled by the AsyncInvoker.
|
| - VerboseLogPacket(data, len, SCTP_DUMP_INBOUND);
|
| - usrsctp_conninput(this, data, len, 0);
|
| - } else {
|
| - // TODO(ldixon): Consider caching the packet for very slightly better
|
| - // reliability.
|
| - }
|
| -}
|
| -
|
| -void SctpTransport::OnSendThresholdCallback() {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - SetReadyToSendData();
|
| -}
|
| -
|
| -sockaddr_conn SctpTransport::GetSctpSockAddr(int port) {
|
| - sockaddr_conn sconn = {0};
|
| - sconn.sconn_family = AF_CONN;
|
| -#ifdef HAVE_SCONN_LEN
|
| - sconn.sconn_len = sizeof(sockaddr_conn);
|
| -#endif
|
| - // Note: conversion from int to uint16_t happens here.
|
| - sconn.sconn_port = rtc::HostToNetwork16(port);
|
| - sconn.sconn_addr = this;
|
| - return sconn;
|
| -}
|
| -
|
| -void SctpTransport::OnPacketFromSctpToNetwork(
|
| - const rtc::CopyOnWriteBuffer& buffer) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - if (buffer.size() > (kSctpMtu)) {
|
| - LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): "
|
| - << "SCTP seems to have made a packet that is bigger "
|
| - << "than its official MTU: " << buffer.size() << " vs max of "
|
| - << kSctpMtu;
|
| - }
|
| - TRACE_EVENT0("webrtc", "SctpTransport::OnPacketFromSctpToNetwork");
|
| -
|
| - // Don't create noise by trying to send a packet when the DTLS channel isn't
|
| - // even writable.
|
| - if (!transport_channel_->writable()) {
|
| - return;
|
| - }
|
| -
|
| - // Bon voyage.
|
| - transport_channel_->SendPacket(buffer.data<char>(), buffer.size(),
|
| - rtc::PacketOptions(), PF_NORMAL);
|
| -}
|
| -
|
| -void SctpTransport::OnInboundPacketFromSctpToChannel(
|
| - const rtc::CopyOnWriteBuffer& buffer,
|
| - ReceiveDataParams params,
|
| - int flags) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
|
| - << "Received SCTP data:"
|
| - << " sid=" << params.sid
|
| - << " notification: " << (flags & MSG_NOTIFICATION)
|
| - << " length=" << buffer.size();
|
| - // Sending a packet with data == NULL (no data) is SCTPs "close the
|
| - // connection" message. This sets sock_ = NULL;
|
| - if (!buffer.size() || !buffer.data()) {
|
| - LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
|
| - "No data, closing.";
|
| - return;
|
| - }
|
| - if (flags & MSG_NOTIFICATION) {
|
| - OnNotificationFromSctp(buffer);
|
| - } else {
|
| - OnDataFromSctpToChannel(params, buffer);
|
| - }
|
| -}
|
| -
|
| -void SctpTransport::OnDataFromSctpToChannel(
|
| - const ReceiveDataParams& params,
|
| - const rtc::CopyOnWriteBuffer& buffer) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): "
|
| - << "Posting with length: " << buffer.size() << " on stream "
|
| - << params.sid;
|
| - // Reports all received messages to upper layers, no matter whether the sid
|
| - // is known.
|
| - SignalDataReceived(params, buffer);
|
| -}
|
| -
|
| -void SctpTransport::OnNotificationFromSctp(
|
| - const rtc::CopyOnWriteBuffer& buffer) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - const sctp_notification& notification =
|
| - reinterpret_cast<const sctp_notification&>(*buffer.data());
|
| - RTC_DCHECK(notification.sn_header.sn_length == buffer.size());
|
| -
|
| - // TODO(ldixon): handle notifications appropriately.
|
| - switch (notification.sn_header.sn_type) {
|
| - case SCTP_ASSOC_CHANGE:
|
| - LOG(LS_VERBOSE) << "SCTP_ASSOC_CHANGE";
|
| - OnNotificationAssocChange(notification.sn_assoc_change);
|
| - break;
|
| - case SCTP_REMOTE_ERROR:
|
| - LOG(LS_INFO) << "SCTP_REMOTE_ERROR";
|
| - break;
|
| - case SCTP_SHUTDOWN_EVENT:
|
| - LOG(LS_INFO) << "SCTP_SHUTDOWN_EVENT";
|
| - break;
|
| - case SCTP_ADAPTATION_INDICATION:
|
| - LOG(LS_INFO) << "SCTP_ADAPTATION_INDICATION";
|
| - break;
|
| - case SCTP_PARTIAL_DELIVERY_EVENT:
|
| - LOG(LS_INFO) << "SCTP_PARTIAL_DELIVERY_EVENT";
|
| - break;
|
| - case SCTP_AUTHENTICATION_EVENT:
|
| - LOG(LS_INFO) << "SCTP_AUTHENTICATION_EVENT";
|
| - break;
|
| - case SCTP_SENDER_DRY_EVENT:
|
| - LOG(LS_VERBOSE) << "SCTP_SENDER_DRY_EVENT";
|
| - SetReadyToSendData();
|
| - break;
|
| - // TODO(ldixon): Unblock after congestion.
|
| - case SCTP_NOTIFICATIONS_STOPPED_EVENT:
|
| - LOG(LS_INFO) << "SCTP_NOTIFICATIONS_STOPPED_EVENT";
|
| - break;
|
| - case SCTP_SEND_FAILED_EVENT:
|
| - LOG(LS_INFO) << "SCTP_SEND_FAILED_EVENT";
|
| - break;
|
| - case SCTP_STREAM_RESET_EVENT:
|
| - OnStreamResetEvent(¬ification.sn_strreset_event);
|
| - break;
|
| - case SCTP_ASSOC_RESET_EVENT:
|
| - LOG(LS_INFO) << "SCTP_ASSOC_RESET_EVENT";
|
| - break;
|
| - case SCTP_STREAM_CHANGE_EVENT:
|
| - LOG(LS_INFO) << "SCTP_STREAM_CHANGE_EVENT";
|
| - // An acknowledgment we get after our stream resets have gone through,
|
| - // if they've failed. We log the message, but don't react -- we don't
|
| - // keep around the last-transmitted set of SSIDs we wanted to close for
|
| - // error recovery. It doesn't seem likely to occur, and if so, likely
|
| - // harmless within the lifetime of a single SCTP association.
|
| - break;
|
| - default:
|
| - LOG(LS_WARNING) << "Unknown SCTP event: "
|
| - << notification.sn_header.sn_type;
|
| - break;
|
| - }
|
| -}
|
| -
|
| -void SctpTransport::OnNotificationAssocChange(const sctp_assoc_change& change) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - switch (change.sac_state) {
|
| - case SCTP_COMM_UP:
|
| - LOG(LS_VERBOSE) << "Association change SCTP_COMM_UP";
|
| - break;
|
| - case SCTP_COMM_LOST:
|
| - LOG(LS_INFO) << "Association change SCTP_COMM_LOST";
|
| - break;
|
| - case SCTP_RESTART:
|
| - LOG(LS_INFO) << "Association change SCTP_RESTART";
|
| - break;
|
| - case SCTP_SHUTDOWN_COMP:
|
| - LOG(LS_INFO) << "Association change SCTP_SHUTDOWN_COMP";
|
| - break;
|
| - case SCTP_CANT_STR_ASSOC:
|
| - LOG(LS_INFO) << "Association change SCTP_CANT_STR_ASSOC";
|
| - break;
|
| - default:
|
| - LOG(LS_INFO) << "Association change UNKNOWN";
|
| - break;
|
| - }
|
| -}
|
| -
|
| -void SctpTransport::OnStreamResetEvent(
|
| - const struct sctp_stream_reset_event* evt) {
|
| - RTC_DCHECK_RUN_ON(network_thread_);
|
| - // A stream reset always involves two RE-CONFIG chunks for us -- we always
|
| - // simultaneously reset a sid's sequence number in both directions. The
|
| - // requesting side transmits a RE-CONFIG chunk and waits for the peer to send
|
| - // one back. Both sides get this SCTP_STREAM_RESET_EVENT when they receive
|
| - // RE-CONFIGs.
|
| - const int num_sids = (evt->strreset_length - sizeof(*evt)) /
|
| - sizeof(evt->strreset_stream_list[0]);
|
| - LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
|
| - << "): Flags = 0x" << std::hex << evt->strreset_flags << " ("
|
| - << ListFlags(evt->strreset_flags) << ")";
|
| - LOG(LS_VERBOSE) << "Assoc = " << evt->strreset_assoc_id << ", Streams = ["
|
| - << ListArray(evt->strreset_stream_list, num_sids)
|
| - << "], Open: [" << ListStreams(open_streams_) << "], Q'd: ["
|
| - << ListStreams(queued_reset_streams_) << "], Sent: ["
|
| - << ListStreams(sent_reset_streams_) << "]";
|
| -
|
| - // If both sides try to reset some streams at the same time (even if they're
|
| - // disjoint sets), we can get reset failures.
|
| - if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) {
|
| - // OK, just try again. The stream IDs sent over when the RESET_FAILED flag
|
| - // is set seem to be garbage values. Ignore them.
|
| - queued_reset_streams_.insert(sent_reset_streams_.begin(),
|
| - sent_reset_streams_.end());
|
| - sent_reset_streams_.clear();
|
| -
|
| - } else if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) {
|
| - // Each side gets an event for each direction of a stream. That is,
|
| - // closing sid k will make each side receive INCOMING and OUTGOING reset
|
| - // events for k. As per RFC6525, Section 5, paragraph 2, each side will
|
| - // get an INCOMING event first.
|
| - for (int i = 0; i < num_sids; i++) {
|
| - const int stream_id = evt->strreset_stream_list[i];
|
| -
|
| - // See if this stream ID was closed by our peer or ourselves.
|
| - StreamSet::iterator it = sent_reset_streams_.find(stream_id);
|
| -
|
| - // The reset was requested locally.
|
| - if (it != sent_reset_streams_.end()) {
|
| - LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
|
| - << "): local sid " << stream_id << " acknowledged.";
|
| - sent_reset_streams_.erase(it);
|
| -
|
| - } else if ((it = open_streams_.find(stream_id)) != open_streams_.end()) {
|
| - // The peer requested the reset.
|
| - LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
|
| - << "): closing sid " << stream_id;
|
| - open_streams_.erase(it);
|
| - SignalStreamClosedRemotely(stream_id);
|
| -
|
| - } else if ((it = queued_reset_streams_.find(stream_id)) !=
|
| - queued_reset_streams_.end()) {
|
| - // The peer requested the reset, but there was a local reset
|
| - // queued.
|
| - LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
|
| - << "): double-sided close for sid " << stream_id;
|
| - // Both sides want the stream closed, and the peer got to send the
|
| - // RE-CONFIG first. Treat it like the local Remove(Send|Recv)Stream
|
| - // finished quickly.
|
| - queued_reset_streams_.erase(it);
|
| -
|
| - } else {
|
| - // This stream is unknown. Sometimes this can be from an
|
| - // RESET_FAILED-related retransmit.
|
| - LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
|
| - << "): Unknown sid " << stream_id;
|
| - }
|
| - }
|
| - }
|
| -
|
| - // Always try to send the queued RESET because this call indicates that the
|
| - // last local RESET or remote RESET has made some progress.
|
| - SendQueuedStreamResets();
|
| -}
|
| -
|
| -} // namespace cricket
|
|
|