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 |