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webrtc/talk/media/sctp/sctpdataengine.cc
kwiberg@webrtc.org eebcab5ce9 rtc::Buffer: Rename length to size, for conformance with the STL 
And add a constructor for creating an uninitialized Buffer of a
specified size.

(I intend to follow up with more Buffer changes, but since it's rather
widely used, the rename is quite noisy and works better as a separate
CL.)

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48579004

Cr-Commit-Position: refs/heads/master@{#8841}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-24 09:20:19 +00:00

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/*
* libjingle
* Copyright 2012 Google Inc. and Robin Seggelmann
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
*
* 1. Redistributions of source code must retain the above copyright notice,
* this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright notice,
* this list of conditions and the following disclaimer in the documentation
* and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/media/sctp/sctpdataengine.h"
#include <stdarg.h>
#include <stdio.h>
#include <sstream>
#include <vector>
#include "talk/media/base/codec.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
#include "usrsctplib/usrsctp.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/safe_conversions.h"
namespace {
typedef cricket::SctpDataMediaChannel::StreamSet 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 (int i = 0; i < ARRAY_SIZE(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* 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();
}
} // namespace
namespace cricket {
typedef rtc::ScopedMessageData<SctpInboundPacket> InboundPacketMessage;
typedef rtc::ScopedMessageData<rtc::Buffer> OutboundPacketMessage;
// The biggest SCTP packet. Starting from a 'safe' wire MTU value of 1280,
// take off 80 bytes for DTLS/TURN/TCP/IP overhead.
static const size_t kSctpMtu = 1200;
enum {
MSG_SCTPINBOUNDPACKET = 1, // MessageData is SctpInboundPacket
MSG_SCTPOUTBOUNDPACKET = 2, // MessageData is rtc:Buffer
};
struct SctpInboundPacket {
rtc::Buffer buffer;
ReceiveDataParams params;
// The |flags| parameter is used by SCTP to distinguish notification packets
// from other types of packets.
int flags;
};
// Helper for logging SCTP messages.
static void debug_sctp_printf(const char *format, ...) {
char s[255];
va_list ap;
va_start(ap, format);
vsnprintf(s, sizeof(s), format, ap);
LOG(LS_INFO) << "SCTP: " << s;
va_end(ap);
}
// Get the PPID to use for the terminating fragment of this type.
static SctpDataMediaChannel::PayloadProtocolIdentifier GetPpid(
cricket::DataMessageType type) {
switch (type) {
default:
case cricket::DMT_NONE:
return SctpDataMediaChannel::PPID_NONE;
case cricket::DMT_CONTROL:
return SctpDataMediaChannel::PPID_CONTROL;
case cricket::DMT_BINARY:
return SctpDataMediaChannel::PPID_BINARY_LAST;
case cricket::DMT_TEXT:
return SctpDataMediaChannel::PPID_TEXT_LAST;
};
}
static bool GetDataMediaType(
SctpDataMediaChannel::PayloadProtocolIdentifier ppid,
cricket::DataMessageType *dest) {
ASSERT(dest != NULL);
switch (ppid) {
case SctpDataMediaChannel::PPID_BINARY_PARTIAL:
case SctpDataMediaChannel::PPID_BINARY_LAST:
*dest = cricket::DMT_BINARY;
return true;
case SctpDataMediaChannel::PPID_TEXT_PARTIAL:
case SctpDataMediaChannel::PPID_TEXT_LAST:
*dest = cricket::DMT_TEXT;
return true;
case SctpDataMediaChannel::PPID_CONTROL:
*dest = cricket::DMT_CONTROL;
return true;
case SctpDataMediaChannel::PPID_NONE:
*dest = cricket::DMT_NONE;
return true;
default:
return false;
}
}
// 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) {
SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(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);
// Note: We have to copy the data; the caller will delete it.
OutboundPacketMessage* msg =
new OutboundPacketMessage(new rtc::Buffer(data, length));
channel->worker_thread()->Post(channel, MSG_SCTPOUTBOUNDPACKET, msg);
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) {
SctpDataMediaChannel* channel = static_cast<SctpDataMediaChannel*>(ulp_info);
// Post data to the channel'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 SctpDataMediaChannel::PayloadProtocolIdentifier ppid =
static_cast<SctpDataMediaChannel::PayloadProtocolIdentifier>(
rtc::HostToNetwork32(rcv.rcv_ppid));
cricket::DataMessageType type = cricket::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 {
SctpInboundPacket* packet = new SctpInboundPacket;
packet->buffer.SetData(data, length);
packet->params.ssrc = rcv.rcv_sid;
packet->params.seq_num = rcv.rcv_ssn;
packet->params.timestamp = rcv.rcv_tsn;
packet->params.type = type;
packet->flags = flags;
// The ownership of |packet| transfers to |msg|.
InboundPacketMessage* msg = new InboundPacketMessage(packet);
channel->worker_thread()->Post(channel, MSG_SCTPINBOUNDPACKET, msg);
}
free(data);
return 1;
}
// Set the initial value of the static SCTP Data Engines reference count.
int SctpDataEngine::usrsctp_engines_count = 0;
SctpDataEngine::SctpDataEngine() {
if (usrsctp_engines_count == 0) {
// First argument is udp_encapsulation_port, which is not releveant for our
// AF_CONN use of sctp.
usrsctp_init(0, cricket::OnSctpOutboundPacket, debug_sctp_printf);
// 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);
// 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. The 'appropriate default' in the
// second paragraph of
// http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-05#section-6.2
// is cricket::kMaxSctpSid.
usrsctp_sysctl_set_sctp_nr_outgoing_streams_default(
cricket::kMaxSctpSid);
}
usrsctp_engines_count++;
cricket::DataCodec codec(kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, 0);
codec.SetParam(kCodecParamPort, kSctpDefaultPort);
codecs_.push_back(codec);
}
SctpDataEngine::~SctpDataEngine() {
usrsctp_engines_count--;
LOG(LS_VERBOSE) << "usrsctp_engines_count:" << usrsctp_engines_count;
if (usrsctp_engines_count == 0) {
// usrsctp_finish() may fail if it's called too soon after the channels 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.";
}
}
DataMediaChannel* SctpDataEngine::CreateChannel(
DataChannelType data_channel_type) {
if (data_channel_type != DCT_SCTP) {
return NULL;
}
return new SctpDataMediaChannel(rtc::Thread::Current());
}
SctpDataMediaChannel::SctpDataMediaChannel(rtc::Thread* thread)
: worker_thread_(thread),
local_port_(kSctpDefaultPort),
remote_port_(kSctpDefaultPort),
sock_(NULL),
sending_(false),
receiving_(false),
debug_name_("SctpDataMediaChannel") {
}
SctpDataMediaChannel::~SctpDataMediaChannel() {
CloseSctpSocket();
}
sockaddr_conn SctpDataMediaChannel::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;
}
bool SctpDataMediaChannel::OpenSctpSocket() {
if (sock_) {
LOG(LS_VERBOSE) << debug_name_
<< "->Ignoring attempt to re-create existing socket.";
return false;
}
sock_ = usrsctp_socket(AF_CONN, SOCK_STREAM, IPPROTO_SCTP,
cricket::OnSctpInboundPacket, NULL, 0, this);
if (!sock_) {
LOG_ERRNO(LS_ERROR) << debug_name_ << "Failed to create SCTP socket.";
return false;
}
// 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_ << "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_ << "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_
<< "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_ << "Failed to set SCTP_NODELAY.";
return false;
}
// Disable MTU discovery
struct sctp_paddrparams params = {0};
params.spp_assoc_id = 0;
params.spp_flags = SPP_PMTUD_DISABLE;
params.spp_pathmtu = kSctpMtu;
if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, &params,
sizeof(params))) {
LOG_ERRNO(LS_ERROR) << debug_name_
<< "Failed to set SCTP_PEER_ADDR_PARAMS.";
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 < ARRAY_SIZE(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_ << "Failed to set SCTP_EVENT type: "
<< event.se_type;
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);
sending_ = true;
return true;
}
void SctpDataMediaChannel::CloseSctpSocket() {
sending_ = false;
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_ = NULL;
usrsctp_deregister_address(this);
}
}
bool SctpDataMediaChannel::Connect() {
LOG(LS_VERBOSE) << debug_name_ << "->Connect().";
// If we already have a socket connection, just return.
if (sock_) {
LOG(LS_WARNING) << 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 (!sock_ && !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_ << "Failed usrsctp_connect. got errno="
<< errno << ", but wanted " << SCTP_EINPROGRESS;
CloseSctpSocket();
return false;
}
return true;
}
void SctpDataMediaChannel::Disconnect() {
// TODO(ldixon): Consider calling |usrsctp_shutdown(sock_, ...)| to do a
// shutdown handshake and remove the association.
CloseSctpSocket();
}
bool SctpDataMediaChannel::SetSend(bool send) {
if (!sending_ && send) {
return Connect();
}
if (sending_ && !send) {
Disconnect();
}
return true;
}
bool SctpDataMediaChannel::SetReceive(bool receive) {
receiving_ = receive;
return true;
}
bool SctpDataMediaChannel::AddSendStream(const StreamParams& stream) {
return AddStream(stream);
}
bool SctpDataMediaChannel::RemoveSendStream(uint32 ssrc) {
return ResetStream(ssrc);
}
bool SctpDataMediaChannel::AddRecvStream(const StreamParams& stream) {
// SCTP DataChannels are always bi-directional and calling AddSendStream will
// enable both sending and receiving on the stream. So AddRecvStream is a
// no-op.
return true;
}
bool SctpDataMediaChannel::RemoveRecvStream(uint32 ssrc) {
// SCTP DataChannels are always bi-directional and calling RemoveSendStream
// will disable both sending and receiving on the stream. So RemoveRecvStream
// is a no-op.
return true;
}
bool SctpDataMediaChannel::SendData(
const SendDataParams& params,
const rtc::Buffer& payload,
SendDataResult* result) {
if (result) {
// Preset |result| to assume an error. If SendData succeeds, we'll
// overwrite |*result| once more at the end.
*result = SDR_ERROR;
}
if (!sending_) {
LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
<< "Not sending packet with ssrc=" << params.ssrc
<< " len=" << payload.size() << " before SetSend(true).";
return false;
}
if (params.type != cricket::DMT_CONTROL &&
open_streams_.find(params.ssrc) == open_streams_.end()) {
LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
<< "Not sending data because ssrc is unknown: "
<< params.ssrc;
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.ssrc;
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;
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;
}
// Called by network interface when a packet has been received.
void SctpDataMediaChannel::OnPacketReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
LOG(LS_VERBOSE) << debug_name_ << "->OnPacketReceived(...): "
<< " length=" << packet->size() << ", sending: " << sending_;
// 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 (sending_) {
// 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 a Post and handled with OnMessage.
usrsctp_conninput(this, packet->data(), packet->size(), 0);
} else {
// TODO(ldixon): Consider caching the packet for very slightly better
// reliability.
}
}
void SctpDataMediaChannel::OnInboundPacketFromSctpToChannel(
SctpInboundPacket* packet) {
LOG(LS_VERBOSE) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
<< "Received SCTP data:"
<< " ssrc=" << packet->params.ssrc
<< " notification: " << (packet->flags & MSG_NOTIFICATION)
<< " length=" << packet->buffer.size();
// Sending a packet with data == NULL (no data) is SCTPs "close the
// connection" message. This sets sock_ = NULL;
if (!packet->buffer.size() || !packet->buffer.data()) {
LOG(LS_INFO) << debug_name_ << "->OnInboundPacketFromSctpToChannel(...): "
"No data, closing.";
return;
}
if (packet->flags & MSG_NOTIFICATION) {
OnNotificationFromSctp(&packet->buffer);
} else {
OnDataFromSctpToChannel(packet->params, &packet->buffer);
}
}
void SctpDataMediaChannel::OnDataFromSctpToChannel(
const ReceiveDataParams& params, rtc::Buffer* buffer) {
if (receiving_) {
LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): "
<< "Posting with length: " << buffer->size()
<< " on stream " << params.ssrc;
// Reports all received messages to upper layers, no matter whether the sid
// is known.
SignalDataReceived(params, buffer->data(), buffer->size());
} else {
LOG(LS_WARNING) << debug_name_ << "->OnDataFromSctpToChannel(...): "
<< "Not receiving packet with sid=" << params.ssrc
<< " len=" << buffer->size() << " before SetReceive(true).";
}
}
bool SctpDataMediaChannel::AddStream(const StreamParams& stream) {
if (!stream.has_ssrcs()) {
return false;
}
const uint32 ssrc = stream.first_ssrc();
if (open_streams_.find(ssrc) != open_streams_.end()) {
LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
<< "Not adding data stream '" << stream.id
<< "' with ssrc=" << ssrc
<< " because stream is already open.";
return false;
} else if (queued_reset_streams_.find(ssrc) != queued_reset_streams_.end()
|| sent_reset_streams_.find(ssrc) != sent_reset_streams_.end()) {
LOG(LS_WARNING) << debug_name_ << "->Add(Send|Recv)Stream(...): "
<< "Not adding data stream '" << stream.id
<< "' with ssrc=" << ssrc
<< " because stream is still closing.";
return false;
}
open_streams_.insert(ssrc);
return true;
}
bool SctpDataMediaChannel::ResetStream(uint32 ssrc) {
// We typically get this called twice for the same stream, once each for
// Send and Recv.
StreamSet::iterator found = open_streams_.find(ssrc);
if (found == open_streams_.end()) {
LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): "
<< "stream not found.";
return false;
} else {
LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << ssrc << "): "
<< "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(ssrc);
// 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;
}
void SctpDataMediaChannel::OnNotificationFromSctp(rtc::Buffer* buffer) {
const sctp_notification& notification =
reinterpret_cast<const sctp_notification&>(*buffer->data());
ASSERT(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";
SignalReadyToSend(true);
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(&notification.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 SctpDataMediaChannel::OnNotificationAssocChange(
const sctp_assoc_change& change) {
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 SctpDataMediaChannel::OnStreamResetEvent(
const struct sctp_stream_reset_event* evt) {
// 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_ssrcs = (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_ssrcs)
<< "], 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_ssrcs; 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();
}
// Puts the specified |param| from the codec identified by |id| into |dest|
// and returns true. Or returns false if it wasn't there, leaving |dest|
// untouched.
static bool GetCodecIntParameter(const std::vector<DataCodec>& codecs,
int id, const std::string& name,
const std::string& param, int* dest) {
std::string value;
Codec match_pattern;
match_pattern.id = id;
match_pattern.name = name;
for (size_t i = 0; i < codecs.size(); ++i) {
if (codecs[i].Matches(match_pattern)) {
if (codecs[i].GetParam(param, &value)) {
*dest = rtc::FromString<int>(value);
return true;
}
}
}
return false;
}
bool SctpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
return GetCodecIntParameter(
codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort,
&remote_port_);
}
bool SctpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
return GetCodecIntParameter(
codecs, kGoogleSctpDataCodecId, kGoogleSctpDataCodecName, kCodecParamPort,
&local_port_);
}
void SctpDataMediaChannel::OnPacketFromSctpToNetwork(
rtc::Buffer* buffer) {
if (buffer->size() > kSctpMtu) {
LOG(LS_ERROR) << debug_name_ << "->OnPacketFromSctpToNetwork(...): "
<< "SCTP seems to have made a packet that is bigger "
"than its official MTU.";
}
MediaChannel::SendPacket(buffer);
}
bool SctpDataMediaChannel::SendQueuedStreamResets() {
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));
std::vector<uint8> 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_ << "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 SctpDataMediaChannel::OnMessage(rtc::Message* msg) {
switch (msg->message_id) {
case MSG_SCTPINBOUNDPACKET: {
rtc::scoped_ptr<InboundPacketMessage> pdata(
static_cast<InboundPacketMessage*>(msg->pdata));
OnInboundPacketFromSctpToChannel(pdata->data().get());
break;
}
case MSG_SCTPOUTBOUNDPACKET: {
rtc::scoped_ptr<OutboundPacketMessage> pdata(
static_cast<OutboundPacketMessage*>(msg->pdata));
OnPacketFromSctpToNetwork(pdata->data().get());
break;
}
}
}
} // namespace cricket
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