webrtc-m79-连通性检测-01
struct IceControllerEvent {
enum Type {
REMOTE_CANDIDATE_GENERATION_CHANGE,
NETWORK_PREFERENCE_CHANGE,
NEW_CONNECTION_FROM_LOCAL_CANDIDATE, //
NEW_CONNECTION_FROM_REMOTE_CANDIDATE, //
NEW_CONNECTION_FROM_UNKNOWN_REMOTE_ADDRESS,
NOMINATION_ON_CONTROLLED_SIDE,
DATA_RECEIVED,
CONNECT_STATE_CHANGE,
SELECTED_CONNECTION_DESTROYED,
// The ICE_CONTROLLER_RECHECK enum value lets an IceController request
// P2PTransportChannel to recheck a switch periodically without an event
// taking place.
ICE_CONTROLLER_RECHECK,
};
IceControllerEvent(const Type& _type) // NOLINT: runtime/explicit
: type(_type) {}
std::string ToString() const;
Type type;
int recheck_delay_ms = 0;
};
1 进行连通性检测有三个地方
在 OnCandidateReady 里我们会用刚分配好的 Port 与每个 remote candidate 建立 Connection，
此外，如果收到了对方的 STUN ping request，那就会立即创建一个 Connection，
再加上添加 remote candidate(通过信令服务器传递过来的) 的情况，
这三种情况下，创建完 Connection 之后，P2PTransportChannel
都会立即执行 SortConnectionsAndUpdateState 函数，其中首先会对 Connection 进行排序（见下文），
此外也会尝试开始 ping Connection。(连通性检测的这三个地方都会通过 SortConnectionsAndUpdateState 进行排序)
创建 Connection 会触发 ping(通过 MaybeStartPinging() 触发，该函数就是在 P2PTransportChannel::SortConnectionsAndUpdateState 中被调用的)，
ping 成功后会触发 Connection 的状态切换（见下文）。
排序后，我们最终会选出一个合适的 Connection，通知上层(通过信号 SignalReadyToSend)可以进行数据通讯了。
2 连通性检测
2.1 搜集 host candidate 后进行连通性检测
BasicPortAllocatorSession::OnCandidateReady
===>
SignalPortReady(this, port); P2PTransportChannel::AddAllocatorSession 中绑定这个信号到 P2PTransportChannel::OnPortReady
// A new port is available, attempt to make connections for it
void P2PTransportChannel::OnPortReady(PortAllocatorSession* session,
PortInterface* port) { 这里的 port 实际上就是 Port 的指针
RTC_DCHECK_RUN_ON(network_thread_); / 这个 Port 的指针具体可以指向 UDPPort 和 TurnPort
// Set in-effect options on the new port
for (OptionMap::const_iterator it = options_.begin(); it != options_.end();
++it) {
int val = port->SetOption(it->first, it->second);
if (val < 0) {
// Errors are frequent, so use LS_INFO. bugs.webrtc.org/9221
RTC_LOG(LS_INFO) << port->ToString() << ": SetOption(" << it->first
<< ", " << it->second
<< ") failed: " << port->GetError();
}
}
// Remember the ports and candidates, and signal that candidates are ready.
// The session will handle this, and send an initiate/accept/modify message
// if one is pending.
port->SetIceRole(ice_role_);
port->SetIceTiebreaker(tiebreaker_);
ports_.push_back(port); // 将收集的 candidate 的 port 保存下来，后面会使用，比如： P2PTransportChannel::CreateConnections
port->SignalUnknownAddress.connect(this,
&P2PTransportChannel::OnUnknownAddress);
port->SignalDestroyed.connect(this, &P2PTransportChannel::OnPortDestroyed);
port->SignalRoleConflict.connect(this, &P2PTransportChannel::OnRoleConflict);
port->SignalSentPacket.connect(this, &P2PTransportChannel::OnSentPacket); / 就是在这里对 Port 的 SignalSentPacket 事件进行了绑定 //
// Attempt to create a connection from this new port to all of the remote
// candidates that we were given so far.
std::vector::iterator iter;
for (iter = remote_candidates_.begin(); iter != remote_candidates_.end();
++iter) {
CreateConnection(port, *iter, iter->origin_port());
}
SortConnectionsAndUpdateState( 连通性检测
IceControllerEvent::NEW_CONNECTION_FROM_LOCAL_CANDIDATE); 注意这里是 local candidate
}
2.2 收到信令服务器发送过来的 remote candidate 后进行连通性检测
收到对端的 candidate 后开始连通性检测，接收 对端 candidate 的流程见前面分析，这里仅分析触发连通性检测的部分
void P2PTransportChannel::FinishAddingRemoteCandidate(
const Candidate& new_remote_candidate) {
RTC_DCHECK_RUN_ON(network_thread_);
// If this candidate matches what was thought to be a peer reflexive
// candidate, we need to update the candidate priority/etc.
for (Connection* conn : connections()) {
conn->MaybeUpdatePeerReflexiveCandidate(new_remote_candidate);
}
// Create connections to this remote candidate.
CreateConnections(new_remote_candidate, NULL);
// Resort the connections list, which may have new elements.
SortConnectionsAndUpdateState( /// 连通性检测
IceControllerEvent::NEW_CONNECTION_FROM_REMOTE_CANDIDATE); 注意这里是 remote candidate
}
STUN ping request 其实就是 STUN binding request，所以它的发送、response 的接收，前面都已经分析过了。
2.3 收到了对方的 STUN ping request 进行连通性检测
/// 接收 STUN ping request /
/
事件循环参考前面描述，这里直接借用前面分析的结果
void SocketDispatcher::OnEvent(uint32_t ff, int err) {
#if defined(WEBRTC_USE_EPOLL)
// Remember currently enabled events so we can combine multiple changes
// into one update call later.
// The signal handlers might re-enable events disabled here, so we can't
// keep a list of events to disable at the end of the method. This list
// would not be updated with the events enabled by the signal handlers.
StartBatchedEventUpdates();
#endif
// Make sure we deliver connect/accept first. Otherwise, consumers may see
// something like a READ followed by a CONNECT, which would be odd.
if ((ff & DE_CONNECT) != 0) {
DisableEvents(DE_CONNECT);
SignalConnectEvent(this);
}
if ((ff & DE_ACCEPT) != 0) {
DisableEvents(DE_ACCEPT);
SignalReadEvent(this);
}
if ((ff & DE_READ) != 0) {
DisableEvents(DE_READ);
SignalReadEvent(this);// 在 BasicPacketSocketFactory::CreateUdpSocket 中将 SocketDispatcher 的指针传递到了 AsyncUDPSocket ，
}// 并在 AsyncUDPSocket 的构造函数中设置了 SocketDispatcher 的 SignalReadEvent 和 SignalWriteEvent ，分别将其绑定到了 AsyncUDPSocket::OnReadEvent AsyncUDPSocket::OnWriteEvent
if ((ff & DE_WRITE) != 0) {
DisableEvents(DE_WRITE);
SignalWriteEvent(this);
}
if ((ff & DE_CLOSE) != 0) {
// The socket is now dead to us, so stop checking it.
SetEnabledEvents(0);
SignalCloseEvent(this, err);
}
#if defined(WEBRTC_USE_EPOLL)
FinishBatchedEventUpdates();
#endif
}
void AsyncUDPSocket::OnReadEvent(AsyncSocket* socket) {
RTC_DCHECK(socket_.get() == socket);
SocketAddress remote_addr;
int64_t timestamp;
int len = socket_->RecvFrom(buf_, size_, &remote_addr, ×tamp); /
if (len < 0) {
// An error here typically means we got an ICMP error in response to our
// send datagram, indicating the remote address was unreachable.
// When doing ICE, this kind of thing will often happen.
// TODO: Do something better like forwarding the error to the user.
SocketAddress local_addr = socket_->GetLocalAddress();
RTC_LOG(LS_INFO) << "AsyncUDPSocket[" << local_addr.ToSensitiveString()
<< "] receive failed with error " << socket_->GetError();
return;
}
// TODO: Make sure that we got all of the packet.
// If we did not, then we should resize our buffer to be large enough.
SignalReadPacket(this, buf_, static_cast(len), remote_addr,
(timestamp > -1 ? timestamp : TimeMicros())); AsyncUDPSocket 的 SignalReadPacket 事件来自其父类 AsyncPacketSocket
} // AsyncPacketSocket 的 SignalReadPacket 事件是在 AllocationSequence::Init 中设置，将其绑定到了 AllocationSequence::OnReadPacket
void AllocationSequence::OnReadPacket(rtc::AsyncPacketSocket* socket,
const char* data,
size_t size,
const rtc::SocketAddress& remote_addr,
const int64_t& packet_time_us) {
RTC_DCHECK(socket == udp_socket_.get()); udp_socket_ 是 AsyncPacketSocket 指针，且该指针指向 AsyncUDPSocket
bool turn_port_found = false;
// Try to find the TurnPort that matches the remote address. Note that the
// message could be a STUN binding response if the TURN server is also used as
// a STUN server. We don't want to parse every message here to check if it is
// a STUN binding response, so we pass the message to TurnPort regardless of
// the message type. The TurnPort will just ignore the message since it will
// not find any request by transaction ID.
for (auto* port : relay_ports_) {
if (port->CanHandleIncomingPacketsFrom(remote_addr)) {
if (port->HandleIncomingPacket(socket, data, size, remote_addr,
packet_time_us)) {
return;
}
turn_port_found = true;
}
}
if (udp_port_) { udp_port_ 是 UDPPort 指针(UDPPort内部包含 rtc::AsyncPacketSocket 指针，实际上指向 AsyncUDPSocket)
const ServerAddresses& stun_servers = udp_port_->server_addresses();
// Pass the packet to the UdpPort if there is no matching TurnPort, or if
// the TURN server is also a STUN server.
if (!turn_port_found ||
stun_servers.find(remote_addr) != stun_servers.end()) {
RTC_DCHECK(udp_port_->SharedSocket());
udp_port_->HandleIncomingPacket(socket, data, size, remote_addr,
packet_time_us); // UDPPort::HandleIncomingPacket
}
}
}
bool UDPPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
const char* data,
size_t size,
const rtc::SocketAddress& remote_addr,
int64_t packet_time_us) {
// All packets given to UDP port will be consumed.
OnReadPacket(socket, data, size, remote_addr, packet_time_us);
return true;
}
void UDPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
const char* data,
size_t size,
const rtc::SocketAddress& remote_addr,
const int64_t& packet_time_us) {
RTC_DCHECK(socket == socket_);
RTC_DCHECK(!remote_addr.IsUnresolvedIP());
// Look for a response from the STUN server.
// Even if the response doesn't match one of our outstanding requests, we
// will eat it because it might be a response to a retransmitted packet, and
// we already cleared the request when we got the first response.
if (server_addresses_.find(remote_addr) != server_addresses_.end()) {
requests_.CheckResponse(data, size); StunRequestManager::CheckResponse
return;
}
// 接收到的是 stun request 且之前建立过 connection
if (Connection* conn = GetConnection(remote_addr)) {
conn->OnReadPacket(data, size, packet_time_us);
} else {// 接收到的是 stun request 且之前没有建立过 connection ，也就是 pflx candidate
Port::OnReadPacket(data, size, remote_addr, PROTO_UDP);
}
}
void Port::OnReadPacket(const char* data,
size_t size,
const rtc::SocketAddress& addr,
ProtocolType proto) {
// If the user has enabled port packets, just hand this over.
if (enable_port_packets_) {
SignalReadPacket(this, data, size, addr);
return;
}
// If this is an authenticated STUN request, then signal unknown address and
// send back a proper binding response.
std::unique_ptr msg;
std::string remote_username;
if (!GetStunMessage(data, size, addr, &msg, &remote_username)) {
RTC_LOG(LS_ERROR) << ToString()
<< ": Received non-STUN packet from unknown address: "
<< addr.ToSensitiveString();
} else if (!msg) {
// STUN message handled already
} else if (msg->type() == STUN_BINDING_REQUEST) { 接收到的是对端的 stun ping request
RTC_LOG(LS_INFO) << "Received STUN ping id="
<< rtc::hex_encode(msg->transaction_id())
<< " from unknown address " << addr.ToSensitiveString();
// We need to signal an unknown address before we handle any role conflict
// below. Otherwise there would be no candidate pair and TURN entry created
// to send the error response in case of a role conflict.
SignalUnknownAddress(this, addr, proto, msg.get(), remote_username, false); / P2PTransportChannel::OnPortReady 中关联的是 P2PTransportChannel::OnUnknownAddress
// Check for role conflicts.
if (!MaybeIceRoleConflict(addr, msg.get(), remote_username)) {
RTC_LOG(LS_INFO) << "Received conflicting role from the peer.";
return;
}
} else {
// NOTE(tschmelcher): STUN_BINDING_RESPONSE is benign. It occurs if we
// pruned a connection for this port while it had STUN requests in flight,
// because we then get back responses for them, which this code correctly
// does not handle.
if (msg->type() != STUN_BINDING_RESPONSE) {
RTC_LOG(LS_ERROR) << ToString()
<< ": Received unexpected STUN message type: "
<< msg->type() << " from unknown address: "
<< addr.ToSensitiveString();
}
}
}
void P2PTransportChannel::OnUnknownAddress(PortInterface* port,
const rtc::SocketAddress& address,
ProtocolType proto,
IceMessage* stun_msg,
const std::string& remote_username,
bool port_muxed) {
RTC_DCHECK_RUN_ON(network_thread_);
// Port has received a valid stun packet from an address that no Connection
// is currently available for. See if we already have a candidate with the
// address. If it isn't we need to create new candidate for it.
//
// TODO(qingsi): There is a caveat of the logic below if we have remote
// candidates with hostnames. We could create a prflx candidate that is
// identical to a host candidate that are currently in the process of name
// resolution. We would not have a duplicate candidate since when adding the
// resolved host candidate, FinishingAddingRemoteCandidate does
// MaybeUpdatePeerReflexiveCandidate, and the prflx candidate would be updated
// to a host candidate. As a result, for a brief moment we would have a prflx
// candidate showing a private IP address, though we do not signal prflx
// candidates to applications and we could obfuscate the IP addresses of prflx
// candidates in P2PTransportChannel::GetStats. The difficulty of preventing
// creating the prflx from the beginning is that we do not have a reliable way
// to claim two candidates are identical without the address information. If
// we always pause the addition of a prflx candidate when there is ongoing
// name resolution and dedup after we have a resolved address, we run into the
// risk of losing/delaying the addition of a non-identical candidate that
// could be the only way to have a connection, if the resolution never
// completes or is significantly delayed.
const Candidate* candidate = nullptr;
for (const Candidate& c : remote_candidates_) {
if (c.username() == remote_username && c.address() == address &&
c.protocol() == ProtoToString(proto)) {
candidate = &c;
break;
}
}
uint32_t remote_generation = 0;
std::string remote_password;
// The STUN binding request may arrive after setRemoteDescription and before
// adding remote candidate, so we need to set the password to the shared
// password and set the generation if the user name matches.
const IceParameters* ice_param =
FindRemoteIceFromUfrag(remote_username, &remote_generation);
// Note: if not found, the remote_generation will still be 0.
if (ice_param != nullptr) {
remote_password = ice_param->pwd;
}
Candidate remote_candidate;
bool remote_candidate_is_new = (candidate == nullptr);
if (!remote_candidate_is_new) {
remote_candidate = *candidate;
} else {
// Create a new candidate with this address.
// The priority of the candidate is set to the PRIORITY attribute
// from the request.
const StunUInt32Attribute* priority_attr =
stun_msg->GetUInt32(STUN_ATTR_PRIORITY);
if (!priority_attr) {
RTC_LOG(LS_WARNING) << "P2PTransportChannel::OnUnknownAddress - "
"No STUN_ATTR_PRIORITY found in the "
"stun request message";
port->SendBindingErrorResponse(stun_msg, address, STUN_ERROR_BAD_REQUEST,
STUN_ERROR_REASON_BAD_REQUEST);
return;
}
int remote_candidate_priority = priority_attr->value();
uint16_t network_id = 0;
uint16_t network_cost = 0;
const StunUInt32Attribute* network_attr =
stun_msg->GetUInt32(STUN_ATTR_NETWORK_INFO);
if (network_attr) {
uint32_t network_info = network_attr->value();
network_id = static_cast(network_info >> 16);
network_cost = static_cast(network_info);
}
// RFC 5245
// If the source transport address of the request does not match any
// existing remote candidates, it represents a new peer reflexive remote
// candidate.
remote_candidate = Candidate(
component(), ProtoToString(proto), address, remote_candidate_priority,
remote_username, remote_password, PRFLX_PORT_TYPE, remote_generation,
"", network_id, network_cost);
// From RFC 5245, section-7.2.1.3:
// The foundation of the candidate is set to an arbitrary value, different
// from the foundation for all other remote candidates.
remote_candidate.set_foundation(
rtc::ToString(rtc::ComputeCrc32(remote_candidate.id())));
}
// RFC5245, the agent constructs a pair whose local candidate is equal to
// the transport address on which the STUN request was received, and a
// remote candidate equal to the source transport address where the
// request came from.
// There shouldn't be an existing connection with this remote address.
// When ports are muxed, this channel might get multiple unknown address
// signals. In that case if the connection is already exists, we should
// simply ignore the signal otherwise send server error.
if (port->GetConnection(remote_candidate.address())) {
if (port_muxed) {
RTC_LOG(LS_INFO) << "Connection already exists for peer reflexive "
"candidate: "
<< remote_candidate.ToSensitiveString();
return;
} else {
RTC_NOTREACHED();
port->SendBindingErrorResponse(stun_msg, address, STUN_ERROR_SERVER_ERROR,
STUN_ERROR_REASON_SERVER_ERROR);
return;
}
}
Connection* connection = // 这里也就是接收了对端的 stun ping request 之后创建的 connection
port->CreateConnection(remote_candidate, PortInterface::ORIGIN_THIS_PORT); /// UDPPort::CreateConnection
if (!connection) {
// This could happen in some scenarios. For example, a TurnPort may have
// had a refresh request timeout, so it won't create connections.
port->SendBindingErrorResponse(stun_msg, address, STUN_ERROR_SERVER_ERROR,
STUN_ERROR_REASON_SERVER_ERROR);
return;
}
RTC_LOG(LS_INFO) << "Adding connection from "
<< (remote_candidate_is_new ? "peer reflexive"
: "resurrected")
<< " candidate: " << remote_candidate.ToSensitiveString();
AddConnection(connection);
connection->HandleBindingRequest(stun_msg);
// Update the list of connections since we just added another. We do this
// after sending the response since it could (in principle) delete the
// connection in question.
SortConnectionsAndUpdateState(
"a new candidate pair created from an unknown remote address");
}
// Sort the available connections to find the best one. We also monitor
// the number of available connections and the current state.
void P2PTransportChannel::SortConnectionsAndUpdateState(
IceControllerEvent reason_to_sort) {
RTC_DCHECK_RUN_ON(network_thread_);
// Make sure the connection states are up-to-date since this affects how they
// will be sorted.
UpdateConnectionStates();
// Any changes after this point will require a re-sort.
sort_dirty_ = false;
// If necessary, switch to the new choice. Note that |top_connection| doesn't
// have to be writable to become the selected connection although it will
// have higher priority if it is writable.
MaybeSwitchSelectedConnection(
reason_to_sort, ice_controller_->SortAndSwitchConnection(reason_to_sort));
// The controlled side can prune only if the selected connection has been
// nominated because otherwise it may prune the connection that will be
// selected by the controlling side.
// TODO(honghaiz): This is not enough to prevent a connection from being
// pruned too early because with aggressive nomination, the controlling side
// will nominate every connection until it becomes writable.
if (ice_role_ == ICEROLE_CONTROLLING ||
(selected_connection_ && selected_connection_->nominated())) {
PruneConnections();
}
// Check if all connections are timedout.
bool all_connections_timedout = true;
for (const Connection* conn : connections()) {
if (conn->write_state() != Connection::STATE_WRITE_TIMEOUT) {
all_connections_timedout = false;
break;
}
}
// Now update the writable state of the channel with the information we have
// so far.
if (all_connections_timedout) {
HandleAllTimedOut();
}
// Update the state of this channel.
UpdateState();
// Also possibly start pinging.
// We could start pinging if:
// * The first connection was created.
// * ICE credentials were provided.
// * A TCP connection became connected.
MaybeStartPinging(); // 这里就是上面说的触发 ping
}
void P2PTransportChannel::MaybeStartPinging() {
RTC_DCHECK_RUN_ON(network_thread_);
if (started_pinging_) {
return;
}
if (ice_controller_->HasPingableConnection()) {
RTC_LOG(LS_INFO) << ToString()
<< ": Have a pingable connection for the first time; "
"starting to ping.";
invoker_.AsyncInvoke(
RTC_FROM_HERE, thread(),
rtc::Bind(&P2PTransportChannel::CheckAndPing, this)); /
regathering_controller_->Start();
started_pinging_ = true;
}
}
// Handle queued up check-and-ping request
void P2PTransportChannel::CheckAndPing() {
RTC_DCHECK_RUN_ON(network_thread_);
// Make sure the states of the connections are up-to-date (since this affects
// which ones are pingable).
UpdateConnectionStates();
auto result = ice_controller_->SelectConnectionToPing(last_ping_sent_ms_);
int delay = result.recheck_delay_ms;
if (result.connection.value_or(nullptr)) {
Connection* conn = FromIceController(*result.connection);
PingConnection(conn); /
MarkConnectionPinged(conn);
}
invoker_.AsyncInvokeDelayed(
RTC_FROM_HERE, thread(),
rtc::Bind(&P2PTransportChannel::CheckAndPing, this), delay);
}
void P2PTransportChannel::PingConnection(Connection* conn) {
RTC_DCHECK_RUN_ON(network_thread_);
bool use_candidate_attr = false;
uint32_t nomination = 0;
if (ice_role_ == ICEROLE_CONTROLLING) {
bool renomination_supported = ice_parameters_.renomination &&
!remote_ice_parameters_.empty() &&
remote_ice_parameters_.back().renomination;
if (renomination_supported) {
nomination = GetNominationAttr(conn);
} else {
use_candidate_attr = GetUseCandidateAttr(conn);
}
}
conn->set_nomination(nomination);
conn->set_use_candidate_attr(use_candidate_attr);
last_ping_sent_ms_ = rtc::TimeMillis();
conn->Ping(last_ping_sent_ms_); conn 是 Connection 指针，不过指向的是 ProxyConnection ， 具体参考 UDPPort::CreateConnection 或者 TurnPort::CreateConnection
} ProxyConnection::Ping 就是 Connection::Ping
class ProxyConnection : public Connection {
public:
ProxyConnection(Port* port, size_t index, const Candidate& remote_candidate);
int Send(const void* data,
size_t size,
const rtc::PacketOptions& options) override;
int GetError() override;
private:
int error_ = 0;
};
ProxyConnection::ProxyConnection(Port* port,
size_t index,
const Candidate& remote_candidate)
: Connection(port, index, remote_candidate) {}
void Connection::Ping(int64_t now) {
last_ping_sent_ = now;
ConnectionRequest* req = new ConnectionRequest(this); /// class ConnectionRequest : public StunRequest
// If not using renomination, we use "1" to mean "nominated" and "0" to mean
// "not nominated". If using renomination, values greater than 1 are used for
// re-nominated pairs.
int nomination = use_candidate_attr_ ? 1 : 0;
if (nomination_ > 0) {
nomination = nomination_;
}
pings_since_last_response_.push_back(SentPing(req->id(), now, nomination)); //
RTC_LOG(LS_VERBOSE) << ToString() << ": Sending STUN ping, id="
<< rtc::hex_encode(req->id())
<< ", nomination=" << nomination_;
requests_.Send(req); / StunRequestManager::Send
state_ = IceCandidatePairState::IN_PROGRESS; ///
num_pings_sent_++;
}
void StunRequestManager::Send(StunRequest* request) {
SendDelayed(request, 0);
}
void StunRequestManager::SendDelayed(StunRequest* request, int delay) {
request->set_manager(this);
RTC_DCHECK(requests_.find(request->id()) == requests_.end());
request->set_origin(origin_);
request->Construct();
requests_[request->id()] = request; / 将 ConnectionRequest 保存下来用来配对响应
if (delay > 0) {
thread_->PostDelayed(RTC_FROM_HERE, delay, request, MSG_STUN_SEND, NULL);
} else {
thread_->Send(RTC_FROM_HERE, request, MSG_STUN_SEND, NULL); /
}
}
void StunRequest::OnMessage(rtc::Message* pmsg) {
RTC_DCHECK(manager_ != NULL);
RTC_DCHECK(pmsg->message_id == MSG_STUN_SEND);
if (timeout_) {
OnTimeout();
delete this;
return;
}
tstamp_ = rtc::TimeMillis();
rtc::ByteBufferWriter buf;
msg_->Write(&buf);
manager_->SignalSendPacket(buf.Data(), buf.Length(), this); / StunRequestManager->SignalSendPacket
StunRequestManager SignalSendPacket 信号的绑定是在 Connection 的构造函数中，具体如下
requests_.SignalSendPacket.connect(this, &Connection::OnSendStunPacket);
区分 UDPPort::Init 和 TurnPort::TurnPort 中对 StunRequestManager SignalSendPacket 信号的绑定
// UDPPort::Init ==> requests_.SignalSendPacket.connect(this, &UDPPort::OnSendPacket); requests_ 就是 StunRequestManager
// TurnPort::TurnPort ===> request_manager_.SignalSendPacket.connect(this, &TurnPort::OnSendStunPacket);
OnSent();
manager_->thread_->PostDelayed(RTC_FROM_HERE, resend_delay(), this,
MSG_STUN_SEND, NULL);
}
ProxyConnection::ProxyConnection(Port* port,
size_t index,
const Candidate& remote_candidate)
: Connection(port, index, remote_candidate) {}
Connection::Connection(Port* port,
size_t index,
const Candidate& remote_candidate)
: id_(rtc::CreateRandomId()),
port_(port), /
local_candidate_index_(index),
remote_candidate_(remote_candidate), //
recv_rate_tracker_(100, 10u),
send_rate_tracker_(100, 10u),
write_state_(STATE_WRITE_INIT),
receiving_(false),
connected_(true),
pruned_(false), /
use_candidate_attr_(false),
remote_ice_mode_(ICEMODE_FULL),
requests_(port->thread()),
rtt_(DEFAULT_RTT),
last_ping_sent_(0),
last_ping_received_(0),
last_data_received_(0),
last_ping_response_received_(0),
reported_(false),
state_(IceCandidatePairState::WAITING),
time_created_ms_(rtc::TimeMillis()),
field_trials_(&kDefaultFieldTrials),
rtt_estimate_(DEFAULT_RTT_ESTIMATE_HALF_TIME_MS) {
// All of our connections start in WAITING state.
// TODO(mallinath) - Start connections from STATE_FROZEN.
// Wire up to send stun packets
requests_.SignalSendPacket.connect(this, &Connection::OnSendStunPacket); / StunRequestManager SignalSendPacket 信号的绑定
RTC_LOG(LS_INFO) << ToString() << ": Connection created";
}
void Connection::OnSendStunPacket(const void* data,
size_t size,
StunRequest* req) {
rtc::PacketOptions options(port_->StunDscpValue());
options.info_signaled_after_sent.packet_type =
rtc::PacketType::kIceConnectivityCheck;
auto err =
port_->SendTo(data, size, remote_candidate_.address(), options, false); / UDPPort::SendTo 或者 TurnPort::SendTo
if (err < 0) {
RTC_LOG(LS_WARNING) << ToString()
<< ": Failed to send STUN ping "
" err="
<< err << " id=" << rtc::hex_encode(req->id());
}
}
/ 这里以 UDPPort::SendTo 为例 /
int UDPPort::SendTo(const void* data,
size_t size,
const rtc::SocketAddress& addr,
const rtc::PacketOptions& options,
bool payload) {
rtc::PacketOptions modified_options(options);
CopyPortInformationToPacketInfo(&modified_options.info_signaled_after_sent);
int sent = socket_->SendTo(data, size, addr, modified_options); AsyncUDPSocket::SendTo 直接发给对端的地址(也就是candidate pair 的另一端)
if (sent < 0) {
error_ = socket_->GetError();
// Rate limiting added for crbug.com/856088.
// TODO(webrtc:9622): Use general rate limiting mechanism once it exists.
if (send_error_count_ < kSendErrorLogLimit) {
++send_error_count_;
RTC_LOG(LS_ERROR) << ToString() << ": UDP send of " << size
<< " bytes failed with error " << error_;
}
} else {
send_error_count_ = 0;
}
return sent;
}
int AsyncUDPSocket::SendTo(const void* pv,
size_t cb,
const SocketAddress& addr,
const rtc::PacketOptions& options) {
rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(),
options.info_signaled_after_sent);
CopySocketInformationToPacketInfo(cb, *this, true, &sent_packet.info); class SocketDispatcher : public Dispatcher, public PhysicalSocket
int ret = socket_->SendTo(pv, cb, addr); / socket_ 实际上就是 SocketDispatcher 的指针，具体见 BasicPacketSocketFactory::CreateUdpSocket
SignalSentPacket(this, sent_packet); / SocketDispatcher::SendTo 就是 PhysicalSocket::SendTo
return ret; / SignalSentPacket 来自 AsyncPacketSocket ， 因为 class AsyncUDPSocket : public AsyncPacketSocket
} / 这里 SignalSentPacket 就会触发 UDPPort::OnSentPacket ， 具体见 UDPPort::Init
AsyncUDPSocket::AsyncUDPSocket(AsyncSocket* socket) : socket_(socket) {
size_ = BUF_SIZE;
buf_ = new char[size_];
// The socket should start out readable but not writable.
socket_->SignalReadEvent.connect(this, &AsyncUDPSocket::OnReadEvent);
socket_->SignalWriteEvent.connect(this, &AsyncUDPSocket::OnWriteEvent);
}
int PhysicalSocket::SendTo(const void* buffer,
size_t length,
const SocketAddress& addr) {
sockaddr_storage saddr;
size_t len = addr.ToSockAddrStorage(&saddr);
int sent =
DoSendTo(s_, static_cast(buffer), static_cast(length),
#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
// Suppress SIGPIPE. See above for explanation.
MSG_NOSIGNAL,
#else
0,
#endif
reinterpret_cast(&saddr), static_cast(len));
UpdateLastError();
MaybeRemapSendError();
// We have seen minidumps where this may be false.
RTC_DCHECK(sent <= static_cast(length));
if ((sent > 0 && sent < static_cast(length)) ||
(sent < 0 && IsBlockingError(GetError()))) {
EnableEvents(DE_WRITE);
}
return sent;
}
int PhysicalSocket::DoSendTo(SOCKET socket,
const char* buf,
int len,
int flags,
const struct sockaddr* dest_addr,
socklen_t addrlen) {
return ::sendto(socket, buf, len, flags, dest_addr, addrlen); //
}