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webrtc / src / a535ee664c93343049d1eeeacad51a27de469fc5 / . / pc / peer_connection.cc
blob: 367341bbb3e5c23d8134e246a2188cc037d120da [file] [log] [blame]
/*
* Copyright 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 "pc/peer_connection.h"
#include <limits.h>
#include <stddef.h>
#include <algorithm>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "api/jsep_ice_candidate.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/uma_metrics.h"
#include "api/video/video_codec_constants.h"
#include "call/audio_state.h"
#include "call/packet_receiver.h"
#include "media/base/media_channel.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "p2p/base/basic_async_resolver_factory.h"
#include "p2p/base/connection.h"
#include "p2p/base/connection_info.h"
#include "p2p/base/dtls_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/p2p_transport_channel.h"
#include "p2p/base/transport_info.h"
#include "pc/ice_server_parsing.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_receiver_proxy.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_sender_proxy.h"
#include "pc/sctp_transport.h"
#include "pc/simulcast_description.h"
#include "pc/webrtc_session_description_factory.h"
#include "rtc_base/helpers.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/net_helper.h"
#include "rtc_base/network.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/unique_id_generator.h"
#include "system_wrappers/include/metrics.h"
using cricket::ContentInfo;
using cricket::ContentInfos;
using cricket::MediaContentDescription;
using cricket::MediaProtocolType;
using cricket::RidDescription;
using cricket::RidDirection;
using cricket::SessionDescription;
using cricket::SimulcastDescription;
using cricket::SimulcastLayer;
using cricket::SimulcastLayerList;
using cricket::StreamParams;
using cricket::TransportInfo;
using cricket::LOCAL_PORT_TYPE;
using cricket::PRFLX_PORT_TYPE;
using cricket::RELAY_PORT_TYPE;
using cricket::STUN_PORT_TYPE;
namespace webrtc {
namespace {
// UMA metric names.
const char kSimulcastNumberOfEncodings[] =
"WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings";
static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
uint32_t ConvertIceTransportTypeToCandidateFilter(
PeerConnectionInterface::IceTransportsType type) {
switch (type) {
case PeerConnectionInterface::kNone:
return cricket::CF_NONE;
case PeerConnectionInterface::kRelay:
return cricket::CF_RELAY;
case PeerConnectionInterface::kNoHost:
return (cricket::CF_ALL & ~cricket::CF_HOST);
case PeerConnectionInterface::kAll:
return cricket::CF_ALL;
default:
RTC_DCHECK_NOTREACHED();
}
return cricket::CF_NONE;
}
IceCandidatePairType GetIceCandidatePairCounter(
const cricket::Candidate& local,
const cricket::Candidate& remote) {
const auto& l = local.type();
const auto& r = remote.type();
const auto& host = LOCAL_PORT_TYPE;
const auto& srflx = STUN_PORT_TYPE;
const auto& relay = RELAY_PORT_TYPE;
const auto& prflx = PRFLX_PORT_TYPE;
if (l == host && r == host) {
bool local_hostname =
!local.address().hostname().empty() && local.address().IsUnresolvedIP();
bool remote_hostname = !remote.address().hostname().empty() &&
remote.address().IsUnresolvedIP();
bool local_private = IPIsPrivate(local.address().ipaddr());
bool remote_private = IPIsPrivate(remote.address().ipaddr());
if (local_hostname) {
if (remote_hostname) {
return kIceCandidatePairHostNameHostName;
} else if (remote_private) {
return kIceCandidatePairHostNameHostPrivate;
} else {
return kIceCandidatePairHostNameHostPublic;
}
} else if (local_private) {
if (remote_hostname) {
return kIceCandidatePairHostPrivateHostName;
} else if (remote_private) {
return kIceCandidatePairHostPrivateHostPrivate;
} else {
return kIceCandidatePairHostPrivateHostPublic;
}
} else {
if (remote_hostname) {
return kIceCandidatePairHostPublicHostName;
} else if (remote_private) {
return kIceCandidatePairHostPublicHostPrivate;
} else {
return kIceCandidatePairHostPublicHostPublic;
}
}
}
if (l == host && r == srflx)
return kIceCandidatePairHostSrflx;
if (l == host && r == relay)
return kIceCandidatePairHostRelay;
if (l == host && r == prflx)
return kIceCandidatePairHostPrflx;
if (l == srflx && r == host)
return kIceCandidatePairSrflxHost;
if (l == srflx && r == srflx)
return kIceCandidatePairSrflxSrflx;
if (l == srflx && r == relay)
return kIceCandidatePairSrflxRelay;
if (l == srflx && r == prflx)
return kIceCandidatePairSrflxPrflx;
if (l == relay && r == host)
return kIceCandidatePairRelayHost;
if (l == relay && r == srflx)
return kIceCandidatePairRelaySrflx;
if (l == relay && r == relay)
return kIceCandidatePairRelayRelay;
if (l == relay && r == prflx)
return kIceCandidatePairRelayPrflx;
if (l == prflx && r == host)
return kIceCandidatePairPrflxHost;
if (l == prflx && r == srflx)
return kIceCandidatePairPrflxSrflx;
if (l == prflx && r == relay)
return kIceCandidatePairPrflxRelay;
return kIceCandidatePairMax;
}
absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
int rtc_configuration_parameter) {
if (rtc_configuration_parameter ==
webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
return absl::nullopt;
}
return rtc_configuration_parameter;
}
// Check if the changes of IceTransportsType motives an ice restart.
bool NeedIceRestart(bool surface_ice_candidates_on_ice_transport_type_changed,
PeerConnectionInterface::IceTransportsType current,
PeerConnectionInterface::IceTransportsType modified) {
if (current == modified) {
return false;
}
if (!surface_ice_candidates_on_ice_transport_type_changed) {
return true;
}
auto current_filter = ConvertIceTransportTypeToCandidateFilter(current);
auto modified_filter = ConvertIceTransportTypeToCandidateFilter(modified);
// If surface_ice_candidates_on_ice_transport_type_changed is true and we
// extend the filter, then no ice restart is needed.
return (current_filter & modified_filter) != current_filter;
}
cricket::IceConfig ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) {
cricket::ContinualGatheringPolicy gathering_policy;
switch (config.continual_gathering_policy) {
case PeerConnectionInterface::GATHER_ONCE:
gathering_policy = cricket::GATHER_ONCE;
break;
case PeerConnectionInterface::GATHER_CONTINUALLY:
gathering_policy = cricket::GATHER_CONTINUALLY;
break;
default:
RTC_DCHECK_NOTREACHED();
gathering_policy = cricket::GATHER_ONCE;
}
cricket::IceConfig ice_config;
ice_config.receiving_timeout = RTCConfigurationToIceConfigOptionalInt(
config.ice_connection_receiving_timeout);
ice_config.prioritize_most_likely_candidate_pairs =
config.prioritize_most_likely_ice_candidate_pairs;
ice_config.backup_connection_ping_interval =
RTCConfigurationToIceConfigOptionalInt(
config.ice_backup_candidate_pair_ping_interval);
ice_config.continual_gathering_policy = gathering_policy;
ice_config.presume_writable_when_fully_relayed =
config.presume_writable_when_fully_relayed;
ice_config.surface_ice_candidates_on_ice_transport_type_changed =
config.surface_ice_candidates_on_ice_transport_type_changed;
ice_config.ice_check_interval_strong_connectivity =
config.ice_check_interval_strong_connectivity;
ice_config.ice_check_interval_weak_connectivity =
config.ice_check_interval_weak_connectivity;
ice_config.ice_check_min_interval = config.ice_check_min_interval;
ice_config.ice_unwritable_timeout = config.ice_unwritable_timeout;
ice_config.ice_unwritable_min_checks = config.ice_unwritable_min_checks;
ice_config.ice_inactive_timeout = config.ice_inactive_timeout;
ice_config.stun_keepalive_interval = config.stun_candidate_keepalive_interval;
ice_config.network_preference = config.network_preference;
ice_config.stable_writable_connection_ping_interval =
config.stable_writable_connection_ping_interval_ms;
return ice_config;
}
// Ensures the configuration doesn't have any parameters with invalid values,
// or values that conflict with other parameters.
//
// Returns RTCError::OK() if there are no issues.
RTCError ValidateConfiguration(
const PeerConnectionInterface::RTCConfiguration& config) {
return cricket::P2PTransportChannel::ValidateIceConfig(
ParseIceConfig(config));
}
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
return content->media_description()->rtcp_mux();
}
bool DtlsEnabled(const PeerConnectionInterface::RTCConfiguration& configuration,
const PeerConnectionFactoryInterface::Options& options,
const PeerConnectionDependencies& dependencies) {
if (options.disable_encryption)
return false;
// Enable DTLS by default if we have an identity store or a certificate.
bool default_enabled =
(dependencies.cert_generator || !configuration.certificates.empty());
#if defined(WEBRTC_FUCHSIA)
// The `configuration` can override the default value.
return configuration.enable_dtls_srtp.value_or(default_enabled);
#else
return default_enabled;
#endif
}
} // namespace
bool PeerConnectionInterface::RTCConfiguration::operator==(
const PeerConnectionInterface::RTCConfiguration& o) const {
// This static_assert prevents us from accidentally breaking operator==.
// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
struct stuff_being_tested_for_equality {
IceServers servers;
IceTransportsType type;
BundlePolicy bundle_policy;
RtcpMuxPolicy rtcp_mux_policy;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
int ice_candidate_pool_size;
bool disable_ipv6;
bool disable_ipv6_on_wifi;
int max_ipv6_networks;
bool disable_link_local_networks;
absl::optional<int> screencast_min_bitrate;
absl::optional<bool> combined_audio_video_bwe;
#if defined(WEBRTC_FUCHSIA)
absl::optional<bool> enable_dtls_srtp;
#endif
TcpCandidatePolicy tcp_candidate_policy;
CandidateNetworkPolicy candidate_network_policy;
int audio_jitter_buffer_max_packets;
bool audio_jitter_buffer_fast_accelerate;
int audio_jitter_buffer_min_delay_ms;
int ice_connection_receiving_timeout;
int ice_backup_candidate_pair_ping_interval;
ContinualGatheringPolicy continual_gathering_policy;
bool prioritize_most_likely_ice_candidate_pairs;
struct cricket::MediaConfig media_config;
bool prune_turn_ports;
PortPrunePolicy turn_port_prune_policy;
bool presume_writable_when_fully_relayed;
bool enable_ice_renomination;
bool redetermine_role_on_ice_restart;
bool surface_ice_candidates_on_ice_transport_type_changed;
absl::optional<int> ice_check_interval_strong_connectivity;
absl::optional<int> ice_check_interval_weak_connectivity;
absl::optional<int> ice_check_min_interval;
absl::optional<int> ice_unwritable_timeout;
absl::optional<int> ice_unwritable_min_checks;
absl::optional<int> ice_inactive_timeout;
absl::optional<int> stun_candidate_keepalive_interval;
webrtc::TurnCustomizer* turn_customizer;
SdpSemantics sdp_semantics;
absl::optional<rtc::AdapterType> network_preference;
bool active_reset_srtp_params;
absl::optional<CryptoOptions> crypto_options;
bool offer_extmap_allow_mixed;
std::string turn_logging_id;
bool enable_implicit_rollback;
absl::optional<bool> allow_codec_switching;
absl::optional<int> report_usage_pattern_delay_ms;
absl::optional<int> stable_writable_connection_ping_interval_ms;
webrtc::VpnPreference vpn_preference;
std::vector<rtc::NetworkMask> vpn_list;
PortAllocatorConfig port_allocator_config;
};
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
"Did you add something to RTCConfiguration and forget to "
"update operator==?");
return type == o.type && servers == o.servers &&
bundle_policy == o.bundle_policy &&
rtcp_mux_policy == o.rtcp_mux_policy &&
tcp_candidate_policy == o.tcp_candidate_policy &&
candidate_network_policy == o.candidate_network_policy &&
audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
audio_jitter_buffer_fast_accelerate ==
o.audio_jitter_buffer_fast_accelerate &&
audio_jitter_buffer_min_delay_ms ==
o.audio_jitter_buffer_min_delay_ms &&
ice_connection_receiving_timeout ==
o.ice_connection_receiving_timeout &&
ice_backup_candidate_pair_ping_interval ==
o.ice_backup_candidate_pair_ping_interval &&
continual_gathering_policy == o.continual_gathering_policy &&
certificates == o.certificates &&
prioritize_most_likely_ice_candidate_pairs ==
o.prioritize_most_likely_ice_candidate_pairs &&
media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
max_ipv6_networks == o.max_ipv6_networks &&
disable_link_local_networks == o.disable_link_local_networks &&
screencast_min_bitrate == o.screencast_min_bitrate &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
#if defined(WEBRTC_FUCHSIA)
enable_dtls_srtp == o.enable_dtls_srtp &&
#endif
ice_candidate_pool_size == o.ice_candidate_pool_size &&
prune_turn_ports == o.prune_turn_ports &&
turn_port_prune_policy == o.turn_port_prune_policy &&
presume_writable_when_fully_relayed ==
o.presume_writable_when_fully_relayed &&
enable_ice_renomination == o.enable_ice_renomination &&
redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
surface_ice_candidates_on_ice_transport_type_changed ==
o.surface_ice_candidates_on_ice_transport_type_changed &&
ice_check_interval_strong_connectivity ==
o.ice_check_interval_strong_connectivity &&
ice_check_interval_weak_connectivity ==
o.ice_check_interval_weak_connectivity &&
ice_check_min_interval == o.ice_check_min_interval &&
ice_unwritable_timeout == o.ice_unwritable_timeout &&
ice_unwritable_min_checks == o.ice_unwritable_min_checks &&
ice_inactive_timeout == o.ice_inactive_timeout &&
stun_candidate_keepalive_interval ==
o.stun_candidate_keepalive_interval &&
turn_customizer == o.turn_customizer &&
sdp_semantics == o.sdp_semantics &&
network_preference == o.network_preference &&
active_reset_srtp_params == o.active_reset_srtp_params &&
crypto_options == o.crypto_options &&
offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
turn_logging_id == o.turn_logging_id &&
enable_implicit_rollback == o.enable_implicit_rollback &&
allow_codec_switching == o.allow_codec_switching &&
report_usage_pattern_delay_ms == o.report_usage_pattern_delay_ms &&
stable_writable_connection_ping_interval_ms ==
o.stable_writable_connection_ping_interval_ms &&
vpn_preference == o.vpn_preference && vpn_list == o.vpn_list &&
port_allocator_config.min_port == o.port_allocator_config.min_port &&
port_allocator_config.max_port == o.port_allocator_config.max_port &&
port_allocator_config.flags == o.port_allocator_config.flags;
}
bool PeerConnectionInterface::RTCConfiguration::operator!=(
const PeerConnectionInterface::RTCConfiguration& o) const {
return !(*this == o);
}
RTCErrorOr<rtc::scoped_refptr<PeerConnection>> PeerConnection::Create(
rtc::scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call,
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
// TODO(https://crbug.com/webrtc/13528): Remove support for kPlanB.
if (configuration.sdp_semantics == SdpSemantics::kPlanB_DEPRECATED) {
RTC_LOG(LS_WARNING)
<< "PeerConnection constructed with legacy SDP semantics!";
}
RTCError config_error = cricket::P2PTransportChannel::ValidateIceConfig(
ParseIceConfig(configuration));
if (!config_error.ok()) {
RTC_LOG(LS_ERROR) << "Invalid ICE configuration: "
<< config_error.message();
return config_error;
}
if (!dependencies.allocator) {
RTC_LOG(LS_ERROR)
<< "PeerConnection initialized without a PortAllocator? "
"This shouldn't happen if using PeerConnectionFactory.";
return RTCError(
RTCErrorType::INVALID_PARAMETER,
"Attempt to create a PeerConnection without a PortAllocatorFactory");
}
if (!dependencies.observer) {
// TODO(deadbeef): Why do we do this?
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
"PeerConnectionObserver";
return RTCError(RTCErrorType::INVALID_PARAMETER,
"Attempt to create a PeerConnection without an observer");
}
bool is_unified_plan =
configuration.sdp_semantics == SdpSemantics::kUnifiedPlan;
bool dtls_enabled = DtlsEnabled(configuration, options, dependencies);
// Interim code: If an AsyncResolverFactory is given, but not an
// AsyncDnsResolverFactory, wrap it in a WrappingAsyncDnsResolverFactory
// If neither is given, create a WrappingAsyncDnsResolverFactory wrapping
// a BasicAsyncResolver.
// TODO(bugs.webrtc.org/12598): Remove code once all callers pass a
// AsyncDnsResolverFactory.
if (dependencies.async_dns_resolver_factory &&
dependencies.async_resolver_factory) {
RTC_LOG(LS_ERROR)
<< "Attempt to set both old and new type of DNS resolver factory";
return RTCError(RTCErrorType::INVALID_PARAMETER,
"Both old and new type of DNS resolver given");
}
if (dependencies.async_resolver_factory) {
dependencies.async_dns_resolver_factory =
std::make_unique<WrappingAsyncDnsResolverFactory>(
std::move(dependencies.async_resolver_factory));
} else {
dependencies.async_dns_resolver_factory =
std::make_unique<WrappingAsyncDnsResolverFactory>(
std::make_unique<BasicAsyncResolverFactory>());
}
// The PeerConnection constructor consumes some, but not all, dependencies.
auto pc = rtc::make_ref_counted<PeerConnection>(
context, options, is_unified_plan, std::move(event_log), std::move(call),
dependencies, dtls_enabled);
RTCError init_error = pc->Initialize(configuration, std::move(dependencies));
if (!init_error.ok()) {
RTC_LOG(LS_ERROR) << "PeerConnection initialization failed";
return init_error;
}
return pc;
}
PeerConnection::PeerConnection(
rtc::scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
bool is_unified_plan,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call,
PeerConnectionDependencies& dependencies,
bool dtls_enabled)
: context_(context),
trials_(std::move(dependencies.trials), &context->field_trials()),
options_(options),
observer_(dependencies.observer),
is_unified_plan_(is_unified_plan),
event_log_(std::move(event_log)),
event_log_ptr_(event_log_.get()),
async_dns_resolver_factory_(
std::move(dependencies.async_dns_resolver_factory)),
port_allocator_(std::move(dependencies.allocator)),
ice_transport_factory_(std::move(dependencies.ice_transport_factory)),
tls_cert_verifier_(std::move(dependencies.tls_cert_verifier)),
call_(std::move(call)),
call_ptr_(call_.get()),
// RFC 3264: The numeric value of the session id and version in the
// o line MUST be representable with a "64 bit signed integer".
// Due to this constraint session id `session_id_` is max limited to
// LLONG_MAX.
session_id_(rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX)),
dtls_enabled_(dtls_enabled),
data_channel_controller_(this),
message_handler_(signaling_thread()),
weak_factory_(this) {
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
worker_thread_safety_ = PendingTaskSafetyFlag::Create();
if (!call_)
worker_thread_safety_->SetNotAlive();
});
}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK_RUN_ON(signaling_thread());
if (sdp_handler_) {
sdp_handler_->PrepareForShutdown();
}
// Need to stop transceivers before destroying the stats collector because
// AudioRtpSender has a reference to the LegacyStatsCollector it will update
// when stopping.
if (rtp_manager()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
transceiver->StopInternal();
}
}
legacy_stats_.reset(nullptr);
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
stats_collector_ = nullptr;
}
if (sdp_handler_) {
// Don't destroy BaseChannels until after stats has been cleaned up so that
// the last stats request can still read from the channels.
sdp_handler_->DestroyAllChannels();
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
sdp_handler_->ResetSessionDescFactory();
}
// port_allocator_ and transport_controller_ live on the network thread and
// should be destroyed there.
transport_controller_copy_ = nullptr;
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread());
TeardownDataChannelTransport_n();
transport_controller_.reset();
port_allocator_.reset();
if (network_thread_safety_)
network_thread_safety_->SetNotAlive();
});
// call_ and event_log_ must be destroyed on the worker thread.
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
worker_thread_safety_->SetNotAlive();
call_.reset();
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
});
}
RTCError PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
RTCErrorType parse_error =
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
if (parse_error != RTCErrorType::NONE) {
return RTCError(parse_error, "ICE server parse failed");
}
// Add the turn logging id to all turn servers
for (cricket::RelayServerConfig& turn_server : turn_servers) {
turn_server.turn_logging_id = configuration.turn_logging_id;
}
// Note if STUN or TURN servers were supplied.
if (!stun_servers.empty()) {
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
// Network thread initialization.
transport_controller_copy_ =
network_thread()->Invoke<JsepTransportController*>(RTC_FROM_HERE, [&] {
RTC_DCHECK_RUN_ON(network_thread());
network_thread_safety_ = PendingTaskSafetyFlag::Create();
InitializePortAllocatorResult pa_result = InitializePortAllocator_n(
stun_servers, turn_servers, configuration);
// Send information about IPv4/IPv6 status.
PeerConnectionAddressFamilyCounter address_family =
pa_result.enable_ipv6 ? kPeerConnection_IPv6 : kPeerConnection_IPv4;
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
address_family,
kPeerConnectionAddressFamilyCounter_Max);
return InitializeTransportController_n(configuration, dependencies);
});
configuration_ = configuration;
legacy_stats_ = std::make_unique<LegacyStatsCollector>(this);
stats_collector_ = RTCStatsCollector::Create(this);
sdp_handler_ = SdpOfferAnswerHandler::Create(this, configuration,
dependencies, context_.get());
rtp_manager_ = std::make_unique<RtpTransmissionManager>(
IsUnifiedPlan(), context_.get(), &usage_pattern_, observer_,
legacy_stats_.get(), [this]() {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->UpdateNegotiationNeeded();
});
// Add default audio/video transceivers for Plan B SDP.
if (!IsUnifiedPlan()) {
rtp_manager()->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), rtc::make_ref_counted<RtpTransceiver>(
cricket::MEDIA_TYPE_AUDIO, context())));
rtp_manager()->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), rtc::make_ref_counted<RtpTransceiver>(
cricket::MEDIA_TYPE_VIDEO, context())));
}
int delay_ms = configuration.report_usage_pattern_delay_ms
? *configuration.report_usage_pattern_delay_ms
: REPORT_USAGE_PATTERN_DELAY_MS;
message_handler_.RequestUsagePatternReport(
[this]() {
RTC_DCHECK_RUN_ON(signaling_thread());
ReportUsagePattern();
},
delay_ms);
// Record the number of configured ICE servers for all connections.
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.PeerConnection.IceServers.Configured",
configuration_.servers.size(), 0, 31, 32);
return RTCError::OK();
}
JsepTransportController* PeerConnection::InitializeTransportController_n(
const RTCConfiguration& configuration,
const PeerConnectionDependencies& dependencies) {
JsepTransportController::Config config;
config.redetermine_role_on_ice_restart =
configuration.redetermine_role_on_ice_restart;
config.ssl_max_version = options_.ssl_max_version;
config.disable_encryption = options_.disable_encryption;
config.bundle_policy = configuration.bundle_policy;
config.rtcp_mux_policy = configuration.rtcp_mux_policy;
// TODO(bugs.webrtc.org/9891) - Remove options_.crypto_options then remove
// this stub.
config.crypto_options = configuration.crypto_options.has_value()
? *configuration.crypto_options
: options_.crypto_options;
config.transport_observer = this;
config.rtcp_handler = InitializeRtcpCallback();
config.event_log = event_log_ptr_;
#if defined(ENABLE_EXTERNAL_AUTH)
config.enable_external_auth = true;
#endif
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
// DTLS has to be enabled to use SCTP.
if (dtls_enabled_) {
config.sctp_factory = context_->sctp_transport_factory();
}
config.ice_transport_factory = ice_transport_factory_.get();
config.on_dtls_handshake_error_ =
[weak_ptr = weak_factory_.GetWeakPtr()](rtc::SSLHandshakeError s) {
if (weak_ptr) {
weak_ptr->OnTransportControllerDtlsHandshakeError(s);
}
};
config.field_trials = trials_.get();
transport_controller_.reset(
new JsepTransportController(network_thread(), port_allocator_.get(),
async_dns_resolver_factory_.get(), config));
transport_controller_->SubscribeIceConnectionState(
[this](cricket::IceConnectionState s) {
RTC_DCHECK_RUN_ON(network_thread());
if (s == cricket::kIceConnectionConnected) {
ReportTransportStats();
}
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerConnectionState(s);
}));
});
transport_controller_->SubscribeConnectionState(
[this](PeerConnectionInterface::PeerConnectionState s) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
SetConnectionState(s);
}));
});
transport_controller_->SubscribeStandardizedIceConnectionState(
[this](PeerConnectionInterface::IceConnectionState s) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
SetStandardizedIceConnectionState(s);
}));
});
transport_controller_->SubscribeIceGatheringState(
[this](cricket::IceGatheringState s) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerGatheringState(s);
}));
});
transport_controller_->SubscribeIceCandidateGathered(
[this](const std::string& transport,
const std::vector<cricket::Candidate>& candidates) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(),
[this, t = transport, c = candidates]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidatesGathered(t, c);
}));
});
transport_controller_->SubscribeIceCandidateError(
[this](const cricket::IceCandidateErrorEvent& event) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, event = event]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidateError(event);
}));
});
transport_controller_->SubscribeIceCandidatesRemoved(
[this](const std::vector<cricket::Candidate>& c) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, c = c]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidatesRemoved(c);
}));
});
transport_controller_->SubscribeIceCandidatePairChanged(
[this](const cricket::CandidatePairChangeEvent& event) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, event = event]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidateChanged(event);
}));
});
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
return transport_controller_.get();
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
"Plan SdpSemantics. Please use GetSenders "
"instead.";
return sdp_handler_->local_streams();
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
"Plan SdpSemantics. Please use GetReceivers "
"instead.";
return sdp_handler_->remote_streams();
}
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
"SdpSemantics. Please use AddTrack instead.";
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
if (!ConfiguredForMedia()) {
RTC_LOG(LS_ERROR) << "AddStream: Not configured for media";
return false;
}
return sdp_handler_->AddStream(local_stream);
}
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(ConfiguredForMedia());
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
"Plan SdpSemantics. Please use RemoveTrack "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
sdp_handler_->RemoveStream(local_stream);
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
}
if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
track->kind() == MediaStreamTrackInterface::kVideoKind)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track has invalid kind: " + track->kind());
}
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (rtp_manager()->FindSenderForTrack(track.get())) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Sender already exists for track " + track->id() + ".");
}
auto sender_or_error = rtp_manager()->AddTrack(track, stream_ids);
if (sender_or_error.ok()) {
sdp_handler_->UpdateNegotiationNeeded();
legacy_stats_->AddTrack(track.get());
}
return sender_or_error;
}
RTCError PeerConnection::RemoveTrackOrError(
rtc::scoped_refptr<RtpSenderInterface> sender) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
if (!sender) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
}
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (IsUnifiedPlan()) {
auto transceiver = FindTransceiverBySender(sender);
if (!transceiver || !sender->track()) {
return RTCError::OK();
}
sender->SetTrack(nullptr);
if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kRecvOnly);
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kInactive);
}
} else {
bool removed;
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
removed = rtp_manager()->GetAudioTransceiver()->internal()->RemoveSender(
sender.get());
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
removed = rtp_manager()->GetVideoTransceiver()->internal()->RemoveSender(
sender.get());
}
if (!removed) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Couldn't find sender " + sender->id() + " to remove.");
}
}
sdp_handler_->UpdateNegotiationNeeded();
return RTCError::OK();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindTransceiverBySender(
rtc::scoped_refptr<RtpSenderInterface> sender) {
return rtp_manager()->transceivers()->FindBySender(sender);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
return AddTransceiver(track, RtpTransceiverInit());
}
RtpTransportInternal* PeerConnection::GetRtpTransport(const std::string& mid) {
// TODO(bugs.webrtc.org/9987): Avoid the thread jump.
// This might be done by caching the value on the signaling thread.
RTC_DCHECK_RUN_ON(signaling_thread());
return network_thread()->Invoke<RtpTransportInternal*>(
RTC_FROM_HERE, [this, &mid] {
RTC_DCHECK_RUN_ON(network_thread());
auto rtp_transport = transport_controller_->GetRtpTransport(mid);
RTC_DCHECK(rtp_transport);
return rtp_transport;
});
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
RTC_CHECK(IsUnifiedPlan())
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
}
cricket::MediaType media_type;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
media_type = cricket::MEDIA_TYPE_AUDIO;
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
media_type = cricket::MEDIA_TYPE_VIDEO;
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track kind is not audio or video");
}
return AddTransceiver(media_type, track, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
return AddTransceiver(media_type, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
RTC_CHECK(IsUnifiedPlan())
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"media type is not audio or video");
}
return AddTransceiver(media_type, nullptr, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool update_negotiation_needed) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO));
if (track) {
RTC_DCHECK_EQ(media_type,
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO));
}
RTC_HISTOGRAM_COUNTS_LINEAR(kSimulcastNumberOfEncodings,
init.send_encodings.size(), 0, 7, 8);
size_t num_rids = absl::c_count_if(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !encoding.rid.empty();
});
if (num_rids > 0 && num_rids != init.send_encodings.size()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"RIDs must be provided for either all or none of the send encodings.");
}
if (num_rids > 0 && absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !IsLegalRsidName(encoding.rid);
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Invalid RID value provided.");
}
if (absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return encoding.ssrc.has_value();
})) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
RtpParameters parameters;
parameters.encodings = init.send_encodings;
// Encodings are dropped from the tail if too many are provided.
size_t max_simulcast_streams =
media_type == cricket::MEDIA_TYPE_VIDEO ? kMaxSimulcastStreams : 1u;
if (parameters.encodings.size() > max_simulcast_streams) {
parameters.encodings.erase(
parameters.encodings.begin() + max_simulcast_streams,
parameters.encodings.end());
}
// Single RID should be removed.
if (parameters.encodings.size() == 1 &&
!parameters.encodings[0].rid.empty()) {
RTC_LOG(LS_INFO) << "Removing RID: " << parameters.encodings[0].rid << ".";
parameters.encodings[0].rid.clear();
}
// If RIDs were not provided, they are generated for simulcast scenario.
if (parameters.encodings.size() > 1 && num_rids == 0) {
rtc::UniqueStringGenerator rid_generator;
for (RtpEncodingParameters& encoding : parameters.encodings) {
encoding.rid = rid_generator();
}
}
if (UnimplementedRtpParameterHasValue(parameters)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
auto result = cricket::CheckRtpParametersValues(parameters);
if (!result.ok()) {
LOG_AND_RETURN_ERROR(result.type(), result.message());
}
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTransceiver.";
// Set the sender ID equal to the track ID if the track is specified unless
// that sender ID is already in use.
std::string sender_id = (track && !rtp_manager()->FindSenderById(track->id())
? track->id()
: rtc::CreateRandomUuid());
auto sender = rtp_manager()->CreateSender(
media_type, sender_id, track, init.stream_ids, parameters.encodings);
auto receiver =
rtp_manager()->CreateReceiver(media_type, rtc::CreateRandomUuid());
auto transceiver = rtp_manager()->CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_direction(init.direction);
if (update_negotiation_needed) {
sdp_handler_->UpdateNegotiationNeeded();
}
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
}
void PeerConnection::OnNegotiationNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsClosed());
sdp_handler_->UpdateNegotiationNeeded();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
const std::string& kind,
const std::string& stream_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
RTC_LOG(LS_ERROR) << "Not configured for media";
return nullptr;
}
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
"Plan SdpSemantics. Please use AddTransceiver "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
if (IsClosed()) {
return nullptr;
}
// Internally we need to have one stream with Plan B semantics, so we
// generate a random stream ID if not specified.
std::vector<std::string> stream_ids;
if (stream_id.empty()) {
stream_ids.push_back(rtc::CreateRandomUuid());
RTC_LOG(LS_INFO)
<< "No stream_id specified for sender. Generated stream ID: "
<< stream_ids[0];
} else {
stream_ids.push_back(stream_id);
}
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
auto audio_sender =
AudioRtpSender::Create(worker_thread(), rtc::CreateRandomUuid(),
legacy_stats_.get(), rtp_manager());
audio_sender->SetMediaChannel(rtp_manager()->voice_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), audio_sender);
rtp_manager()->GetAudioTransceiver()->internal()->AddSender(new_sender);
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
auto video_sender = VideoRtpSender::Create(
worker_thread(), rtc::CreateRandomUuid(), rtp_manager());
video_sender->SetMediaChannel(rtp_manager()->video_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), video_sender);
rtp_manager()->GetVideoTransceiver()->internal()->AddSender(new_sender);
} else {
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
return nullptr;
}
new_sender->internal()->set_stream_ids(stream_ids);
return new_sender;
}
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
if (ConfiguredForMedia()) {
for (const auto& sender : rtp_manager()->GetSendersInternal()) {
ret.push_back(sender);
}
}
return ret;
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
if (ConfiguredForMedia()) {
for (const auto& receiver : rtp_manager()->GetReceiversInternal()) {
ret.push_back(receiver);
}
}
return ret;
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::GetTransceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(IsUnifiedPlan())
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
if (ConfiguredForMedia()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
all_transceivers.push_back(transceiver);
}
}
return all_transceivers;
}
bool PeerConnection::GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats (legacy)");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!observer) {
RTC_LOG(LS_ERROR) << "Legacy GetStats - observer is NULL.";
return false;
}
RTC_LOG_THREAD_BLOCK_COUNT();
legacy_stats_->UpdateStats(level);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(4);
// The LegacyStatsCollector is used to tell if a track is valid because it may
// remember tracks that the PeerConnection previously removed.
if (track && !legacy_stats_->IsValidTrack(track->id())) {
RTC_LOG(LS_WARNING) << "Legacy GetStats is called with an invalid track: "
<< track->id();
return false;
}
message_handler_.PostGetStats(observer, legacy_stats_.get(), track);
return true;
}
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(stats_collector_);
RTC_DCHECK(callback);
RTC_LOG_THREAD_BLOCK_COUNT();
stats_collector_->GetStatsReport(
rtc::scoped_refptr<RTCStatsCollectorCallback>(callback));
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
void PeerConnection::GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_DCHECK(stats_collector_);
RTC_LOG_THREAD_BLOCK_COUNT();
rtc::scoped_refptr<RtpSenderInternal> internal_sender;
if (selector) {
for (const auto& proxy_transceiver :
rtp_manager()->transceivers()->List()) {
for (const auto& proxy_sender :
proxy_transceiver->internal()->senders()) {
if (proxy_sender == selector) {
internal_sender = proxy_sender->internal();
break;
}
}
if (internal_sender)
break;
}
}
// If there is no `internal_sender` then `selector` is either null or does not
// belong to the PeerConnection (in Plan B, senders can be removed from the
// PeerConnection). This means that "all the stats objects representing the
// selector" is an empty set. Invoking GetStatsReport() with a null selector
// produces an empty stats report.
stats_collector_->GetStatsReport(internal_sender, callback);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
void PeerConnection::GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_DCHECK(stats_collector_);
RTC_LOG_THREAD_BLOCK_COUNT();
rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
if (selector) {
for (const auto& proxy_transceiver :
rtp_manager()->transceivers()->List()) {
for (const auto& proxy_receiver :
proxy_transceiver->internal()->receivers()) {
if (proxy_receiver == selector) {
internal_receiver = proxy_receiver->internal();
break;
}
}
if (internal_receiver)
break;
}
}
// If there is no `internal_receiver` then `selector` is either null or does
// not belong to the PeerConnection (in Plan B, receivers can be removed from
// the PeerConnection). This means that "all the stats objects representing
// the selector" is an empty set. Invoking GetStatsReport() with a null
// selector produces an empty stats report.
stats_collector_->GetStatsReport(internal_receiver, callback);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->signaling_state();
}
PeerConnectionInterface::IceConnectionState
PeerConnection::ice_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return ice_connection_state_;
}
PeerConnectionInterface::IceConnectionState
PeerConnection::standardized_ice_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return standardized_ice_connection_state_;
}
PeerConnectionInterface::PeerConnectionState
PeerConnection::peer_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return connection_state_;
}
PeerConnectionInterface::IceGatheringState
PeerConnection::ice_gathering_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return ice_gathering_state_;
}
absl::optional<bool> PeerConnection::can_trickle_ice_candidates() {
RTC_DCHECK_RUN_ON(signaling_thread());
const SessionDescriptionInterface* description = current_remote_description();
if (!description) {
description = pending_remote_description();
}
if (!description) {
return absl::nullopt;
}
// TODO(bugs.webrtc.org/7443): Change to retrieve from session-level option.
if (description->description()->transport_infos().size() < 1) {
return absl::nullopt;
}
return description->description()->transport_infos()[0].description.HasOption(
"trickle");
}
RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
PeerConnection::CreateDataChannelOrError(const std::string& label,
const DataChannelInit* config) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
bool first_datachannel = !data_channel_controller_.HasDataChannels();
std::unique_ptr<InternalDataChannelInit> internal_config;
if (config) {
internal_config.reset(new InternalDataChannelInit(*config));
}
// TODO(bugs.webrtc.org/12796): Return a more specific error.
rtc::scoped_refptr<DataChannelInterface> channel(
data_channel_controller_.InternalCreateDataChannelWithProxy(
label, internal_config.get()));
if (!channel.get()) {
return RTCError(RTCErrorType::INTERNAL_ERROR,
"Data channel creation failed");
}
// Trigger the onRenegotiationNeeded event for
// the first SCTP DataChannel.
if (first_datachannel) {
sdp_handler_->UpdateNegotiationNeeded();
}
NoteUsageEvent(UsageEvent::DATA_ADDED);
return channel;
}
void PeerConnection::RestartIce() {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->RestartIce();
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->CreateOffer(observer, options);
}
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->CreateAnswer(observer, options);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer, desc_ptr);
}
void PeerConnection::SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(std::move(desc), observer);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer);
}
void PeerConnection::SetLocalDescription(
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer);
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetRemoteDescription(observer, desc_ptr);
}
void PeerConnection::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetRemoteDescription(std::move(desc), observer);
}
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
RTC_DCHECK_RUN_ON(signaling_thread());
return configuration_;
}
RTCError PeerConnection::SetConfiguration(
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"SetConfiguration: PeerConnection is closed.");
}
// According to JSEP, after setLocalDescription, changing the candidate pool
// size is not allowed, and changing the set of ICE servers will not result
// in new candidates being gathered.
if (local_description() && configuration.ice_candidate_pool_size !=
configuration_.ice_candidate_pool_size) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Can't change candidate pool size after calling "
"SetLocalDescription.");
}
if (local_description() &&
configuration.crypto_options != configuration_.crypto_options) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Can't change crypto_options after calling "
"SetLocalDescription.");
}
// The simplest (and most future-compatible) way to tell if the config was
// modified in an invalid way is to copy each property we do support
// modifying, then use operator==. There are far more properties we don't
// support modifying than those we do, and more could be added.
RTCConfiguration modified_config = configuration_;
modified_config.servers = configuration.servers;
modified_config.type = configuration.type;
modified_config.ice_candidate_pool_size =
configuration.ice_candidate_pool_size;
modified_config.prune_turn_ports = configuration.prune_turn_ports;
modified_config.turn_port_prune_policy = configuration.turn_port_prune_policy;
modified_config.surface_ice_candidates_on_ice_transport_type_changed =
configuration.surface_ice_candidates_on_ice_transport_type_changed;
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
modified_config.ice_check_interval_strong_connectivity =
configuration.ice_check_interval_strong_connectivity;
modified_config.ice_check_interval_weak_connectivity =
configuration.ice_check_interval_weak_connectivity;
modified_config.ice_unwritable_timeout = configuration.ice_unwritable_timeout;
modified_config.ice_unwritable_min_checks =
configuration.ice_unwritable_min_checks;
modified_config.ice_inactive_timeout = configuration.ice_inactive_timeout;
modified_config.stun_candidate_keepalive_interval =
configuration.stun_candidate_keepalive_interval;
modified_config.turn_customizer = configuration.turn_customizer;
modified_config.network_preference = configuration.network_preference;
modified_config.active_reset_srtp_params =
configuration.active_reset_srtp_params;
modified_config.turn_logging_id = configuration.turn_logging_id;
modified_config.allow_codec_switching = configuration.allow_codec_switching;
modified_config.stable_writable_connection_ping_interval_ms =
configuration.stable_writable_connection_ping_interval_ms;
if (configuration != modified_config) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Modifying the configuration in an unsupported way.");
}
// Validate the modified configuration.
RTCError validate_error = ValidateConfiguration(modified_config);
if (!validate_error.ok()) {
return validate_error;
}
// Note that this isn't possible through chromium, since it's an unsigned
// short in WebIDL.
if (configuration.ice_candidate_pool_size < 0 ||
configuration.ice_candidate_pool_size > static_cast<int>(UINT16_MAX)) {
return RTCError(RTCErrorType::INVALID_RANGE);
}
// Parse ICE servers before hopping to network thread.
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
RTCErrorType parse_error =
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
if (parse_error != RTCErrorType::NONE) {
return RTCError(parse_error, "ICE server parse failed");
}
// Add the turn logging id to all turn servers
for (cricket::RelayServerConfig& turn_server : turn_servers) {
turn_server.turn_logging_id = configuration.turn_logging_id;
}
// Note if STUN or TURN servers were supplied.
if (!stun_servers.empty()) {
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
const bool has_local_description = local_description() != nullptr;
const bool needs_ice_restart =
modified_config.servers != configuration_.servers ||
NeedIceRestart(
configuration_.surface_ice_candidates_on_ice_transport_type_changed,
configuration_.type, modified_config.type) ||
modified_config.GetTurnPortPrunePolicy() !=
configuration_.GetTurnPortPrunePolicy();
cricket::IceConfig ice_config = ParseIceConfig(modified_config);
// Apply part of the configuration on the network thread. In theory this
// shouldn't fail.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
[this, needs_ice_restart, &ice_config, &stun_servers, &turn_servers,
&modified_config, has_local_description] {
RTC_DCHECK_RUN_ON(network_thread());
// As described in JSEP, calling setConfiguration with new ICE
// servers or candidate policy must set a "needs-ice-restart" bit so
// that the next offer triggers an ICE restart which will pick up
// the changes.
if (needs_ice_restart)
transport_controller_->SetNeedsIceRestartFlag();
transport_controller_->SetIceConfig(ice_config);
return ReconfigurePortAllocator_n(
stun_servers, turn_servers, modified_config.type,
modified_config.ice_candidate_pool_size,
modified_config.GetTurnPortPrunePolicy(),
modified_config.turn_customizer,
modified_config.stun_candidate_keepalive_interval,
has_local_description);
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to apply configuration to PortAllocator.");
}
if (configuration_.active_reset_srtp_params !=
modified_config.active_reset_srtp_params) {
// TODO(tommi): merge invokes
network_thread()->Invoke<void>(RTC_FROM_HERE, [this, &modified_config] {
RTC_DCHECK_RUN_ON(network_thread());
transport_controller_->SetActiveResetSrtpParams(
modified_config.active_reset_srtp_params);
});
}
if (modified_config.allow_codec_switching.has_value()) {
std::vector<cricket::VideoMediaChannel*> channels;
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
if (transceiver->media_type() != cricket::MEDIA_TYPE_VIDEO)
continue;
auto* video_channel = transceiver->internal()->channel();
if (video_channel)
channels.push_back(static_cast<cricket::VideoMediaChannel*>(
video_channel->media_channel()));
}
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
[channels = std::move(channels),
allow_codec_switching = *modified_config.allow_codec_switching]() {
for (auto* ch : channels)
ch->SetVideoCodecSwitchingEnabled(allow_codec_switching);
});
}
configuration_ = modified_config;
return RTCError::OK();
}
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
ClearStatsCache();
return sdp_handler_->AddIceCandidate(ice_candidate);
}
void PeerConnection::AddIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate,
std::function<void(RTCError)> callback) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->AddIceCandidate(std::move(candidate),
[this, callback](webrtc::RTCError result) {
ClearStatsCache();
callback(result);
});
}
bool PeerConnection::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->RemoveIceCandidates(candidates);
}
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<RTCError>(
RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
}
RTC_DCHECK_RUN_ON(worker_thread());
const bool has_min = bitrate.min_bitrate_bps.has_value();
const bool has_start = bitrate.start_bitrate_bps.has_value();
const bool has_max = bitrate.max_bitrate_bps.has_value();
if (has_min && *bitrate.min_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"min_bitrate_bps <= 0");
}
if (has_start) {
if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"start_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.start_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"curent_bitrate_bps < 0");
}
}
if (has_max) {
if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < start_bitrate_bps");
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.max_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < 0");
}
}
RTC_DCHECK(call_.get());
call_->SetClientBitratePreferences(bitrate);
return RTCError::OK();
}
void PeerConnection::SetAudioPlayout(bool playout) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE, [this, playout] { SetAudioPlayout(playout); });
return;
}
auto audio_state = context_->media_engine()->voice().GetAudioState();
audio_state->SetPlayout(playout);
}
void PeerConnection::SetAudioRecording(bool recording) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE, [this, recording] { SetAudioRecording(recording); });
return;
}
auto audio_state = context_->media_engine()->voice().GetAudioState();
audio_state->SetRecording(recording);
}
void PeerConnection::AddAdaptationResource(
rtc::scoped_refptr<Resource> resource) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<void>(RTC_FROM_HERE, [this, resource]() {
return AddAdaptationResource(resource);
});
}
RTC_DCHECK_RUN_ON(worker_thread());
if (!call_) {
// The PeerConnection has been closed.
return;
}
call_->AddAdaptationResource(resource);
}
bool PeerConnection::ConfiguredForMedia() const {
return context_->media_engine();
}
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
return worker_thread()->Invoke<bool>(
RTC_FROM_HERE,
[this, output = std::move(output), output_period_ms]() mutable {
return StartRtcEventLog_w(std::move(output), output_period_ms);
});
}
bool PeerConnection::StartRtcEventLog(
std::unique_ptr<RtcEventLogOutput> output) {
int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
if (trials().IsEnabled("WebRTC-RtcEventLogNewFormat")) {
output_period_ms = 5000;
}
return StartRtcEventLog(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog() {
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] { StopRtcEventLog_w(); });
}
rtc::scoped_refptr<DtlsTransportInterface>
PeerConnection::LookupDtlsTransportByMid(const std::string& mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
}
rtc::scoped_refptr<DtlsTransport>
PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
// TODO(bugs.webrtc.org/9987): Avoid the thread jump.
// This might be done by caching the value on the signaling thread.
return network_thread()->Invoke<rtc::scoped_refptr<DtlsTransport>>(
RTC_FROM_HERE, [this, mid]() {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
});
}
rtc::scoped_refptr<SctpTransportInterface> PeerConnection::GetSctpTransport()
const {
RTC_DCHECK_RUN_ON(network_thread());
if (!sctp_mid_n_)
return nullptr;
return transport_controller_->GetSctpTransport(*sctp_mid_n_);
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->local_description();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->remote_description();
}
const SessionDescriptionInterface* PeerConnection::current_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->current_local_description();
}
const SessionDescriptionInterface* PeerConnection::current_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->current_remote_description();
}
const SessionDescriptionInterface* PeerConnection::pending_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->pending_local_description();
}
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->pending_remote_description();
}
void PeerConnection::Close() {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::Close");
RTC_LOG_THREAD_BLOCK_COUNT();
if (IsClosed()) {
return;
}
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
legacy_stats_->UpdateStats(kStatsOutputLevelStandard);
ice_connection_state_ = PeerConnectionInterface::kIceConnectionClosed;
Observer()->OnIceConnectionChange(ice_connection_state_);
standardized_ice_connection_state_ =
PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
Observer()->OnConnectionChange(connection_state_);
sdp_handler_->Close();
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
if (ConfiguredForMedia()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
transceiver->internal()->SetPeerConnectionClosed();
if (!transceiver->stopped())
transceiver->StopInternal();
}
}
// Ensure that all asynchronous stats requests are completed before destroying
// the transport controller below.
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
}
// Don't destroy BaseChannels until after stats has been cleaned up so that
// the last stats request can still read from the channels.
sdp_handler_->DestroyAllChannels();
// The event log is used in the transport controller, which must be outlived
// by the former. CreateOffer by the peer connection is implemented
// asynchronously and if the peer connection is closed without resetting the
// WebRTC session description factory, the session description factory would
// call the transport controller.
sdp_handler_->ResetSessionDescFactory();
if (ConfiguredForMedia()) {
rtp_manager_->Close();
}
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
// Data channels will already have been unset via the DestroyAllChannels()
// call above, which triggers a call to TeardownDataChannelTransport_n().
// TODO(tommi): ^^ That's not exactly optimal since this is yet another
// blocking hop to the network thread during Close(). Further still, the
// voice/video/data channels will be cleared on the worker thread.
RTC_DCHECK_RUN_ON(network_thread());
transport_controller_.reset();
port_allocator_->DiscardCandidatePool();
if (network_thread_safety_) {
network_thread_safety_->SetNotAlive();
}
});
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
worker_thread_safety_->SetNotAlive();
call_.reset();
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
});
ReportUsagePattern();
// The .h file says that observer can be discarded after close() returns.
// Make sure this is true.
observer_ = nullptr;
// Signal shutdown to the sdp handler. This invalidates weak pointers for
// internal pending callbacks.
sdp_handler_->PrepareForShutdown();
}
void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (ice_connection_state_ == new_state) {
return;
}
// After transitioning to "closed", ignore any additional states from
// TransportController (such as "disconnected").
if (IsClosed()) {
return;
}
RTC_LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
<< " => " << new_state;
RTC_DCHECK(ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionClosed);
ice_connection_state_ = new_state;
Observer()->OnIceConnectionChange(ice_connection_state_);
}
void PeerConnection::SetStandardizedIceConnectionState(
PeerConnectionInterface::IceConnectionState new_state) {
if (standardized_ice_connection_state_ == new_state) {
return;
}
if (IsClosed()) {
return;
}
RTC_LOG(LS_INFO) << "Changing standardized IceConnectionState "
<< standardized_ice_connection_state_ << " => " << new_state;
standardized_ice_connection_state_ = new_state;
Observer()->OnStandardizedIceConnectionChange(new_state);
}
void PeerConnection::SetConnectionState(
PeerConnectionInterface::PeerConnectionState new_state) {
if (connection_state_ == new_state)
return;
if (IsClosed())
return;
connection_state_ = new_state;
Observer()->OnConnectionChange(new_state);
// The first connection state change to connected happens once per
// connection which makes it a good point to report metrics.
if (new_state == PeerConnectionState::kConnected && !was_ever_connected_) {
was_ever_connected_ = true;
ReportFirstConnectUsageMetrics();
}
}
void PeerConnection::ReportFirstConnectUsageMetrics() {
// Record bundle-policy from configuration. Done here from
// connectionStateChange to limit to actually established connections.
BundlePolicyUsage policy = kBundlePolicyUsageMax;
switch (configuration_.bundle_policy) {
case kBundlePolicyBalanced:
policy = kBundlePolicyUsageBalanced;
break;
case kBundlePolicyMaxBundle:
policy = kBundlePolicyUsageMaxBundle;
break;
case kBundlePolicyMaxCompat:
policy = kBundlePolicyUsageMaxCompat;
break;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.BundlePolicy", policy,
kBundlePolicyUsageMax);
// Record configured ice candidate pool size depending on the
// BUNDLE policy. See
// https://w3c.github.io/webrtc-pc/#dom-rtcconfiguration-icecandidatepoolsize
// The ICE candidate pool size is an optimization and it may be desirable
// to restrict the maximum size of the pre-gathered candidates.
switch (configuration_.bundle_policy) {
case kBundlePolicyBalanced:
RTC_HISTOGRAM_COUNTS_LINEAR(
"WebRTC.PeerConnection.CandidatePoolUsage.Balanced",
configuration_.ice_candidate_pool_size, 0, 255, 256);
break;
case kBundlePolicyMaxBundle:
RTC_HISTOGRAM_COUNTS_LINEAR(
"WebRTC.PeerConnection.CandidatePoolUsage.MaxBundle",
configuration_.ice_candidate_pool_size, 0, 255, 256);
break;
case kBundlePolicyMaxCompat:
RTC_HISTOGRAM_COUNTS_LINEAR(
"WebRTC.PeerConnection.CandidatePoolUsage.MaxCompat",
configuration_.ice_candidate_pool_size, 0, 255, 256);
break;
}
// Record whether there was a local or remote provisional answer.
ProvisionalAnswerUsage pranswer = kProvisionalAnswerNotUsed;
if (local_description()->GetType() == SdpType::kPrAnswer) {
pranswer = kProvisionalAnswerLocal;
} else if (remote_description()->GetType() == SdpType::kPrAnswer) {
pranswer = kProvisionalAnswerRemote;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.ProvisionalAnswer", pranswer,
kProvisionalAnswerMax);
// Record the number of configured ICE servers for connected connections.
RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.PeerConnection.IceServers.Connected",
configuration_.servers.size(), 0, 31, 32);
// Record the number of valid / invalid ice-ufrag. We do allow certain
// non-spec ice-char for backward-compat reasons. At this point we know
// that the ufrag/pwd consists of a valid ice-char or one of the four
// not allowed characters since we have passed the IsIceChar check done
// by the p2p transport description on setRemoteDescription calls.
auto transport_infos = remote_description()->description()->transport_infos();
if (transport_infos.size() > 0) {
auto ice_parameters = transport_infos[0].description.GetIceParameters();
auto is_invalid_char = [](char c) {
return c == '-' || c == '=' || c == '#' || c == '_';
};
bool isUsingInvalidIceCharInUfrag =
absl::c_any_of(ice_parameters.ufrag, is_invalid_char);
bool isUsingInvalidIceCharInPwd =
absl::c_any_of(ice_parameters.pwd, is_invalid_char);
RTC_HISTOGRAM_BOOLEAN(
"WebRTC.PeerConnection.ValidIceChars",
!(isUsingInvalidIceCharInUfrag || isUsingInvalidIceCharInPwd));
}
}
void PeerConnection::OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
if (IsClosed()) {
return;
}
ice_gathering_state_ = new_state;
Observer()->OnIceGatheringChange(ice_gathering_state_);
}
void PeerConnection::OnIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate) {
if (IsClosed()) {
return;
}
ReportIceCandidateCollected(candidate->candidate());
ClearStatsCache();
Observer()->OnIceCandidate(candidate.get());
}
void PeerConnection::OnIceCandidateError(const std::string& address,
int port,
const std::string& url,
int error_code,
const std::string& error_text) {
if (IsClosed()) {
return;
}
Observer()->OnIceCandidateError(address, port, url, error_code, error_text);
}
void PeerConnection::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
if (IsClosed()) {
return;
}
Observer()->OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnSelectedCandidatePairChanged(
const cricket::CandidatePairChangeEvent& event) {
if (IsClosed()) {
return;
}
if (event.selected_candidate_pair.local_candidate().type() ==
LOCAL_PORT_TYPE &&
event.selected_candidate_pair.remote_candidate().type() ==
LOCAL_PORT_TYPE) {
NoteUsageEvent(UsageEvent::DIRECT_CONNECTION_SELECTED);
}
Observer()->OnIceSelectedCandidatePairChanged(event);
}
absl::optional<std::string> PeerConnection::GetDataMid() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sctp_mid_s_;
}
void PeerConnection::SetSctpDataMid(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
sctp_mid_s_ = mid;
}
void PeerConnection::ResetSctpDataMid() {
RTC_DCHECK_RUN_ON(signaling_thread());
sctp_mid_s_.reset();
sctp_transport_name_s_.clear();
}
void PeerConnection::OnSctpDataChannelClosed(DataChannelInterface* channel) {
// Since data_channel_controller doesn't do signals, this
// signal is relayed here.
data_channel_controller_.OnSctpDataChannelClosed(
static_cast<SctpDataChannel*>(channel));
}
PeerConnection::InitializePortAllocatorResult
PeerConnection::InitializePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_->Initialize();
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
int port_allocator_flags = port_allocator_->flags();
port_allocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
cricket::PORTALLOCATOR_ENABLE_IPV6 |
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
// If the disable-IPv6 flag was specified, we'll not override it
// by experiment.
if (configuration.disable_ipv6) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
} else if (trials().IsDisabled("WebRTC-IPv6Default")) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.disable_ipv6_on_wifi) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
}
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
}
if (configuration.candidate_network_policy ==
kCandidateNetworkPolicyLowCost) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
}
if (configuration.disable_link_local_networks) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS;
RTC_LOG(LS_INFO) << "Disable candidates on link-local network interfaces.";
}
port_allocator_->set_flags(port_allocator_flags);
// No step delay is used while allocating ports.
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
port_allocator_->SetCandidateFilter(
ConvertIceTransportTypeToCandidateFilter(configuration.type));
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
auto turn_servers_copy = turn_servers;
for (auto& turn_server : turn_servers_copy) {
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
}
// Call this last since it may create pooled allocator sessions using the
// properties set above.
port_allocator_->SetConfiguration(
stun_servers, std::move(turn_servers_copy),
configuration.ice_candidate_pool_size,
configuration.GetTurnPortPrunePolicy(), configuration.turn_customizer,
configuration.stun_candidate_keepalive_interval);
InitializePortAllocatorResult res;
res.enable_ipv6 = port_allocator_flags & cricket::PORTALLOCATOR_ENABLE_IPV6;
return res;
}
bool PeerConnection::ReconfigurePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
IceTransportsType type,
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
webrtc::TurnCustomizer* turn_customizer,
absl::optional<int> stun_candidate_keepalive_interval,
bool have_local_description) {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_->SetCandidateFilter(
ConvertIceTransportTypeToCandidateFilter(type));
// According to JSEP, after setLocalDescription, changing the candidate pool
// size is not allowed, and changing the set of ICE servers will not result
// in new candidates being gathered.
if (have_local_description) {
port_allocator_->FreezeCandidatePool();
}
// Add the custom tls turn servers if they exist.
auto turn_servers_copy = turn_servers;
for (auto& turn_server : turn_servers_copy) {
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
}
// Call this last since it may create pooled allocator sessions using the
// candidate filter set above.
return port_allocator_->SetConfiguration(
stun_servers, std::move(turn_servers_copy), candidate_pool_size,
turn_port_prune_policy, turn_customizer,
stun_candidate_keepalive_interval);
}
bool PeerConnection::StartRtcEventLog_w(
std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
RTC_DCHECK_RUN_ON(worker_thread());
if (!event_log_) {
return false;
}
return event_log_->StartLogging(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog_w() {
RTC_DCHECK_RUN_ON(worker_thread());
if (event_log_) {
event_log_->StopLogging();
}
}
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_LOG(LS_VERBOSE)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the SCTP transport.";
return false;
}
if (!data_channel_controller_.data_channel_transport()) {
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
"SSL Role of the SCTP transport.";
return false;
}
absl::optional<rtc::SSLRole> dtls_role;
if (sctp_mid_s_) {
dtls_role = network_thread()->Invoke<absl::optional<rtc::SSLRole>>(
RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->GetDtlsRole(*sctp_mid_n_);
});
if (!dtls_role && sdp_handler_->is_caller().has_value()) {
// This works fine if we are the offerer, but can be a mistake if
// we are the answerer and the remote offer is ACTIVE. In that
// case, we will guess the role wrong.
// TODO(bugs.webrtc.org/13668): Check if this actually happens.
RTC_LOG(LS_ERROR)
<< "Possible risk: DTLS role guesser is active, is_caller is "
<< *sdp_handler_->is_caller();
dtls_role =
*sdp_handler_->is_caller() ? rtc::SSL_SERVER : rtc::SSL_CLIENT;
}
if (dtls_role) {
*role = *dtls_role;
return true;
}
}
return false;
}
bool PeerConnection::GetSslRole(const std::string& content_name,
rtc::SSLRole* role) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the session.";
return false;
}
auto dtls_role = network_thread()->Invoke<absl::optional<rtc::SSLRole>>(
RTC_FROM_HERE, [this, content_name]() {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->GetDtlsRole(content_name);
});
if (dtls_role) {
*role = *dtls_role;
return true;
}
return false;
}
bool PeerConnection::GetTransportDescription(
const SessionDescription* description,
const std::string& content_name,
cricket::TransportDescription* tdesc) {
if (!description || !tdesc) {
return false;
}
const TransportInfo* transport_info =
description->GetTransportInfoByName(content_name);
if (!transport_info) {
return false;
}
*tdesc = transport_info->description;
return true;
}
std::vector<DataChannelStats> PeerConnection::GetDataChannelStats() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return data_channel_controller_.GetDataChannelStats();
}
absl::optional<std::string> PeerConnection::sctp_transport_name() const {
RTC_DCHECK_RUN_ON(signaling_thread());
if (sctp_mid_s_ && transport_controller_copy_)
return sctp_transport_name_s_;
return absl::optional<std::string>();
}
absl::optional<std::string> PeerConnection::sctp_mid() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sctp_mid_s_;
}
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive())
return {};
cricket::CandidateStatsList candidate_stats_list;
port_allocator_->GetCandidateStatsFromPooledSessions(&candidate_stats_list);
return candidate_stats_list;
}
std::map<std::string, cricket::TransportStats>
PeerConnection::GetTransportStatsByNames(
const std::set<std::string>& transport_names) {
TRACE_EVENT0("webrtc", "PeerConnection::GetTransportStatsByNames");
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive())
return {};
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, cricket::TransportStats> transport_stats_by_name;
for (const std::string& transport_name : transport_names) {
cricket::TransportStats transport_stats;
bool success =
transport_controller_->GetStats(transport_name, &transport_stats);
if (success) {
transport_stats_by_name[transport_name] = std::move(transport_stats);
} else {
RTC_LOG(LS_ERROR) << "Failed to get transport stats for transport_name="
<< transport_name;
}
}
return transport_stats_by_name;
}
bool PeerConnection::GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive() || !certificate) {
return false;
}
*certificate = transport_controller_->GetLocalCertificate(transport_name);
return *certificate != nullptr;
}
std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
const std::string& transport_name) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->GetRemoteSSLCertChain(transport_name);
}
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->IceRestartPending(content_name);
}
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
return network_thread()->Invoke<bool>(RTC_FROM_HERE, [this, &content_name] {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->NeedsIceRestart(content_name);
});
}
void PeerConnection::OnTransportControllerConnectionState(
cricket::IceConnectionState state) {
switch (state) {
case cricket::kIceConnectionConnecting:
// If the current state is Connected or Completed, then there were
// writable channels but now there are not, so the next state must
// be Disconnected.
// kIceConnectionConnecting is currently used as the default,
// un-connected state by the TransportController, so its only use is
// detecting disconnections.
if (ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionConnected ||
ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionCompleted) {
SetIceConnectionState(
PeerConnectionInterface::kIceConnectionDisconnected);
}
break;
case cricket::kIceConnectionFailed:
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
break;
case cricket::kIceConnectionConnected:
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
"all transports are writable.";
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
break;
case cricket::kIceConnectionCompleted:
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
"all transports are complete.";
if (ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionConnected) {
// If jumping directly from "checking" to "connected",
// signal "connected" first.
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
break;
default:
RTC_DCHECK_NOTREACHED();
}
}
void PeerConnection::OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const cricket::Candidates& candidates) {
// TODO(bugs.webrtc.org/12427): Expect this to come in on the network thread
// (not signaling as it currently does), handle appropriately.
int sdp_mline_index;
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
RTC_LOG(LS_ERROR)
<< "OnTransportControllerCandidatesGathered: content name "
<< transport_name << " not found";
return;
}
for (cricket::Candidates::const_iterator citer = candidates.begin();
citer != candidates.end(); ++citer) {
// Use transport_name as the candidate media id.
std::unique_ptr<JsepIceCandidate> candidate(
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
sdp_handler_->AddLocalIceCandidate(candidate.get());
OnIceCandidate(std::move(candidate));
}
}
void PeerConnection::OnTransportControllerCandidateError(
const cricket::IceCandidateErrorEvent& event) {
OnIceCandidateError(event.address, event.port, event.url, event.error_code,
event.error_text);
}
void PeerConnection::OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
// Sanity check.
for (const cricket::Candidate& candidate : candidates) {
if (candidate.transport_name().empty()) {
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
"empty content name in candidate "
<< candidate.ToString();
return;
}
}
sdp_handler_->RemoveLocalIceCandidates(candidates);
OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnTransportControllerCandidateChanged(
const cricket::CandidatePairChangeEvent& event) {
OnSelectedCandidatePairChanged(event);
}
void PeerConnection::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
}
// Returns the media index for a local ice candidate given the content name.
bool PeerConnection::GetLocalCandidateMediaIndex(
const std::string& content_name,
int* sdp_mline_index) {
if (!local_description() || !sdp_mline_index) {
return false;
}
bool content_found = false;
const ContentInfos& contents = local_description()->description()->contents();
for (size_t index = 0; index < contents.size(); ++index) {
if (contents[index].name == content_name) {
*sdp_mline_index = static_cast<int>(index);
content_found = true;
break;
}
}
return content_found;
}
Call::Stats PeerConnection::GetCallStats() {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<Call::Stats>(
RTC_FROM_HERE, [this] { return GetCallStats(); });
}
RTC_DCHECK_RUN_ON(worker_thread());
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
if (call_) {
return call_->GetStats();
} else {
return Call::Stats();
}
}
bool PeerConnection::SetupDataChannelTransport_n(const std::string& mid) {
DataChannelTransportInterface* transport =
transport_controller_->GetDataChannelTransport(mid);
if (!transport) {
RTC_LOG(LS_ERROR)
<< "Data channel transport is not available for data channels, mid="
<< mid;
return false;
}
RTC_LOG(LS_INFO) << "Setting up data channel transport for mid=" << mid;
data_channel_controller_.set_data_channel_transport(transport);
data_channel_controller_.SetupDataChannelTransport_n();
sctp_mid_n_ = mid;
cricket::DtlsTransportInternal* dtls_transport =
transport_controller_->GetDtlsTransport(mid);
if (dtls_transport) {
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(),
[this, name = dtls_transport->transport_name()] {
RTC_DCHECK_RUN_ON(signaling_thread());
sctp_transport_name_s_ = std::move(name);
}));
}
// Note: setting the data sink and checking initial state must be done last,
// after setting up the data channel. Setting the data sink may trigger
// callbacks to PeerConnection which require the transport to be completely
// set up (eg. OnReadyToSend()).
transport->SetDataSink(&data_channel_controller_);
return true;
}
void PeerConnection::TeardownDataChannelTransport_n() {
if (sctp_mid_n_) {
// `sctp_mid_` may still be active through an SCTP transport. If not, unset
// it.
RTC_LOG(LS_INFO) << "Tearing down data channel transport for mid="
<< *sctp_mid_n_;
sctp_mid_n_.reset();
}
data_channel_controller_.TeardownDataChannelTransport_n();
}
// Returns false if bundle is enabled and rtcp_mux is disabled.
bool PeerConnection::ValidateBundleSettings(
const SessionDescription* desc,
const std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) {
if (bundle_groups_by_mid.empty())
return true;
const cricket::ContentInfos& contents = desc->contents();
for (cricket::ContentInfos::const_iterator citer = contents.begin();
citer != contents.end(); ++citer) {
const cricket::ContentInfo* content = (&*citer);
RTC_DCHECK(content != NULL);
auto it = bundle_groups_by_mid.find(content->name);
if (it != bundle_groups_by_mid.end() && !content->rejected &&
content->type == MediaProtocolType::kRtp) {
if (!HasRtcpMuxEnabled(content))
return false;
}
}
// RTCP-MUX is enabled in all the contents.
return true;
}
void PeerConnection::ReportSdpFormatReceived(
const SessionDescriptionInterface& remote_description) {
int num_audio_mlines = 0;
int num_video_mlines = 0;
int num_audio_tracks = 0;
int num_video_tracks = 0;
for (const ContentInfo& content :
remote_description.description()->contents()) {
cricket::MediaType media_type = content.media_description()->type();
int num_tracks = std::max(
1, static_cast<int>(content.media_description()->streams().size()));
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
num_audio_mlines += 1;
num_audio_tracks += num_tracks;
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
num_video_mlines += 1;
num_video_tracks += num_tracks;
}
}
SdpFormatReceived format = kSdpFormatReceivedNoTracks;
if (num_audio_mlines > 1 || num_video_mlines > 1) {
format = kSdpFormatReceivedComplexUnifiedPlan;
} else if (num_audio_tracks > 1 || num_video_tracks > 1) {
format = kSdpFormatReceivedComplexPlanB;
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
format = kSdpFormatReceivedSimple;
}
switch (remote_description.GetType()) {
case SdpType::kOffer:
// Historically only offers were counted.
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived",
format, kSdpFormatReceivedMax);
break;
case SdpType::kAnswer:
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceivedAnswer",
format, kSdpFormatReceivedMax);
break;
default:
RTC_LOG(LS_ERROR) << "Can not report SdpFormatReceived for "
<< SdpTypeToString(remote_description.GetType());
break;
}
}
void PeerConnection::ReportSdpBundleUsage(
const SessionDescriptionInterface& remote_description) {
RTC_DCHECK_RUN_ON(signaling_thread());
bool using_bundle =
remote_description.description()->HasGroup(cricket::GROUP_TYPE_BUNDLE);
int num_audio_mlines = 0;
int num_video_mlines = 0;
int num_data_mlines = 0;
for (const ContentInfo& content :
remote_description.description()->contents()) {
cricket::MediaType media_type = content.media_description()->type();
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
num_audio_mlines += 1;
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
num_video_mlines += 1;
} else if (media_type == cricket::MEDIA_TYPE_DATA) {
num_data_mlines += 1;
}
}
bool simple = num_audio_mlines <= 1 && num_video_mlines <= 1;
BundleUsage usage = kBundleUsageMax;
if (num_audio_mlines == 0 && num_video_mlines == 0) {
if (num_data_mlines > 0) {
usage = using_bundle ? kBundleUsageBundleDatachannelOnly
: kBundleUsageNoBundleDatachannelOnly;
} else {
usage = kBundleUsageEmpty;
}
} else if (configuration_.sdp_semantics == SdpSemantics::kPlanB_DEPRECATED) {
// In plan-b, simple/complex usage will not show up in the number of
// m-lines or BUNDLE.
usage = using_bundle ? kBundleUsageBundlePlanB : kBundleUsageNoBundlePlanB;
} else {
if (simple) {
usage =
using_bundle ? kBundleUsageBundleSimple : kBundleUsageNoBundleSimple;
} else {
usage = using_bundle ? kBundleUsageBundleComplex
: kBundleUsageNoBundleComplex;
}
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.BundleUsage", usage,
kBundleUsageMax);
}
void PeerConnection::ReportIceCandidateCollected(
const cricket::Candidate& candidate) {
NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
if (candidate.address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
}
if (candidate.address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::MDNS_CANDIDATE_COLLECTED);
}
if (candidate.address().family() == AF_INET6) {
NoteUsageEvent(UsageEvent::IPV6_CANDIDATE_COLLECTED);
}
}
void PeerConnection::NoteUsageEvent(UsageEvent event) {
RTC_DCHECK_RUN_ON(signaling_thread());
usage_pattern_.NoteUsageEvent(event);
}
// Asynchronously adds remote candidates on the network thread.
void PeerConnection::AddRemoteCandidate(const std::string& mid,
const cricket::Candidate& candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
network_thread()->PostTask(SafeTask(
network_thread_safety_, [this, mid = mid, candidate = candidate] {
RTC_DCHECK_RUN_ON(network_thread());
std::vector<cricket::Candidate> candidates = {candidate};
RTCError error =
transport_controller_->AddRemoteCandidates(mid, candidates);
if (error.ok()) {
signaling_thread()->PostTask(SafeTask(
signaling_thread_safety_.flag(),
[this, candidate = std::move(candidate)] {
ReportRemoteIceCandidateAdded(candidate);
// Candidates successfully submitted for checking.
if (ice_connection_state() ==
PeerConnectionInterface::kIceConnectionNew ||
ice_connection_state() ==
PeerConnectionInterface::kIceConnectionDisconnected) {
// If state is New, then the session has just gotten its first
// remote ICE candidates, so go to Checking. If state is
// Disconnected, the session is re-using old candidates or
// receiving additional ones, so go to Checking. If state is
// Connected, stay Connected.
// TODO(bemasc): If state is Connected, and the new candidates
// are for a newly added transport, then the state actually
// _should_ move to checking. Add a way to distinguish that
// case.
SetIceConnectionState(
PeerConnectionInterface::kIceConnectionChecking);
}
// TODO(bemasc): If state is Completed, go back to Connected.
}));
} else {
RTC_LOG(LS_WARNING) << error.message();
}
}));
}
void PeerConnection::ReportUsagePattern() const {
usage_pattern_.ReportUsagePattern(observer_);
}
void PeerConnection::ReportRemoteIceCandidateAdded(
const cricket::Candidate& candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
if (candidate.address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED);
}
if (candidate.address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED);
}
if (candidate.address().family() == AF_INET6) {
NoteUsageEvent(UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED);
}
}
bool PeerConnection::SrtpRequired() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return (dtls_enabled_ ||
sdp_handler_->webrtc_session_desc_factory()->SdesPolicy() ==
cricket::SEC_REQUIRED);
}
void PeerConnection::OnTransportControllerGatheringState(
cricket::IceGatheringState state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (state == cricket::kIceGatheringGathering) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
} else if (state == cricket::kIceGatheringComplete) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
} else if (state == cricket::kIceGatheringNew) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringNew);
} else {
RTC_LOG(LS_ERROR) << "Unknown state received: " << state;
RTC_DCHECK_NOTREACHED();
}
}
// Runs on network_thread().
void PeerConnection::ReportTransportStats() {
TRACE_EVENT0("webrtc", "PeerConnection::ReportTransportStats");
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, std::set<cricket::MediaType>>
media_types_by_transport_name;
if (ConfiguredForMedia()) {
for (const auto& transceiver :
rtp_manager()->transceivers()->UnsafeList()) {
if (transceiver->internal()->channel()) {
std::string transport_name(
transceiver->internal()->channel()->transport_name());
media_types_by_transport_name[transport_name].insert(
transceiver->media_type());
}
}
}
if (sctp_mid_n_) {
cricket::DtlsTransportInternal* dtls_transport =
transport_controller_->GetDtlsTransport(*sctp_mid_n_);
if (dtls_transport) {
media_types_by_transport_name[dtls_transport->transport_name()].insert(
cricket::MEDIA_TYPE_DATA);
}
}
for (const auto& entry : media_types_by_transport_name) {
const std::string& transport_name = entry.first;
const std::set<cricket::MediaType> media_types = entry.second;
cricket::TransportStats stats;
if (transport_controller_->GetStats(transport_name, &stats)) {
ReportBestConnectionState(stats);
ReportNegotiatedCiphers(dtls_enabled_, stats, media_types);
}
}
}
// Walk through the ConnectionInfos to gather best connection usage
// for IPv4 and IPv6.
// static (no member state required)
void PeerConnection::ReportBestConnectionState(
const cricket::TransportStats& stats) {
for (const cricket::TransportChannelStats& channel_stats :
stats.channel_stats) {
for (const cricket::ConnectionInfo& connection_info :
channel_stats.ice_transport_stats.connection_infos) {
if (!connection_info.best_connection) {
continue;
}
const cricket::Candidate& local = connection_info.local_candidate;
const cricket::Candidate& remote = connection_info.remote_candidate;
// Increment the counter for IceCandidatePairType.
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
(local.type() == RELAY_PORT_TYPE &&
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else {
RTC_CHECK_NOTREACHED();
}
// Increment the counter for IP type.
if (local.address().family() == AF_INET) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
kBestConnections_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
} else if (local.address().family() == AF_INET6) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
kBestConnections_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
RTC_CHECK(!local.address().hostname().empty() &&
local.address().IsUnresolvedIP());
}
return;
}
}
}
// static
void PeerConnection::ReportNegotiatedCiphers(
bool dtls_enabled,
const cricket::TransportStats& stats,
const std::set<cricket::MediaType>& media_types) {
if (!dtls_enabled || stats.channel_stats.empty()) {
return;
}
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
if (srtp_crypto_suite == rtc::kSrtpInvalidCryptoSuite &&
ssl_cipher_suite == rtc::kTlsNullWithNullNull) {
return;
}
if (srtp_crypto_suite != rtc::kSrtpInvalidCryptoSuite) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
rtc::kSrtpCryptoSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
rtc::kSrtpCryptoSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
rtc::kSrtpCryptoSuiteMaxValue);
break;
default:
RTC_DCHECK_NOTREACHED();
continue;
}
}
}
if (ssl_cipher_suite != rtc::kTlsNullWithNullNull) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
rtc::kSslCipherSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
rtc::kSslCipherSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
rtc::kSslCipherSuiteMaxValue);
break;
default:
RTC_DCHECK_NOTREACHED();
continue;
}
}
}
}
bool PeerConnection::OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
DataChannelTransportInterface* data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread());
bool ret = true;
if (ConfiguredForMedia()) {
for (const auto& transceiver :
rtp_manager()->transceivers()->UnsafeList()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && channel->mid() == mid) {
ret = channel->SetRtpTransport(rtp_transport);
}
}
}
if (mid == sctp_mid_n_) {
data_channel_controller_.OnTransportChanged(data_channel_transport);
if (dtls_transport) {
signaling_thread()->PostTask(SafeTask(
signaling_thread_safety_.flag(),
[this,
name = std::string(dtls_transport->internal()->transport_name())] {
RTC_DCHECK_RUN_ON(signaling_thread());
sctp_transport_name_s_ = std::move(name);
}));
}
}
return ret;
}
PeerConnectionObserver* PeerConnection::Observer() const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(observer_);
return observer_;
}
void PeerConnection::StartSctpTransport(int local_port,
int remote_port,
int max_message_size) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!sctp_mid_s_)
return;
network_thread()->PostTask(SafeTask(
network_thread_safety_,
[this, mid = *sctp_mid_s_, local_port, remote_port, max_message_size] {
rtc::scoped_refptr<SctpTransport> sctp_transport =
transport_controller_n()->GetSctpTransport(mid);
if (sctp_transport)
sctp_transport->Start(local_port, remote_port, max_message_size);
}));
}
CryptoOptions PeerConnection::GetCryptoOptions() {
RTC_DCHECK_RUN_ON(signaling_thread());
// TODO(bugs.webrtc.org/9891) - Remove PeerConnectionFactory::CryptoOptions
// after it has been removed.
return configuration_.crypto_options.has_value()
? *configuration_.crypto_options
: options_.crypto_options;
}
void PeerConnection::ClearStatsCache() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (stats_collector_) {
stats_collector_->ClearCachedStatsReport();
}
}
bool PeerConnection::ShouldFireNegotiationNeededEvent(uint32_t event_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->ShouldFireNegotiationNeededEvent(event_id);
}
void PeerConnection::RequestUsagePatternReportForTesting() {
RTC_DCHECK_RUN_ON(signaling_thread());
message_handler_.RequestUsagePatternReport(
[this]() {
RTC_DCHECK_RUN_ON(signaling_thread());
ReportUsagePattern();
},
/* delay_ms= */ 0);
}
std::function<void(const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us)>
PeerConnection::InitializeRtcpCallback() {
RTC_DCHECK_RUN_ON(network_thread());
return [this](const rtc::CopyOnWriteBuffer& packet, int64_t packet_time_us) {
RTC_DCHECK_RUN_ON(network_thread());
call_ptr_->Receiver()->DeliverPacket(MediaType::ANY, packet,
packet_time_us);
};
}
} // namespace webrtc