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webrtc/src/2bc24d44be71c186ee0756725abaed7015fbc8bc/./pc/peer_connection.cc
blob: 8c9fdf6b1531cbcffcf4afcbe686447a2e979942 [file]
/*
* 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 <climits>
#include <cstddef>
#include <cstdint>
#include <functional>
#include <map>
#include <memory>
#include <optional>
#include <set>
#include <string>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/functional/any_invocable.h"
#include "absl/strings/match.h"
#include "absl/strings/str_cat.h"
#include "absl/strings/string_view.h"
#include "api/adaptation/resource.h"
#include "api/audio/audio_device.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/data_channel_event_observer_interface.h"
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/environment/environment.h"
#include "api/jsep.h"
#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sctp_transport_interface.h"
#include "api/sequence_checker.h"
#include "api/set_local_description_observer_interface.h"
#include "api/set_remote_description_observer_interface.h"
#include "api/stats/rtc_stats_collector_callback.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/bandwidth_estimation_settings.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/data_channel_transport_interface.h"
#include "api/transport/enums.h"
#include "api/turn_customizer.h"
#include "api/uma_metrics.h"
#include "api/units/time_delta.h"
#include "api/video/video_codec_constants.h"
#include "call/audio_state.h"
#include "call/packet_receiver.h"
#include "call/payload_type.h"
#include "media/base/codec.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/connection_info.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "p2p/dtls/dtls_transport_internal.h"
#include "pc/channel_interface.h"
#include "pc/codec_vendor.h"
#include "pc/connection_context.h"
#include "pc/data_channel_utils.h"
#include "pc/dtls_transport.h"
#include "pc/ice_server_parsing.h"
#include "pc/jsep_transport_controller.h"
#include "pc/legacy_stats_collector.h"
#include "pc/media_session.h"
#include "pc/rtc_stats_collector.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/rtp_transceiver.h"
#include "pc/rtp_transmission_manager.h"
#include "pc/rtp_transport_internal.h"
#include "pc/sctp_data_channel.h"
#include "pc/sctp_transport.h"
#include "pc/sdp_offer_answer.h"
#include "pc/sdp_state_provider.h"
#include "pc/session_description.h"
#include "pc/transceiver_list.h"
#include "pc/transport_stats.h"
#include "pc/usage_pattern.h"
#include "rtc_base/checks.h"
#include "rtc_base/containers/flat_map.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/crypto_random.h"
#include "rtc_base/ip_address.h"
#include "rtc_base/logging.h"
#include "rtc_base/net_helper.h"
#include "rtc_base/net_helpers.h"
#include "rtc_base/network.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/unique_id_generator.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace {
const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
class CodecLookupHelperForPeerConnection : public CodecLookupHelper {
public:
explicit CodecLookupHelperForPeerConnection(PeerConnection* self)
: self_(self),
codec_vendor_(self_->context()->media_engine(),
self_->context()->use_rtx(),
self_->trials()) {}
webrtc::PayloadTypeSuggester* PayloadTypeSuggester() override {
return self_->pt_suggester();
}
CodecVendor* GetCodecVendor() override { return &codec_vendor_; }
private:
PeerConnection* self_;
CodecVendor codec_vendor_;
};
uint32_t ConvertIceTransportTypeToCandidateFilter(
PeerConnectionInterface::IceTransportsType type) {
switch (type) {
case PeerConnectionInterface::kNone:
return CF_NONE;
case PeerConnectionInterface::kRelay:
return CF_RELAY;
case PeerConnectionInterface::kNoHost:
return (CF_ALL & ~CF_HOST);
case PeerConnectionInterface::kAll:
return CF_ALL;
default:
RTC_DCHECK_NOTREACHED();
}
return CF_NONE;
}
IceCandidatePairType GetIceCandidatePairType(const Candidate& local,
const Candidate& remote) {
if (local.is_local() && remote.is_local()) {
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 (local.is_local()) {
if (remote.is_stun())
return kIceCandidatePairHostSrflx;
if (remote.is_relay())
return kIceCandidatePairHostRelay;
if (remote.is_prflx())
return kIceCandidatePairHostPrflx;
} else if (local.is_stun()) {
if (remote.is_local())
return kIceCandidatePairSrflxHost;
if (remote.is_stun())
return kIceCandidatePairSrflxSrflx;
if (remote.is_relay())
return kIceCandidatePairSrflxRelay;
if (remote.is_prflx())
return kIceCandidatePairSrflxPrflx;
} else if (local.is_relay()) {
if (remote.is_local())
return kIceCandidatePairRelayHost;
if (remote.is_stun())
return kIceCandidatePairRelaySrflx;
if (remote.is_relay())
return kIceCandidatePairRelayRelay;
if (remote.is_prflx())
return kIceCandidatePairRelayPrflx;
} else if (local.is_prflx()) {
if (remote.is_local())
return kIceCandidatePairPrflxHost;
if (remote.is_stun())
return kIceCandidatePairPrflxSrflx;
if (remote.is_relay())
return kIceCandidatePairPrflxRelay;
}
return kIceCandidatePairMax;
}
// 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;
}
// Checks for valid pool size range and if a previous value has already been
// set, which is done via SetLocalDescription.
RTCError ValidateIceCandidatePoolSize(
int ice_candidate_pool_size,
std::optional<int> previous_ice_candidate_pool_size) {
// Note that this isn't possible through chromium, since it's an unsigned
// short in WebIDL.
if (ice_candidate_pool_size < 0 ||
ice_candidate_pool_size > static_cast<int>(UINT16_MAX)) {
return RTCError(RTCErrorType::INVALID_RANGE);
}
// 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 (previous_ice_candidate_pool_size.has_value() &&
ice_candidate_pool_size != previous_ice_candidate_pool_size.value()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Can't change candidate pool size after calling "
"SetLocalDescription.");
}
return RTCError::OK();
}
// The simplest (and most future-compatible) way to tell if a 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.
// This helper function accepts a proposed new `configuration` object, an
// existing configuration and returns a valid, modified, configuration that's
// based on the existing configuration, with modified properties copied from
// `configuration`.
// If the result of creating a modified configuration doesn't pass the above
// `operator==` test or a call to `ValidateConfiguration()`, then the function
// will return an error. Otherwise, the return value will be the new config.
RTCErrorOr<PeerConnectionInterface::RTCConfiguration> ApplyConfiguration(
const PeerConnectionInterface::RTCConfiguration& configuration,
const PeerConnectionInterface::RTCConfiguration& existing_configuration) {
PeerConnectionInterface::RTCConfiguration modified_config =
existing_configuration;
modified_config.type = configuration.type;
modified_config.crypto_options = configuration.crypto_options;
modified_config.always_negotiate_data_channels =
configuration.always_negotiate_data_channels;
// ICE configuration.
modified_config.servers = configuration.servers;
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.turn_logging_id = configuration.turn_logging_id;
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.");
}
RTCError err = IceConfig(modified_config).IsValid();
if (!err.ok()) {
return err;
}
return modified_config;
}
bool HasRtcpMuxEnabled(const 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.
return (dependencies.cert_generator || !configuration.certificates.empty());
}
void NoteServerUsage(UsagePattern& usage_pattern,
const ServerAddresses& stun_servers,
const std::vector<RelayServerConfig>& turn_servers) {
if (!stun_servers.empty()) {
usage_pattern.NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
usage_pattern.NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
}
template <typename Observer,
typename std::enable_if_t<
std::is_same_v<Observer, SetSessionDescriptionObserver*>,
bool> = true>
absl::AnyInvocable<void() &&> ReportFailure(Observer& o, RTCError error) {
return [o = scoped_refptr<SetSessionDescriptionObserver>(o),
error = std::move(error)]() { o->OnFailure(error); };
}
template <
typename Observer,
typename std::enable_if_t<
std::is_same_v<Observer,
scoped_refptr<SetLocalDescriptionObserverInterface>>,
bool> = true>
absl::AnyInvocable<void() &&> ReportFailure(Observer& o, RTCError error) {
return [o = std::move(o), error = std::move(error)]() {
o->OnSetLocalDescriptionComplete(error);
};
}
template <
typename Observer,
typename std::enable_if_t<
std::is_same_v<Observer,
scoped_refptr<SetRemoteDescriptionObserverInterface>>,
bool> = true>
absl::AnyInvocable<void() &&> ReportFailure(Observer& o, RTCError error) {
return [o = std::move(o), error = std::move(error)]() {
o->OnSetRemoteDescriptionComplete(error);
};
}
// Checks if the observer and description pointers are valid.
// If there's an error, the function will return false and optionally
// notify the observer asynchronously of the error.
// If both are valid, the function will reset the thread ownership of
// the description object and return true.
template <typename Observer, typename Description>
bool CheckValidSetDescription(Observer& observer,
Description& desc,
Thread* signaling) {
if (!observer) {
RTC_LOG(LS_ERROR) << "Observer is NULL.";
return false;
} else if (!desc) {
signaling->PostTask(
ReportFailure(observer, RTCError(RTCErrorType::INVALID_PARAMETER,
"SessionDescription is NULL.")));
return false;
}
// Make sure the description object now considers the current thread its home
// by detaching from any potential previous thread.
desc->RelinquishThreadOwnership();
return true;
}
} // 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<scoped_refptr<RTCCertificate>> certificates;
int ice_candidate_pool_size;
bool disable_ipv6_on_wifi;
int max_ipv6_networks;
bool disable_link_local_networks;
std::optional<int> screencast_min_bitrate;
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 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;
std::optional<int> ice_check_interval_strong_connectivity;
std::optional<int> ice_check_interval_weak_connectivity;
std::optional<int> ice_check_min_interval;
std::optional<int> ice_unwritable_timeout;
std::optional<int> ice_unwritable_min_checks;
std::optional<int> ice_inactive_timeout;
std::optional<int> stun_candidate_keepalive_interval;
TurnCustomizer* turn_customizer;
SdpSemantics sdp_semantics;
std::optional<AdapterType> network_preference;
CryptoOptions crypto_options;
bool offer_extmap_allow_mixed;
std::string turn_logging_id;
bool enable_implicit_rollback;
std::optional<int> report_usage_pattern_delay_ms;
std::optional<int> stable_writable_connection_ping_interval_ms;
VpnPreference vpn_preference;
std::vector<NetworkMask> vpn_list;
PortAllocatorConfig port_allocator_config;
std::optional<TimeDelta> pacer_burst_interval;
bool always_negotiate_data_channels;
int max_sctp_streams;
};
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_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 &&
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 &&
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 &&
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 &&
pacer_burst_interval == o.pacer_burst_interval &&
always_negotiate_data_channels == o.always_negotiate_data_channels &&
max_sctp_streams == o.max_sctp_streams;
}
bool PeerConnectionInterface::RTCConfiguration::operator!=(
const PeerConnectionInterface::RTCConfiguration& o) const {
return !(*this == o);
}
scoped_refptr<PeerConnection> PeerConnection::Create(
const Environment& env,
scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
std::unique_ptr<Call> call,
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies& dependencies,
const ServerAddresses& stun_servers,
const std::vector<RelayServerConfig>& turn_servers) {
RTC_DCHECK(IceConfig(configuration).IsValid().ok());
RTC_DCHECK(dependencies.observer);
RTC_DCHECK(dependencies.async_dns_resolver_factory);
RTC_DCHECK(dependencies.allocator);
bool is_unified_plan =
configuration.sdp_semantics == SdpSemantics::kUnifiedPlan;
bool dtls_enabled = DtlsEnabled(configuration, options, dependencies);
TRACE_EVENT0("webrtc", "PeerConnection::Create");
return make_ref_counted<PeerConnection>(
configuration, env, context, options, is_unified_plan, std::move(call),
dependencies, stun_servers, turn_servers, dtls_enabled);
}
PeerConnection::PeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const Environment& env,
scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
bool is_unified_plan,
std::unique_ptr<Call> call,
PeerConnectionDependencies& dependencies,
const ServerAddresses& stun_servers,
const std::vector<RelayServerConfig>& turn_servers,
bool dtls_enabled)
: env_(env),
context_(context),
options_(options),
observer_(dependencies.observer),
is_unified_plan_(is_unified_plan),
dtls_enabled_(dtls_enabled),
configuration_(configuration),
async_dns_resolver_factory_(
std::move(dependencies.async_dns_resolver_factory)),
port_allocator_(std::move(dependencies.allocator)),
lna_permission_factory_(std::move(dependencies.lna_permission_factory)),
ice_transport_factory_(std::move(dependencies.ice_transport_factory)),
dtls_transport_factory_(std::move(dependencies.dtls_transport_factory)),
rtp_transport_factory_(std::move(dependencies.rtp_transport_factory)),
tls_cert_verifier_(std::move(dependencies.tls_cert_verifier)),
call_(std::move(call)),
network_thread_safety_(
PendingTaskSafetyFlag::CreateAttachedToTaskQueue(true,
network_thread())),
worker_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
/*alive=*/call_ != nullptr,
worker_thread())),
call_ptr_(call_.get()),
legacy_stats_(std::make_unique<LegacyStatsCollector>(this, env_.clock())),
stats_collector_(this, env_),
// 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_(absl::StrCat(CreateRandomId64() & LLONG_MAX)),
data_channel_controller_(this),
message_handler_(signaling_thread()),
codec_lookup_helper_(
std::make_unique<CodecLookupHelperForPeerConnection>(this)),
weak_factory_(this) {
// Field trials specific to the peerconnection should be owned by the `env`,
RTC_DCHECK(dependencies.trials == nullptr);
transport_controller_copy_ =
InitializeNetworkThread(stun_servers, turn_servers);
if (call_ptr_) {
worker_thread()->BlockingCall([this] {
RTC_DCHECK_RUN_ON(worker_thread());
if (context_->is_configured_for_media()) {
media_engine_ref_ =
std::make_unique<ConnectionContext::MediaEngineReference>(context_);
}
});
}
sdp_handler_ = SdpOfferAnswerHandler::Create(
env_, this, configuration_, std::move(dependencies.cert_generator),
std::move(dependencies.video_bitrate_allocator_factory), context_.get(),
codec_lookup_helper_.get());
rtp_manager_ = std::make_unique<RtpTransmissionManager>(
env_, call_ptr_, IsUnifiedPlan(), context_.get(),
codec_lookup_helper_.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() && ConfiguredForMedia()) {
rtp_manager_->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(),
make_ref_counted<RtpTransceiver>(
env_, MediaType::AUDIO, context_.get(),
codec_lookup_helper_.get(), legacy_stats_.get())));
rtp_manager_->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(),
make_ref_counted<RtpTransceiver>(
env_, MediaType::VIDEO, context_.get(),
codec_lookup_helper_.get(), legacy_stats_.get())));
}
const 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);
}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_LOG_THREAD_BLOCK_COUNT();
sdp_handler_->PrepareForShutdown();
// In case `Close()` wasn't called, always make sure the controller cancels
// potentially pending operations.
data_channel_controller_.PrepareForShutdown();
std::vector<absl::AnyInvocable<void() &&>> network_tasks;
std::vector<absl::AnyInvocable<void() &&>> worker_tasks;
// Stop transceivers before destroying the stats collector because
// AudioRtpSender has a reference to the LegacyStatsCollector that it will
// update when stopping. The BaseChannels will eventually be deleted below
// when all the network and worker tasks are executed.
sdp_handler_->GetMediaChannelTeardownTasks(network_tasks, worker_tasks);
legacy_stats_.reset(nullptr);
network_tasks.push_back(
stats_collector_.CancelPendingRequestAndGetShutdownTask());
CloseOnNetworkThread(network_tasks);
// call_ must be destroyed on the worker thread.
worker_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(worker_thread());
for (auto& task : worker_tasks) {
std::move(task)();
task = nullptr;
}
worker_thread_safety_->SetNotAlive();
call_.reset();
media_engine_ref_.reset();
});
if (sdp_handler_) {
sdp_handler_->ResetSessionDescFactory();
}
data_channel_controller_.PrepareForShutdown();
// The expectation is that there will have been 1 blocking call for the worker
// thread and optionally 1 task for the network thread.
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
JsepTransportController* PeerConnection::InitializeNetworkThread(
const ServerAddresses& stun_servers,
const std::vector<RelayServerConfig>& turn_servers) {
RTC_DCHECK_RUN_ON(signaling_thread());
NoteServerUsage(usage_pattern_, stun_servers, turn_servers);
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;
config.crypto_options = configuration_.crypto_options;
// Maybe enable or disable PQC from FieldTrials
config.crypto_options.ephemeral_key_exchange_cipher_groups.Update(&trials());
config.transport_observer = this;
config.rtcp_handler = InitializeRtcpCallback();
config.un_demuxable_packet_handler = InitializeUnDemuxablePacketHandler();
#if defined(ENABLE_EXTERNAL_AUTH)
config.enable_external_auth = true;
#endif
// 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.dtls_transport_factory = dtls_transport_factory_.get();
config.rtp_transport_factory = rtp_transport_factory_.get();
config.on_dtls_handshake_error =
[weak_ptr = weak_factory_.GetWeakPtr()](SSLHandshakeError s) {
if (weak_ptr) {
weak_ptr->OnTransportControllerDtlsHandshakeError(s);
}
};
config.signal_ice_candidates_gathered =
[this](absl::string_view transport,
const std::vector<Candidate>& candidates) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(),
[this, t = std::string(transport), c = candidates]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidatesGathered(t, c);
}));
};
config.signal_ice_connection_state = [this](webrtc::IceConnectionState s) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerConnectionState(s);
}));
};
config.signal_connection_state =
[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);
}));
};
config.signal_standardized_ice_connection_state =
[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);
}));
};
config.signal_ice_gathering_state = [this](::webrtc::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);
}));
};
config.signal_ice_candidate_error =
[this](const 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);
}));
};
config.signal_ice_candidates_removed =
[this](IceTransportInternal* transport, const std::vector<Candidate>& c) {
RTC_DCHECK_RUN_ON(network_thread());
std::string mid = transport->transport_name();
signaling_thread()->PostTask(SafeTask(
signaling_thread_safety_.flag(), [this, mid = mid, c = c]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidatesRemoved(mid, c);
}));
};
config.signal_ice_candidate_pair_changed =
[this](const 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);
}));
};
auto transport_controller = std::make_unique<JsepTransportController>(
env_, signaling_thread(), network_thread(), port_allocator_.get(),
async_dns_resolver_factory_.get(), lna_permission_factory_.get(),
std::move(config));
return network_thread()->BlockingCall([&, config = &configuration_,
controller = std::move(
transport_controller)]() mutable {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(network_thread_safety_->alive());
InitializePortAllocatorResult pa_result =
InitializePortAllocator_n(stun_servers, turn_servers, *config);
// 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(std::move(controller), *config);
});
}
void PeerConnection::CloseOnNetworkThread(
std::vector<absl::AnyInvocable<void() &&>>& network_tasks) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (transport_controller_copy_ || !network_tasks.empty()) {
network_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(network_thread());
for (auto& task : network_tasks) {
std::move(task)();
task = nullptr;
}
if (network_thread_safety_->alive()) {
// port_allocator_ and transport_controller_ live on the network thread
// and must be destroyed there.
TeardownDataChannelTransport_n(RTCError::OK());
port_allocator_->DiscardCandidatePool();
transport_controller_.reset();
port_allocator_.reset();
network_thread_safety_->SetNotAlive();
}
});
}
if (transport_controller_copy_) {
transport_controller_copy_ = nullptr;
sctp_mid_s_.reset();
SetSctpTransportName("");
} else {
RTC_DCHECK(!sctp_mid_s_);
RTC_DCHECK(sctp_transport_name_s_.empty());
}
}
JsepTransportController* PeerConnection::InitializeTransportController_n(
std::unique_ptr<JsepTransportController> controller,
const RTCConfiguration& configuration) {
transport_controller_ = std::move(controller);
IceConfig ice_config(configuration);
ice_config.dtls_handshake_in_stun = CanAttemptDtlsStunPiggybacking();
transport_controller_->SetIceConfig(ice_config);
return transport_controller_.get();
}
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();
}
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<scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
return AddTrack(std::move(track), stream_ids, nullptr);
}
RTCErrorOr<scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
const std::vector<RtpEncodingParameters>& init_send_encodings) {
return AddTrack(std::move(track), stream_ids, &init_send_encodings);
}
RTCErrorOr<scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
const std::vector<RtpEncodingParameters>* init_send_encodings) {
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() + ".");
}
RTCErrorOr<scoped_refptr<RtpSenderInterface>> sender_or_error;
if (IsUnifiedPlan()) {
sender_or_error = rtp_manager()->AddTrackUnifiedPlan(
configuration_.media_config, sdp_handler_->audio_options(),
sdp_handler_->video_options(), GetCryptoOptions(),
sdp_handler_->video_bitrate_allocator_factory(), track, stream_ids,
init_send_encodings);
} else {
sender_or_error =
rtp_manager()->AddTrackPlanB(track, stream_ids, init_send_encodings);
}
if (sender_or_error.ok()) {
sdp_handler_->UpdateNegotiationNeeded();
legacy_stats_->AddTrack(track.get());
}
return sender_or_error;
}
RTCError PeerConnection::RemoveTrackOrError(
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() == webrtc::MediaType::AUDIO) {
removed =
rtp_manager()->GetAudioTransceiver()->internal()->RemoveSenderPlanB(
sender.get());
} else {
RTC_DCHECK_EQ(webrtc::MediaType::VIDEO, sender->media_type());
removed =
rtp_manager()->GetVideoTransceiver()->internal()->RemoveSenderPlanB(
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();
}
scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindTransceiverBySender(
scoped_refptr<RtpSenderInterface> sender) {
return rtp_manager()->transceivers()->FindBySender(sender);
}
RTCErrorOr<scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(scoped_refptr<MediaStreamTrackInterface> track) {
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
return AddTransceiver(track, RtpTransceiverInit());
}
RTCErrorOr<scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(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");
}
MediaType media_type;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
media_type = MediaType::AUDIO;
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
media_type = MediaType::VIDEO;
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track kind is not audio or video");
}
return AddTransceiver(media_type, track, init);
}
RTCErrorOr<scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(MediaType media_type) {
return AddTransceiver(media_type, RtpTransceiverInit());
}
RTCErrorOr<scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(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 == MediaType::AUDIO || media_type == MediaType::VIDEO)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"media type is not audio or video");
}
return AddTransceiver(media_type, nullptr, init);
}
RTCErrorOr<scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(MediaType media_type,
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 == MediaType::AUDIO || media_type == MediaType::VIDEO));
if (track) {
RTC_DCHECK_EQ(media_type,
(track->kind() == MediaStreamTrackInterface::kAudioKind
? MediaType::AUDIO
: MediaType::VIDEO));
}
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 == MediaType::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) {
UniqueStringGenerator rid_generator;
for (RtpEncodingParameters& encoding : parameters.encodings) {
encoding.rid = rid_generator.GenerateString();
}
}
// If no encoding parameters were provided, a default entry is created.
if (parameters.encodings.empty()) {
parameters.encodings.push_back({});
}
if (UnimplementedRtpParameterHasValue(parameters)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
std::vector<Codec> codecs;
// Gather the current codec capabilities to allow checking scalabilityMode and
// codec selection against supported values.
CodecVendor codec_vendor(context_->media_engine(), false, trials());
if (media_type == MediaType::VIDEO) {
codecs = codec_vendor.video_send_codecs().codecs();
} else {
codecs = codec_vendor.audio_send_codecs().codecs();
}
auto result = CheckRtpParametersValues(parameters, codecs, std::nullopt,
env_.field_trials());
if (!result.ok()) {
if (result.type() == RTCErrorType::INVALID_MODIFICATION) {
result.set_type(RTCErrorType::UNSUPPORTED_OPERATION);
}
LOG_AND_RETURN_ERROR(result.type(), result.message());
}
RTC_LOG(LS_INFO) << "Adding " << 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()
: CreateRandomUuid());
auto transceiver = rtp_manager()->CreateAndAddTransceiver(
configuration_.media_config, sdp_handler_->audio_options(),
sdp_handler_->video_options(), configuration_.crypto_options,
sdp_handler_->video_bitrate_allocator_factory(), media_type, track,
init.stream_ids, parameters.encodings,
/*header_extensions_to_negotiate=*/{}, sender_id);
transceiver->internal()->set_direction(init.direction);
if (update_negotiation_needed) {
sdp_handler_->UpdateNegotiationNeeded();
}
return scoped_refptr<RtpTransceiverInterface>(transceiver);
}
void PeerConnection::OnNegotiationNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsClosed());
sdp_handler_->UpdateNegotiationNeeded();
}
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(CreateRandomUuid());
RTC_LOG(LS_INFO)
<< "No stream_id specified for sender. Generated stream ID: "
<< stream_ids[0];
} else {
stream_ids.push_back(stream_id);
}
scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
auto audio_sender = AudioRtpSender::Create(
env_, worker_thread(), CreateRandomUuid(), legacy_stats_.get(), nullptr,
rtp_manager()->voice_media_send_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), audio_sender);
rtp_manager()->GetAudioTransceiver()->internal()->AddSenderPlanB(
new_sender);
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
auto video_sender = VideoRtpSender::Create(
env_, worker_thread(), CreateRandomUuid(), nullptr,
rtp_manager()->video_media_send_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), video_sender);
rtp_manager()->GetVideoTransceiver()->internal()->AddSenderPlanB(
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<scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<scoped_refptr<RtpSenderInterface>> ret;
if (ConfiguredForMedia()) {
for (const auto& sender : rtp_manager()->GetSendersInternal()) {
ret.push_back(sender);
}
}
return ret;
}
std::vector<scoped_refptr<RtpReceiverInterface>> PeerConnection::GetReceivers()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<scoped_refptr<RtpReceiverInterface>> ret;
if (ConfiguredForMedia()) {
for (const auto& receiver : rtp_manager()->GetReceiversInternal()) {
ret.push_back(receiver);
}
}
return ret;
}
std::vector<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<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(callback);
RTC_LOG_THREAD_BLOCK_COUNT();
stats_collector_.GetStatsReport(
scoped_refptr<RTCStatsCollectorCallback>(callback));
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
void PeerConnection::GetStats(
scoped_refptr<RtpSenderInterface> selector,
scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_LOG_THREAD_BLOCK_COUNT();
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(
scoped_refptr<RtpReceiverInterface> selector,
scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_LOG_THREAD_BLOCK_COUNT();
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_;
}
std::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 std::nullopt;
}
// TODO(bugs.webrtc.org/7443): Change to retrieve from session-level option.
if (description->description()->transport_infos().empty()) {
return std::nullopt;
}
return description->description()->transport_infos()[0].description.HasOption(
"trickle");
}
RTCErrorOr<scoped_refptr<DataChannelInterface>>
PeerConnection::CreateDataChannelOrError(const std::string& label,
const DataChannelInit* config) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"CreateDataChannelOrError: PeerConnection is closed.");
}
bool first_datachannel = !data_channel_controller_.HasUsedDataChannels();
InternalDataChannelInit internal_config;
if (config) {
internal_config = InternalDataChannelInit(*config);
}
internal_config.fallback_ssl_role = sdp_handler_->GuessSslRole();
RTCErrorOr<scoped_refptr<DataChannelInterface>> ret =
data_channel_controller_.InternalCreateDataChannelWithProxy(
label, internal_config);
if (!ret.ok()) {
return ret.MoveError();
}
ClearStatsCache();
scoped_refptr<DataChannelInterface> channel = ret.MoveValue();
// Check the onRenegotiationNeeded event (with plan-b backward compat)
if (configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan ||
(configuration_.sdp_semantics == SdpSemantics::kPlanB_DEPRECATED &&
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) {
if (!CheckValidSetDescription(observer, desc, signaling_thread()))
return;
RunOnSignalingThread([this, desc, observer]() mutable {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer, desc);
});
}
// This method bypasses the proxy, so can be called from any thread.
void PeerConnection::SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
if (!CheckValidSetDescription(observer, desc, signaling_thread()))
return;
RunOnSignalingThread(
[this, desc = std::move(desc), observer = std::move(observer)]() mutable {
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(
scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer);
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
if (!CheckValidSetDescription(observer, desc, signaling_thread()))
return;
RunOnSignalingThread([this, desc, observer]() mutable {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetRemoteDescription(observer, desc);
});
}
void PeerConnection::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
if (!CheckValidSetDescription(observer, desc, signaling_thread()))
return;
RunOnSignalingThread(
[this, desc = std::move(desc), observer = std::move(observer)]() mutable {
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.");
}
const bool has_local_description = local_description() != nullptr;
RTCError validate_error = ValidateIceCandidatePoolSize(
configuration.ice_candidate_pool_size,
has_local_description
? std::optional<int>(configuration_.ice_candidate_pool_size)
: std::nullopt);
if (!validate_error.ok()) {
return validate_error;
}
if (has_local_description &&
configuration.crypto_options != configuration_.crypto_options) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
"Can't change crypto_options after calling "
"SetLocalDescription.");
}
// Create a new, configuration object whose peerconnection config
// will have been validated.
RTCErrorOr<RTCConfiguration> validated_config =
ApplyConfiguration(configuration, configuration_);
if (!validated_config.ok()) {
return validated_config.error();
}
// Parse ICE servers before hopping to network thread.
ServerAddresses stun_servers;
std::vector<RelayServerConfig> turn_servers;
validate_error = ParseAndValidateIceServersFromConfiguration(
configuration, stun_servers, turn_servers);
if (!validate_error.ok()) {
return validate_error;
}
NoteServerUsage(usage_pattern_, stun_servers, turn_servers);
const RTCConfiguration& modified_config = validated_config.value();
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();
IceConfig ice_config(modified_config);
ice_config.dtls_handshake_in_stun = CanAttemptDtlsStunPiggybacking();
// Apply part of the configuration on the network thread. In theory this
// shouldn't fail.
if (!network_thread()->BlockingCall(
[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.");
}
configuration_ = modified_config;
return RTCError::OK();
}
bool PeerConnection::AddIceCandidate(const IceCandidate* ice_candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
ClearStatsCache();
return sdp_handler_->AddIceCandidate(ice_candidate);
}
void PeerConnection::AddIceCandidate(std::unique_ptr<IceCandidate> candidate,
std::function<void(RTCError)> callback) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->AddIceCandidate(std::move(candidate),
[this, callback](RTCError result) {
ClearStatsCache();
callback(result);
});
}
bool PeerConnection::RemoveIceCandidate(const IceCandidate* candidate) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidate");
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->RemoveIceCandidate(candidate);
}
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->BlockingCall([&]() { return SetBitrate(bitrate); });
}
RTC_DCHECK_RUN_ON(worker_thread());
if (!call_) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
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");
}
}
call_->SetClientBitratePreferences(bitrate);
return RTCError::OK();
}
void PeerConnection::ReconfigureBandwidthEstimation(
const BandwidthEstimationSettings& settings) {
worker_thread()->PostTask(SafeTask(worker_thread_safety_, [this, settings]() {
RTC_DCHECK_RUN_ON(worker_thread());
call_->GetTransportControllerSend()->ReconfigureBandwidthEstimation(
settings);
}));
}
void PeerConnection::SetAudioPlayout(bool playout) {
RTC_DCHECK(ConfiguredForMedia());
if (!worker_thread()->IsCurrent()) {
worker_thread()->BlockingCall(
[this, playout] { SetAudioPlayout(playout); });
return;
}
RTC_DCHECK_RUN_ON(worker_thread());
auto audio_state = media_engine()->voice().GetAudioState();
audio_state->SetPlayout(playout);
}
void PeerConnection::SetAudioRecording(bool recording) {
RTC_DCHECK(ConfiguredForMedia());
if (!worker_thread()->IsCurrent()) {
worker_thread()->BlockingCall(
[this, recording] { SetAudioRecording(recording); });
return;
}
RTC_DCHECK_RUN_ON(worker_thread());
auto audio_state = media_engine()->voice().GetAudioState();
audio_state->SetRecording(recording);
}
void PeerConnection::AddAdaptationResource(scoped_refptr<Resource> resource) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->BlockingCall(
[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_->is_configured_for_media();
}
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
return worker_thread()->BlockingCall(
[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 = 5000;
if (trials().IsDisabled("WebRTC-RtcEventLogNewFormat")) {
output_period_ms = RtcEventLog::kImmediateOutput;
}
return StartRtcEventLog(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog() {
worker_thread()->BlockingCall([this] { StopRtcEventLog_w(); });
}
void PeerConnection::SetDataChannelEventObserver(
std::unique_ptr<DataChannelEventObserverInterface> observer) {
network_thread()->PostTask(SafeTask(
network_thread_safety_, [this, obs = std::move(observer)]() mutable {
RTC_DCHECK_RUN_ON(network_thread());
data_channel_controller_.SetEventObserver(std::move(obs));
}));
}
scoped_refptr<DtlsTransportInterface> PeerConnection::LookupDtlsTransportByMid(
const std::string& mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->LookupDtlsTransportByMid_n(mid);
}
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 {
return HandleSessionDescriptionAccessor<&SdpStateProvider::local_description>(
local_description_clone_);
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
return HandleSessionDescriptionAccessor<
&SdpStateProvider::remote_description>(remote_description_clone_);
}
const SessionDescriptionInterface* PeerConnection::current_local_description()
const {
return HandleSessionDescriptionAccessor<
&SdpStateProvider::current_local_description>(
current_local_description_clone_);
}
const SessionDescriptionInterface* PeerConnection::current_remote_description()
const {
return HandleSessionDescriptionAccessor<
&SdpStateProvider::current_remote_description>(
current_remote_description_clone_);
}
const SessionDescriptionInterface* PeerConnection::pending_local_description()
const {
return HandleSessionDescriptionAccessor<
&SdpStateProvider::pending_local_description>(
pending_local_description_clone_);
}
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
const {
return HandleSessionDescriptionAccessor<
&SdpStateProvider::pending_remote_description>(
pending_remote_description_clone_);
}
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;
RunWithObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
observer->OnIceConnectionChange(ice_connection_state_);
});
standardized_ice_connection_state_ =
PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
RunWithObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
observer->OnConnectionChange(connection_state_);
});
sdp_handler_->Close();
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
if (ConfiguredForMedia()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
if (!transceiver->stopped())
transceiver->StopInternal();
}
}
// Ensure that all asynchronous stats requests are completed before destroying
// the transport controller below.
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.
// TODO(tommi): The voice/video channels will be partially uninitialized on
// the network thread (see `RtpTransceiver::ClearChannel`), partially on the
// worker thread (see `PushNewMediaChannelAndDeleteChannel`) and then
// eventually freed on the signaling thread.
// It would be good to combine those steps with the teardown steps here.
std::vector<absl::AnyInvocable<void() &&>> network_tasks;
std::vector<absl::AnyInvocable<void() &&>> worker_tasks;
sdp_handler_->GetMediaChannelTeardownTasks(network_tasks, worker_tasks);
CloseOnNetworkThread(network_tasks);
// 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();
}
worker_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(worker_thread());
for (auto& task : worker_tasks) {
std::move(task)();
task = nullptr;
}
worker_thread_safety_->SetNotAlive();
call_.reset();
StopRtcEventLog_w();
});
ReportUsagePattern();
ReportCloseUsageMetrics();
// Signal shutdown to the sdp handler. This invalidates weak pointers for
// internal pending callbacks.
sdp_handler_->PrepareForShutdown();
data_channel_controller_.PrepareForShutdown();
// The .h file says that observer can be discarded after close() returns.
// Make sure this is true.
observer_ = nullptr;
}
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;
RunWithObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
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;
RunWithObserver([&](auto observer) {
observer->OnStandardizedIceConnectionChange(new_state);
});
}
void PeerConnection::SetConnectionState(
PeerConnectionInterface::PeerConnectionState new_state) {
if (connection_state_ == new_state)
return;
if (IsClosed())
return;
connection_state_ = new_state;
RunWithObserver(
[&](auto observer) { 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 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);
auto transport_infos = remote_description()->description()->transport_infos();
if (!transport_infos.empty()) {
// 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 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));
// Record whether the hash algorithm of the first transport's
// DTLS fingerprint is still using SHA-1.
if (transport_infos[0].description.identity_fingerprint) {
RTC_HISTOGRAM_BOOLEAN(
"WebRTC.PeerConnection.DtlsFingerprintLegacySha1",
absl::EqualsIgnoreCase(
transport_infos[0].description.identity_fingerprint->algorithm,
"sha-1"));
}
}
// Record RtcpMuxPolicy setting.
RtcpMuxPolicyUsage rtcp_mux_policy = kRtcpMuxPolicyUsageMax;
switch (configuration_.rtcp_mux_policy) {
case kRtcpMuxPolicyNegotiate:
rtcp_mux_policy = kRtcpMuxPolicyUsageNegotiate;
break;
case kRtcpMuxPolicyRequire:
rtcp_mux_policy = kRtcpMuxPolicyUsageRequire;
break;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.RtcpMuxPolicy",
rtcp_mux_policy, kRtcpMuxPolicyUsageMax);
switch (local_description()->GetType()) {
case SdpType::kOffer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.Offer.ConnectionEstablished",
sdp_handler_->sdp_munging_type(), SdpMungingType::kMaxValue);
break;
case SdpType::kAnswer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.Answer.ConnectionEstablished",
sdp_handler_->sdp_munging_type(), SdpMungingType::kMaxValue);
break;
case SdpType::kPrAnswer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.PrAnswer.ConnectionEstablished",
sdp_handler_->sdp_munging_type(), SdpMungingType::kMaxValue);
break;
case SdpType::kRollback:
// Rollback does not have SDP so can not be munged.
break;
}
bool negotiated_sctp_snap = false;
const SessionDescription* desc = nullptr;
if (local_description()->GetType() == SdpType::kAnswer ||
local_description()->GetType() == SdpType::kPrAnswer) {
desc = local_description()->description();
} else if (remote_description()->GetType() == SdpType::kAnswer ||
remote_description()->GetType() == SdpType::kPrAnswer) {
desc = remote_description()->description();
}
if (!desc) {
RTC_LOG(LS_INFO) << "Connection established without an answer, local="
<< local_description()->GetType()
<< ", remote=" << remote_description()->GetType();
return;
}
// Below this point, we assume that we have an answer in `desc`
const ContentInfo* sctp_content = GetFirstDataContent(desc);
if (sctp_content && !sctp_content->rejected) {
const SctpDataContentDescription* sctp_desc =
sctp_content->media_description()->as_sctp();
if (sctp_desc) {
negotiated_sctp_snap |= sctp_desc->sctp_init().has_value();
}
}
RTC_HISTOGRAM_BOOLEAN("WebRTC.PeerConnection.NegotiatedSctpSnap",
negotiated_sctp_snap);
// Record congestion control mechanism in use, if any.
// The information is taken from the last seen Answer SDP.
std::optional<RtcpFeedbackType> feedback_type;
for (const auto& content : desc->contents()) {
std::optional<RtcpFeedbackType> this_feedback_type =
content.media_description()->preferred_rtcp_cc_ack_type();
if (this_feedback_type) {
feedback_type = this_feedback_type;
break;
}
}
if (!feedback_type) {
feedback_type = RtcpFeedbackType::NONE;
}
// Note that NONE will be reported for datachannel-only calls.
// Only TRANSPORT_CC and CCFB are currently reported.
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.NegotiatedFeedbackType",
static_cast<int>(*feedback_type),
static_cast<int>(RtcpFeedbackType::MAX));
}
void PeerConnection::ReportCloseUsageMetrics() {
if (!was_ever_connected_) {
return;
}
RTC_DCHECK(local_description());
RTC_DCHECK(sdp_handler_);
switch (local_description()->GetType()) {
case SdpType::kOffer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.Offer.ConnectionClosed",
sdp_handler_->sdp_munging_type(), SdpMungingType::kMaxValue);
break;
case SdpType::kAnswer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.Answer.ConnectionClosed",
sdp_handler_->sdp_munging_type(), SdpMungingType::kMaxValue);
break;
case SdpType::kPrAnswer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.PrAnswer.ConnectionClosed",
sdp_handler_->sdp_munging_type(), SdpMungingType::kMaxValue);
break;
case SdpType::kRollback:
// Rollback does not have SDP so can not be munged.
break;
}
}
void PeerConnection::OnIceGatheringChange(
PeerConnectionInterface::IceGatheringState new_state) {
if (IsClosed()) {
return;
}
ice_gathering_state_ = new_state;
RunWithObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
observer->OnIceGatheringChange(ice_gathering_state_);
});
}
void PeerConnection::OnIceCandidate(std::unique_ptr<IceCandidate> candidate) {
if (IsClosed()) {
return;
}
ReportIceCandidateCollected(candidate->candidate());
ClearStatsCache();
RunWithObserver(
[&](auto observer) { 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;
}
RunWithObserver([&](auto observer) {
observer->OnIceCandidateError(address, port, url, error_code, error_text);
});
}
void PeerConnection::OnIceCandidatesRemoved(
absl::string_view mid,
const std::vector<Candidate>& candidates) {
if (IsClosed()) {
return;
}
for (const Candidate& candidate : candidates) {
IceCandidate c(mid, -1, candidate);
RunWithObserver([&](auto o) { o->OnIceCandidateRemoved(&c); });
}
}
void PeerConnection::OnSelectedCandidatePairChanged(
const CandidatePairChangeEvent& event) {
if (IsClosed()) {
return;
}
if (event.selected_candidate_pair.local_candidate().is_local() &&
event.selected_candidate_pair.remote_candidate().is_local()) {
NoteUsageEvent(UsageEvent::DIRECT_CONNECTION_SELECTED);
}
RunWithObserver([&](auto observer) {
observer->OnIceSelectedCandidatePairChanged(event);
});
}
bool PeerConnection::CreateDataChannelTransport(absl::string_view mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!sctp_mid().has_value() || mid == sctp_mid().value());
RTC_LOG(LS_INFO) << "Creating data channel, mid=" << mid;
std::optional<std::string> transport_name =
network_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(network_thread());
return SetupDataChannelTransport_n(mid);
});
if (!transport_name)
return false;
sctp_mid_s_ = std::string(mid);
SetSctpTransportName(transport_name.value());
return true;
}
void PeerConnection::DestroyDataChannelTransport(RTCError error) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!sctp_mid_s_.has_value()) {
RTC_LOG(LS_INFO) << "Data channel not configured.";
RTC_DCHECK(sctp_transport_name_s_.empty());
return;
}
network_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(network_thread());
TeardownDataChannelTransport_n(error);
});
sctp_mid_s_.reset();
SetSctpTransportName("");
}
void PeerConnection::OnSctpDataChannelStateChanged(
int channel_id,
DataChannelInterface::DataState state) {
RTC_DCHECK_RUN_ON(signaling_thread());
stats_collector_.OnSctpDataChannelStateChanged(channel_id, state);
}
PeerConnection::InitializePortAllocatorResult
PeerConnection::InitializePortAllocator_n(
const ServerAddresses& stun_servers,
const std::vector<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 |= PORTALLOCATOR_ENABLE_SHARED_SOCKET |
PORTALLOCATOR_ENABLE_IPV6 |
PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
if (trials().IsDisabled("WebRTC-IPv6Default")) {
port_allocator_flags &= ~(PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.disable_ipv6_on_wifi) {
port_allocator_flags &= ~(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 |= PORTALLOCATOR_DISABLE_TCP;
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
}
if (configuration.candidate_network_policy ==
kCandidateNetworkPolicyLowCost) {
port_allocator_flags |= 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 |= 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(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 & PORTALLOCATOR_ENABLE_IPV6;
return res;
}
bool PeerConnection::ReconfigurePortAllocator_n(
const ServerAddresses& stun_servers,
const std::vector<RelayServerConfig>& turn_servers,
IceTransportsType type,
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
TurnCustomizer* turn_customizer,
std::optional<int> stun_candidate_keepalive_interval,
bool have_local_description) {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_->SetCandidateFilter(
ConvertIceTransportTypeToCandidateFilter(type));
// 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 (!worker_thread_safety_->alive()) {
return false;
}
return env_.event_log().StartLogging(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog_w() {
RTC_DCHECK_RUN_ON(worker_thread());
env_.event_log().StopLogging();
}
std::optional<SSLRole> PeerConnection::GetSctpSslRole_n() {
RTC_DCHECK_RUN_ON(network_thread());
return sctp_mid_n_ ? transport_controller_->GetDtlsRole(*sctp_mid_n_)
: std::nullopt;
}
bool PeerConnection::GetSslRole(const std::string& content_name,
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 = transport_controller_s()->GetDtlsRole(content_name);
if (dtls_role) {
*role = *dtls_role;
return true;
}
return false;
}
bool PeerConnection::GetTransportDescription(
const SessionDescription* description,
const std::string& content_name,
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(network_thread());
return data_channel_controller_.GetDataChannelStats();
}
std::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 std::optional<std::string>();
}
void PeerConnection::SetSctpTransportName(std::string sctp_transport_name) {
RTC_DCHECK_RUN_ON(signaling_thread());
sctp_transport_name_s_ = std::move(sctp_transport_name);
ClearStatsCache();
}
// RTC_RUN_ON(worker_thread())
MediaEngineInterface* PeerConnection::media_engine() const {
RTC_DCHECK(media_engine_ref_);
return media_engine_ref_->media_engine();
}
std::optional<std::string> PeerConnection::sctp_mid() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sctp_mid_s_;
}
CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive())
return {};
CandidateStatsList candidate_stats_list;
port_allocator_->GetCandidateStatsFromPooledSessions(&candidate_stats_list);
return candidate_stats_list;
}
std::map<std::string, 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 {};
Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, TransportStats> transport_stats_by_name;
for (const std::string& transport_name : transport_names) {
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,
scoped_refptr<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<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 {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_s()->NeedsIceRestart(content_name);
}
void PeerConnection::OnTransportControllerConnectionState(
webrtc::IceConnectionState state) {
switch (state) {
case webrtc::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 webrtc::kIceConnectionFailed:
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
break;
case webrtc::kIceConnectionConnected:
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
"all transports are writable.";
{
std::vector<std::pair<std::string, MediaType>> transceiver_info;
if (ConfiguredForMedia()) {
for (const auto& t : rtp_manager()->transceivers()->List()) {
if (t->internal()->channel()) {
std::optional<std::string> mid = t->mid();
if (mid) {
transceiver_info.emplace_back(*mid, t->media_type());
}
}
}
}
network_thread()->PostTask(
SafeTask(network_thread_safety_,
[this, transceiver_info = std::move(transceiver_info)] {
RTC_DCHECK_RUN_ON(network_thread());
ReportTransportStats(std::move(transceiver_info));
}));
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
break;
case webrtc::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(
absl::string_view transport_name,
const 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 (Candidates::const_iterator citer = candidates.begin();
citer != candidates.end(); ++citer) {
// Use transport_name as the candidate media id.
std::unique_ptr<IceCandidate> candidate(
new IceCandidate(transport_name, sdp_mline_index, *citer));
sdp_handler_->AddLocalIceCandidate(candidate.get());
OnIceCandidate(std::move(candidate));
}
}
void PeerConnection::OnTransportControllerCandidateError(
const IceCandidateErrorEvent& event) {
OnIceCandidateError(event.address, event.port, event.url, event.error_code,
event.error_text);
}
void PeerConnection::OnTransportControllerCandidatesRemoved(
absl::string_view mid,
const std::vector<Candidate>& candidates) {
RTC_DCHECK(!mid.empty());
sdp_handler_->RemoveLocalIceCandidates(mid, candidates);
OnIceCandidatesRemoved(mid, candidates);
}
void PeerConnection::OnTransportControllerCandidateChanged(
const CandidatePairChangeEvent& event) {
OnSelectedCandidatePairChanged(event);
}
void PeerConnection::OnTransportControllerDtlsHandshakeError(
SSLHandshakeError error) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.DtlsHandshakeError",
static_cast<int>(error),
static_cast<int>(SSLHandshakeError::MAX_VALUE));
}
// Returns the media index for a local ice candidate given the content name.
bool PeerConnection::GetLocalCandidateMediaIndex(absl::string_view 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].mid() == 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()->BlockingCall([this] { return GetCallStats(); });
}
RTC_DCHECK_RUN_ON(worker_thread());
Thread::ScopedDisallowBlockingCalls no_blocking_calls;
if (call_) {
return call_->GetStats();
} else {
return Call::Stats();
}
}
std::optional<AudioDeviceModule::Stats> PeerConnection::GetAudioDeviceStats() {
RTC_DCHECK_RUN_ON(worker_thread());
if (context_->is_configured_for_media()) {
return media_engine()->voice().GetAudioDeviceStats();
}
return std::nullopt;
}
std::optional<std::string> PeerConnection::SetupDataChannelTransport_n(
absl::string_view mid) {
sctp_mid_n_ = std::string(mid);
DataChannelTransportInterface* transport =
transport_controller_->GetDataChannelTransport(*sctp_mid_n_);
if (!transport) {
#ifndef WEBRTC_HAVE_SCTP
RTC_LOG(LS_ERROR) << "Data channel transport is not available"
<< " as WebRTC has been compiled without SCTP support "
"(WEBRTC_HAVE_SCTP), mid="
<< mid;
#else
RTC_LOG(LS_ERROR)
<< "Data channel transport is not available for data channels, mid="
<< mid;
#endif
sctp_mid_n_ = std::nullopt;
return std::nullopt;
}
std::optional<std::string> transport_name;
DtlsTransportInternal* dtls_transport =
transport_controller_->GetDtlsTransport(*sctp_mid_n_);
if (dtls_transport) {
transport_name = dtls_transport->transport_name();
} else {
// Make sure we still set a valid string.
transport_name = std::string("");
}
data_channel_controller_.SetupDataChannelTransport_n(transport);
return transport_name;
}
void PeerConnection::TeardownDataChannelTransport_n(RTCError error) {
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(error);
}
// Returns false if bundle is enabled and rtcp_mux is disabled.
bool PeerConnection::ValidateBundleSettings(
const SessionDescription* desc,
const flat_map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
if (bundle_groups_by_mid.empty())
return true;
const ContentInfos& contents = desc->contents();
for (ContentInfos::const_iterator citer = contents.begin();
citer != contents.end(); ++citer) {
const ContentInfo* content = (&*citer);
RTC_DCHECK(content != nullptr);
auto it = bundle_groups_by_mid.find(content->mid());
if (it != bundle_groups_by_mid.end() &&
!(content->rejected || content->bundle_only) &&
content->type == MediaProtocolType::kRtp) {
if (!HasRtcpMuxEnabled(content))
return false;
}
}
// RTCP-MUX is enabled in all the contents.
return true;
}
void PeerConnection::ReportSdpBundleUsage(
const SessionDescriptionInterface& remote_description) {
RTC_DCHECK_RUN_ON(signaling_thread());
bool using_bundle =
remote_description.description()->HasGroup(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()) {
MediaType media_type = content.media_description()->type();
if (media_type == MediaType::AUDIO) {
num_audio_mlines += 1;
} else if (media_type == MediaType::VIDEO) {
num_video_mlines += 1;
} else if (media_type == MediaType::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 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(absl::string_view mid,
const Candidate& candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (candidate.network_type() != ADAPTER_TYPE_UNKNOWN) {
RTC_DLOG(LS_WARNING) << "Using candidate with adapter type set - this "
"should only happen in test";
}
// Clear fields that do not make sense as remote candidates.
Candidate new_candidate(candidate);
new_candidate.set_network_type(ADAPTER_TYPE_UNKNOWN);
new_candidate.set_relay_protocol("");
new_candidate.set_underlying_type_for_vpn(ADAPTER_TYPE_UNKNOWN);
network_thread()->PostTask(SafeTask(
network_thread_safety_,
[this, mid = std::string(mid), candidate = new_candidate] {
RTC_DCHECK_RUN_ON(network_thread());
std::vector<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 {
RunWithMaybeNullObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
usage_pattern_.ReportUsagePattern(observer);
});
}
void PeerConnection::ReportRemoteIceCandidateAdded(const 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_;
}
void PeerConnection::OnTransportControllerGatheringState(
webrtc::IceGatheringState state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (state == webrtc::kIceGatheringGathering) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
} else if (state == webrtc::kIceGatheringComplete) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
} else if (state == webrtc::kIceGatheringNew) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringNew);
} else {
RTC_LOG(LS_ERROR) << "Unknown state received: " << state;
RTC_DCHECK_NOTREACHED();
}
}
// Runs on network_thread().
void PeerConnection::ReportTransportStats(
std::vector<std::pair<std::string, MediaType>> transceiver_info) {
RTC_DCHECK_RUN_ON(network_thread());
TRACE_EVENT0("webrtc", "PeerConnection::ReportTransportStats");
Thread::ScopedDisallowBlockingCalls no_blocking_calls;
flat_map<absl::string_view, std::set<MediaType>>
media_types_by_transport_name;
for (const auto& item : transceiver_info) {
const auto& mid = item.first;
const auto& media_type = item.second;
RtpTransportInternal* transport =
transport_controller_->GetRtpTransport(mid);
if (transport) {
media_types_by_transport_name[transport->transport_name()].insert(
media_type);
}
}
if (sctp_mid_n_) {
DtlsTransportInternal* dtls_transport =
transport_controller_->GetDtlsTransport(*sctp_mid_n_);
if (dtls_transport) {
media_types_by_transport_name[dtls_transport->transport_name()].insert(
MediaType::DATA);
}
}
for (const auto& entry : media_types_by_transport_name) {
TransportStats stats;
if (transport_controller_->GetStats(entry.first, &stats)) {
ReportBestConnectionState(stats);
ReportNegotiatedCiphers(dtls_enabled_, stats, entry.second);
}
}
}
// Walk through the ConnectionInfos to gather best connection usage
// for IPv4 and IPv6.
// static (no member state required)
void PeerConnection::ReportBestConnectionState(const TransportStats& stats) {
for (const TransportChannelStats& channel_stats : stats.channel_stats) {
for (const ConnectionInfo& connection_info :
channel_stats.ice_transport_stats.connection_infos) {
if (!connection_info.best_connection) {
continue;
}
const Candidate& local = connection_info.local_candidate;
const Candidate& remote = connection_info.remote_candidate;
// Increment the counter for IceCandidatePairType.
if (local.protocol() == TCP_PROTOCOL_NAME ||
(local.is_relay() && local.relay_protocol() == TCP_PROTOCOL_NAME)) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
GetIceCandidatePairType(local, remote),
kIceCandidatePairMax);
} else if (local.protocol() == UDP_PROTOCOL_NAME) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
GetIceCandidatePairType(local, remote),
kIceCandidatePairMax);
} else {
RTC_LOG(LS_WARNING) << "ReportBestConnectionState: No histogram for "
<< local.protocol();
}
// 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 TransportStats& stats,
const std::set<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 == kSrtpInvalidCryptoSuite &&
ssl_cipher_suite == kTlsNullWithNullNull) {
return;
}
if (ssl_cipher_suite != kTlsNullWithNullNull) {
for (MediaType media_type : media_types) {
switch (media_type) {
case MediaType::AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
kSslCipherSuiteMaxValue);
break;
case MediaType::VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
kSslCipherSuiteMaxValue);
break;
case MediaType::DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
kSslCipherSuiteMaxValue);
break;
default:
RTC_DCHECK_NOTREACHED();
continue;
}
}
}
uint16_t ssl_peer_signature_algorithm =
stats.channel_stats[0].ssl_peer_signature_algorithm;
if (ssl_peer_signature_algorithm != kSslSignatureAlgorithmUnknown) {
for (MediaType media_type : media_types) {
switch (media_type) {
case MediaType::AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Audio",
ssl_peer_signature_algorithm, kSslSignatureAlgorithmMaxValue);
break;
case MediaType::VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Video",
ssl_peer_signature_algorithm, kSslSignatureAlgorithmMaxValue);
break;
case MediaType::DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Data",
ssl_peer_signature_algorithm, kSslSignatureAlgorithmMaxValue);
break;
default:
RTC_DCHECK_NOTREACHED();
continue;
}
}
}
}
bool PeerConnection::OnTransportChanged(
absl::string_view mid,
RtpTransportInternal* rtp_transport,
scoped_refptr<DtlsTransport> dtls_transport,
DataChannelTransportInterface* data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(!mid.empty());
bool ret = true;
if (ConfiguredForMedia()) {
for (const auto& transceiver :
rtp_manager()->transceivers()->UnsafeList()) {
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(rtp_transport->transport_name())] {
RTC_DCHECK_RUN_ON(signaling_thread());
SetSctpTransportName(std::move(name));
}));
}
}
return ret;
}
void PeerConnection::RunWithObserver(
absl::AnyInvocable<void(PeerConnectionObserver*) &&> task) { // NOLINT
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(observer_);
std::move(task)(observer_);
}
void PeerConnection::RunWithMaybeNullObserver(
absl::AnyInvocable<void(PeerConnectionObserver*) &&> task) // NOLINT
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::move(task)(observer_);
}
RTCError PeerConnection::StartSctpTransport(const SctpOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(sctp_mid_s_);
network_thread()->PostTask(
SafeTask(network_thread_safety_, [this, mid = *sctp_mid_s_, options] {
scoped_refptr<SctpTransport> sctp_transport =
transport_controller_n()->GetSctpTransport(mid);
if (sctp_transport)
sctp_transport->Start(options);
}));
return RTCError::OK();
}
CryptoOptions PeerConnection::GetCryptoOptions() {
RTC_DCHECK_RUN_ON(signaling_thread());
return configuration_.crypto_options;
}
void PeerConnection::ClearStatsCache() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (legacy_stats_) {
legacy_stats_->InvalidateCache();
}
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);
}
int PeerConnection::FeedbackAccordingToRfc8888CountForTesting() const {
return worker_thread()->BlockingCall([this]() {
RTC_DCHECK_RUN_ON(worker_thread());
return call_->FeedbackAccordingToRfc8888Count().value_or(0);
});
}
int PeerConnection::FeedbackAccordingToTransportCcCountForTesting() const {
return worker_thread()->BlockingCall([this]() {
RTC_DCHECK_RUN_ON(worker_thread());
return call_->FeedbackAccordingToTransportCcCount().value_or(0);
});
}
absl::AnyInvocable<void(const CopyOnWriteBuffer& packet, int64_t packet_time_us)
const>
PeerConnection::InitializeRtcpCallback() {
return [this](const CopyOnWriteBuffer& packet, int64_t /*packet_time_us*/) {
worker_thread()->PostTask(SafeTask(worker_thread_safety_, [this, packet]() {
call_ptr_->Receiver()->DeliverRtcpPacket(packet);
}));
};
}
absl::AnyInvocable<void(const RtpPacketReceived& parsed_packet) const>
PeerConnection::InitializeUnDemuxablePacketHandler() {
return [this](const RtpPacketReceived& parsed_packet) {
worker_thread()->PostTask(
SafeTask(worker_thread_safety_, [this, parsed_packet]() {
// Deliver the packet anyway to Call to allow Call to do BWE.
// Even if there is no media receiver, the packet has still
// been received on the network and has been correcly parsed.
call_ptr_->Receiver()->DeliverRtpPacket(
MediaType::ANY, parsed_packet,
/*undemuxable_packet_handler=*/
[](const RtpPacketReceived& packet) { return false; });
}));
};
}
bool PeerConnection::CanAttemptDtlsStunPiggybacking() {
return dtls_enabled_ &&
env_.field_trials().IsEnabled("WebRTC-IceHandshakeDtls");
}
void PeerConnection::RunOnSignalingThread(absl::AnyInvocable<void() &&> task) {
if (signaling_thread()->IsCurrent()) {
std::move(task)();
} else {
// Consider if we can use PostTask instead in some situations:
// signaling_thread()->PostTask(
// SafeTask(signaling_thread_safety_.flag(), std::move(task)));
// Currently we use BlockingCall() to be compatible with how the
// api proxies work by default.
signaling_thread()->BlockingCall([&] { std::move(task)(); });
}
}
} // namespace webrtc