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webrtc_m130/pc/peer_connection.cc
Philipp Hancke bce57cda1e Piggyback DTLS handshake in initial STUN packets 
This change puts the DTLS handshake as payload of STUN packets with a custom STUN attribute (registered with the IANA) and starts the DTLS handshake before the ICE transport becomes writable. Effectively, STUN acts as a transport layer for DTLS during the handshake phase.

This will theoretically reduce the call setup time by one RTT for aggressive nomination or two RTTs for regular nomination.

The latest DTLS packet (flight) is cached and sent on every STUN request or response. DTLS packets are extracted from every authenticated STUN request or response and handled to the DTLS layer for processing.
The caching also increases the resilience to packet loss as STUN pacing is more aggressive (every 20ms) than the exponential backoff used by DTLS which should reduce call setup time in lossy networks.

If the other side of the connection does not support this feature the fallback to normal DTLS happens as soon as the ICE transport becomes writable. This also handles edge-cases like fragmentation of the DTLS handshake.

The feature is only supported when ECDSA certificates are used since RSA certificates are too large to transport as STUN attributes. The observed attributes for the server and client flights with the certificates were around 600 to 650 bytes. This may be further reduced by using raw public keys defined in RFC 7250.

This feature is disabled by default and guarded by the field trial
  WebRTC-IceHandshakeDtls
and requires experimentation and standardization before roll-out in the browser.

Parts of this landed in
  https://webrtc-review.googlesource.com/c/src/+/370679

BUG=webrtc:367395350

Change-Id: I4809438b2a267c4690a9b2bd6f1766d2f959500d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362480
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43742}
2025-01-15 07:54:23 -08:00

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/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/peer_connection.h"
#include <limits.h>
#include <stddef.h>
#include <cstdint>
#include <functional>
#include <map>
#include <memory>
#include <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/match.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_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/environment/environment.h"
#include "api/jsep.h"
#include "api/jsep_ice_candidate.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_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 "media/base/codec.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/basic_async_resolver_factory.h"
#include "p2p/base/connection_info.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/p2p_transport_channel.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/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/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/session_description.h"
#include "pc/simulcast_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/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/string_encode.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"
using cricket::ContentInfo;
using cricket::ContentInfos;
using cricket::MediaContentDescription;
using cricket::MediaProtocolType;
using cricket::RidDescription;
using cricket::RidDirection;
using cricket::SessionDescription;
using cricket::SimulcastDescription;
using cricket::SimulcastLayer;
using cricket::SimulcastLayerList;
using cricket::StreamParams;
using cricket::TransportInfo;
namespace webrtc {
namespace {
static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
uint32_t ConvertIceTransportTypeToCandidateFilter(
PeerConnectionInterface::IceTransportsType type) {
switch (type) {
case PeerConnectionInterface::kNone:
return cricket::CF_NONE;
case PeerConnectionInterface::kRelay:
return cricket::CF_RELAY;
case PeerConnectionInterface::kNoHost:
return (cricket::CF_ALL & ~cricket::CF_HOST);
case PeerConnectionInterface::kAll:
return cricket::CF_ALL;
default:
RTC_DCHECK_NOTREACHED();
}
return cricket::CF_NONE;
}
IceCandidatePairType GetIceCandidatePairCounter(
const cricket::Candidate& local,
const cricket::Candidate& remote) {
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;
}
std::optional<int> RTCConfigurationToIceConfigOptionalInt(
int rtc_configuration_parameter) {
if (rtc_configuration_parameter ==
PeerConnectionInterface::RTCConfiguration::kUndefined) {
return std::nullopt;
}
return rtc_configuration_parameter;
}
// Check if the changes of IceTransportsType motives an ice restart.
bool NeedIceRestart(bool surface_ice_candidates_on_ice_transport_type_changed,
PeerConnectionInterface::IceTransportsType current,
PeerConnectionInterface::IceTransportsType modified) {
if (current == modified) {
return false;
}
if (!surface_ice_candidates_on_ice_transport_type_changed) {
return true;
}
auto current_filter = ConvertIceTransportTypeToCandidateFilter(current);
auto modified_filter = ConvertIceTransportTypeToCandidateFilter(modified);
// If surface_ice_candidates_on_ice_transport_type_changed is true and we
// extend the filter, then no ice restart is needed.
return (current_filter & modified_filter) != current_filter;
}
cricket::IceConfig ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) {
cricket::ContinualGatheringPolicy gathering_policy;
switch (config.continual_gathering_policy) {
case PeerConnectionInterface::GATHER_ONCE:
gathering_policy = cricket::GATHER_ONCE;
break;
case PeerConnectionInterface::GATHER_CONTINUALLY:
gathering_policy = cricket::GATHER_CONTINUALLY;
break;
default:
RTC_DCHECK_NOTREACHED();
gathering_policy = cricket::GATHER_ONCE;
}
cricket::IceConfig ice_config;
ice_config.receiving_timeout = RTCConfigurationToIceConfigOptionalInt(
config.ice_connection_receiving_timeout);
ice_config.prioritize_most_likely_candidate_pairs =
config.prioritize_most_likely_ice_candidate_pairs;
ice_config.backup_connection_ping_interval =
RTCConfigurationToIceConfigOptionalInt(
config.ice_backup_candidate_pair_ping_interval);
ice_config.continual_gathering_policy = gathering_policy;
ice_config.presume_writable_when_fully_relayed =
config.presume_writable_when_fully_relayed;
ice_config.surface_ice_candidates_on_ice_transport_type_changed =
config.surface_ice_candidates_on_ice_transport_type_changed;
ice_config.ice_check_interval_strong_connectivity =
config.ice_check_interval_strong_connectivity;
ice_config.ice_check_interval_weak_connectivity =
config.ice_check_interval_weak_connectivity;
ice_config.ice_check_min_interval = config.ice_check_min_interval;
ice_config.ice_unwritable_timeout = config.ice_unwritable_timeout;
ice_config.ice_unwritable_min_checks = config.ice_unwritable_min_checks;
ice_config.ice_inactive_timeout = config.ice_inactive_timeout;
ice_config.stun_keepalive_interval = config.stun_candidate_keepalive_interval;
ice_config.network_preference = config.network_preference;
ice_config.stable_writable_connection_ping_interval =
config.stable_writable_connection_ping_interval_ms;
ice_config.dtls_handshake_in_stun =
false; // Filled in later based on field trial.
return ice_config;
}
// Ensures the configuration doesn't have any parameters with invalid values,
// or values that conflict with other parameters.
//
// Returns RTCError::OK() if there are no issues.
RTCError ValidateConfiguration(
const PeerConnectionInterface::RTCConfiguration& config) {
return cricket::P2PTransportChannel::ValidateIceConfig(
ParseIceConfig(config));
}
// 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.servers = configuration.servers;
modified_config.type = configuration.type;
modified_config.ice_candidate_pool_size =
configuration.ice_candidate_pool_size;
modified_config.prune_turn_ports = configuration.prune_turn_ports;
modified_config.turn_port_prune_policy = configuration.turn_port_prune_policy;
modified_config.surface_ice_candidates_on_ice_transport_type_changed =
configuration.surface_ice_candidates_on_ice_transport_type_changed;
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
modified_config.ice_check_interval_strong_connectivity =
configuration.ice_check_interval_strong_connectivity;
modified_config.ice_check_interval_weak_connectivity =
configuration.ice_check_interval_weak_connectivity;
modified_config.ice_unwritable_timeout = configuration.ice_unwritable_timeout;
modified_config.ice_unwritable_min_checks =
configuration.ice_unwritable_min_checks;
modified_config.ice_inactive_timeout = configuration.ice_inactive_timeout;
modified_config.stun_candidate_keepalive_interval =
configuration.stun_candidate_keepalive_interval;
modified_config.turn_customizer = configuration.turn_customizer;
modified_config.network_preference = configuration.network_preference;
modified_config.active_reset_srtp_params =
configuration.active_reset_srtp_params;
modified_config.turn_logging_id = configuration.turn_logging_id;
modified_config.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 = ValidateConfiguration(modified_config);
if (!err.ok()) {
return err;
}
return modified_config;
}
bool HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
return content->media_description()->rtcp_mux();
}
bool DtlsEnabled(const PeerConnectionInterface::RTCConfiguration& configuration,
const PeerConnectionFactoryInterface::Options& options,
const PeerConnectionDependencies& dependencies) {
if (options.disable_encryption)
return false;
// Enable DTLS by default if we have an identity store or a certificate.
return (dependencies.cert_generator || !configuration.certificates.empty());
}
// Calls `ParseIceServersOrError` to extract ice server information from the
// `configuration` and then validates the extracted configuration. For a
// non-empty list of servers, usage gets recorded via `usage_pattern`.
RTCError ParseAndValidateIceServersFromConfiguration(
const PeerConnectionInterface::RTCConfiguration& configuration,
cricket::ServerAddresses& stun_servers,
std::vector<cricket::RelayServerConfig>& turn_servers,
UsagePattern& usage_pattern) {
RTC_DCHECK(stun_servers.empty());
RTC_DCHECK(turn_servers.empty());
RTCError err = ParseIceServersOrError(configuration.servers, &stun_servers,
&turn_servers);
if (!err.ok()) {
return err;
}
// Restrict number of TURN servers.
if (turn_servers.size() > cricket::kMaxTurnServers) {
RTC_LOG(LS_WARNING) << "Number of configured TURN servers is "
<< turn_servers.size()
<< " which exceeds the maximum allowed number of "
<< cricket::kMaxTurnServers;
turn_servers.resize(cricket::kMaxTurnServers);
}
// Add the turn logging id to all turn servers
for (cricket::RelayServerConfig& turn_server : turn_servers) {
turn_server.turn_logging_id = configuration.turn_logging_id;
}
// Note if STUN or TURN servers were supplied.
if (!stun_servers.empty()) {
usage_pattern.NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
usage_pattern.NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
return RTCError::OK();
}
} // namespace
bool PeerConnectionInterface::RTCConfiguration::operator==(
const PeerConnectionInterface::RTCConfiguration& o) const {
// This static_assert prevents us from accidentally breaking operator==.
// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
struct stuff_being_tested_for_equality {
IceServers servers;
IceTransportsType type;
BundlePolicy bundle_policy;
RtcpMuxPolicy rtcp_mux_policy;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
int ice_candidate_pool_size;
bool disable_ipv6_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 cricket::MediaConfig media_config;
bool prune_turn_ports;
PortPrunePolicy turn_port_prune_policy;
bool presume_writable_when_fully_relayed;
bool enable_ice_renomination;
bool redetermine_role_on_ice_restart;
bool surface_ice_candidates_on_ice_transport_type_changed;
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<rtc::AdapterType> network_preference;
bool active_reset_srtp_params;
std::optional<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<rtc::NetworkMask> vpn_list;
PortAllocatorConfig port_allocator_config;
std::optional<TimeDelta> pacer_burst_interval;
};
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 &&
active_reset_srtp_params == o.active_reset_srtp_params &&
crypto_options == o.crypto_options &&
offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
turn_logging_id == o.turn_logging_id &&
enable_implicit_rollback == o.enable_implicit_rollback &&
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;
}
bool PeerConnectionInterface::RTCConfiguration::operator!=(
const PeerConnectionInterface::RTCConfiguration& o) const {
return !(*this == o);
}
RTCErrorOr<rtc::scoped_refptr<PeerConnection>> PeerConnection::Create(
const Environment& env,
rtc::scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
std::unique_ptr<Call> call,
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
// TODO(https://crbug.com/webrtc/13528): Remove support for kPlanB.
if (configuration.sdp_semantics == SdpSemantics::kPlanB_DEPRECATED) {
RTC_LOG(LS_WARNING)
<< "PeerConnection constructed with legacy SDP semantics!";
}
RTCError config_error = ValidateConfiguration(configuration);
if (!config_error.ok()) {
RTC_LOG(LS_ERROR) << "Invalid ICE configuration: "
<< config_error.message();
return config_error;
}
if (!dependencies.allocator) {
RTC_LOG(LS_ERROR)
<< "PeerConnection initialized without a PortAllocator? "
"This shouldn't happen if using PeerConnectionFactory.";
return RTCError(
RTCErrorType::INVALID_PARAMETER,
"Attempt to create a PeerConnection without a PortAllocatorFactory");
}
if (!dependencies.observer) {
// TODO(deadbeef): Why do we do this?
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
"PeerConnectionObserver";
return RTCError(RTCErrorType::INVALID_PARAMETER,
"Attempt to create a PeerConnection without an observer");
}
bool is_unified_plan =
configuration.sdp_semantics == SdpSemantics::kUnifiedPlan;
bool dtls_enabled = DtlsEnabled(configuration, options, dependencies);
if (!dependencies.async_dns_resolver_factory) {
dependencies.async_dns_resolver_factory =
std::make_unique<BasicAsyncDnsResolverFactory>();
}
// The PeerConnection constructor consumes some, but not all, dependencies.
auto pc = rtc::make_ref_counted<PeerConnection>(
env, context, options, is_unified_plan, std::move(call), dependencies,
dtls_enabled);
RTCError init_error = pc->Initialize(configuration, std::move(dependencies));
if (!init_error.ok()) {
RTC_LOG(LS_ERROR) << "PeerConnection initialization failed";
return init_error;
}
return pc;
}
PeerConnection::PeerConnection(
const Environment& env,
rtc::scoped_refptr<ConnectionContext> context,
const PeerConnectionFactoryInterface::Options& options,
bool is_unified_plan,
std::unique_ptr<Call> call,
PeerConnectionDependencies& dependencies,
bool dtls_enabled)
: env_(env),
context_(context),
options_(options),
observer_(dependencies.observer),
is_unified_plan_(is_unified_plan),
async_dns_resolver_factory_(
std::move(dependencies.async_dns_resolver_factory)),
port_allocator_(std::move(dependencies.allocator)),
ice_transport_factory_(std::move(dependencies.ice_transport_factory)),
tls_cert_verifier_(std::move(dependencies.tls_cert_verifier)),
call_(std::move(call)),
worker_thread_safety_(PendingTaskSafetyFlag::CreateAttachedToTaskQueue(
/*alive=*/call_ != nullptr,
worker_thread())),
call_ptr_(call_.get()),
// RFC 3264: The numeric value of the session id and version in the
// o line MUST be representable with a "64 bit signed integer".
// Due to this constraint session id `session_id_` is max limited to
// LLONG_MAX.
session_id_(rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX)),
dtls_enabled_(dtls_enabled),
data_channel_controller_(this),
message_handler_(signaling_thread()),
weak_factory_(this) {
// Field trials specific to the peerconnection should be owned by the `env`,
RTC_DCHECK(dependencies.trials == nullptr);
}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK_RUN_ON(signaling_thread());
if (sdp_handler_) {
sdp_handler_->PrepareForShutdown();
}
// In case `Close()` wasn't called, always make sure the controller cancels
// potentially pending operations.
data_channel_controller_.PrepareForShutdown();
// Need to stop transceivers before destroying the stats collector because
// AudioRtpSender has a reference to the LegacyStatsCollector it will update
// when stopping.
if (rtp_manager()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
transceiver->StopInternal();
}
}
legacy_stats_.reset(nullptr);
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
stats_collector_ = nullptr;
}
if (sdp_handler_) {
// Don't destroy BaseChannels until after stats has been cleaned up so that
// the last stats request can still read from the channels.
sdp_handler_->DestroyMediaChannels();
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
sdp_handler_->ResetSessionDescFactory();
}
// port_allocator_ and transport_controller_ live on the network thread and
// should be destroyed there.
transport_controller_copy_ = nullptr;
network_thread()->BlockingCall([this] {
RTC_DCHECK_RUN_ON(network_thread());
TeardownDataChannelTransport_n(RTCError::OK());
transport_controller_.reset();
port_allocator_.reset();
if (network_thread_safety_)
network_thread_safety_->SetNotAlive();
});
sctp_mid_s_.reset();
SetSctpTransportName("");
// call_ must be destroyed on the worker thread.
worker_thread()->BlockingCall([this] {
RTC_DCHECK_RUN_ON(worker_thread());
worker_thread_safety_->SetNotAlive();
call_.reset();
});
data_channel_controller_.PrepareForShutdown();
}
RTCError PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
RTCError parse_error = ParseAndValidateIceServersFromConfiguration(
configuration, stun_servers, turn_servers, usage_pattern_);
if (!parse_error.ok()) {
return parse_error;
}
// Network thread initialization.
transport_controller_copy_ = network_thread()->BlockingCall([&] {
RTC_DCHECK_RUN_ON(network_thread());
network_thread_safety_ = PendingTaskSafetyFlag::Create();
InitializePortAllocatorResult pa_result =
InitializePortAllocator_n(stun_servers, turn_servers, configuration);
// Send information about IPv4/IPv6 status.
PeerConnectionAddressFamilyCounter address_family =
pa_result.enable_ipv6 ? kPeerConnection_IPv6 : kPeerConnection_IPv4;
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
kPeerConnectionAddressFamilyCounter_Max);
return InitializeTransportController_n(configuration, dependencies);
});
if (call_ptr_) {
worker_thread()->BlockingCall([this, tc = transport_controller_copy_] {
RTC_DCHECK_RUN_ON(worker_thread());
call_->SetPayloadTypeSuggester(tc);
});
}
configuration_ = configuration;
legacy_stats_ = std::make_unique<LegacyStatsCollector>(this);
stats_collector_ = RTCStatsCollector::Create(this, env_);
sdp_handler_ =
SdpOfferAnswerHandler::Create(this, configuration, dependencies,
context_.get(), transport_controller_copy_);
rtp_manager_ = std::make_unique<RtpTransmissionManager>(
env_, IsUnifiedPlan(), context_.get(), &usage_pattern_, observer_,
legacy_stats_.get(), [this]() {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->UpdateNegotiationNeeded();
});
// Add default audio/video transceivers for Plan B SDP.
if (!IsUnifiedPlan()) {
rtp_manager()->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), rtc::make_ref_counted<RtpTransceiver>(
cricket::MEDIA_TYPE_AUDIO, context())));
rtp_manager()->transceivers()->Add(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), rtc::make_ref_counted<RtpTransceiver>(
cricket::MEDIA_TYPE_VIDEO, context())));
}
int delay_ms = configuration.report_usage_pattern_delay_ms
? *configuration.report_usage_pattern_delay_ms
: REPORT_USAGE_PATTERN_DELAY_MS;
message_handler_.RequestUsagePatternReport(
[this]() {
RTC_DCHECK_RUN_ON(signaling_thread());
ReportUsagePattern();
},
delay_ms);
return RTCError::OK();
}
JsepTransportController* PeerConnection::InitializeTransportController_n(
const RTCConfiguration& configuration,
const PeerConnectionDependencies& dependencies) {
JsepTransportController::Config config;
config.redetermine_role_on_ice_restart =
configuration.redetermine_role_on_ice_restart;
config.ssl_max_version = options_.ssl_max_version;
config.disable_encryption = options_.disable_encryption;
config.bundle_policy = configuration.bundle_policy;
config.rtcp_mux_policy = configuration.rtcp_mux_policy;
// TODO(bugs.webrtc.org/9891) - Remove options_.crypto_options then remove
// this stub.
config.crypto_options = configuration.crypto_options.has_value()
? *configuration.crypto_options
: options_.crypto_options;
config.transport_observer = this;
config.rtcp_handler = InitializeRtcpCallback();
config.un_demuxable_packet_handler = InitializeUnDemuxablePacketHandler();
config.event_log = &env_.event_log();
#if defined(ENABLE_EXTERNAL_AUTH)
config.enable_external_auth = true;
#endif
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
// DTLS has to be enabled to use SCTP.
if (dtls_enabled_) {
config.sctp_factory = context_->sctp_transport_factory();
}
config.ice_transport_factory = ice_transport_factory_.get();
config.on_dtls_handshake_error_ =
[weak_ptr = weak_factory_.GetWeakPtr()](rtc::SSLHandshakeError s) {
if (weak_ptr) {
weak_ptr->OnTransportControllerDtlsHandshakeError(s);
}
};
transport_controller_.reset(
new JsepTransportController(env_, network_thread(), port_allocator_.get(),
async_dns_resolver_factory_.get(),
payload_type_picker_, std::move(config)));
transport_controller_->SubscribeIceConnectionState(
[this](cricket::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);
}));
});
transport_controller_->SubscribeConnectionState(
[this](PeerConnectionInterface::PeerConnectionState s) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
SetConnectionState(s);
}));
});
transport_controller_->SubscribeStandardizedIceConnectionState(
[this](PeerConnectionInterface::IceConnectionState s) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
SetStandardizedIceConnectionState(s);
}));
});
transport_controller_->SubscribeIceGatheringState(
[this](cricket::IceGatheringState s) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, s]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerGatheringState(s);
}));
});
transport_controller_->SubscribeIceCandidateGathered(
[this](const std::string& transport,
const std::vector<cricket::Candidate>& candidates) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(),
[this, t = transport, c = candidates]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidatesGathered(t, c);
}));
});
transport_controller_->SubscribeIceCandidateError(
[this](const cricket::IceCandidateErrorEvent& event) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, event = event]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidateError(event);
}));
});
transport_controller_->SubscribeIceCandidatesRemoved(
[this](const std::vector<cricket::Candidate>& c) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, c = c]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidatesRemoved(c);
}));
});
transport_controller_->SubscribeIceCandidatePairChanged(
[this](const cricket::CandidatePairChangeEvent& event) {
RTC_DCHECK_RUN_ON(network_thread());
signaling_thread()->PostTask(
SafeTask(signaling_thread_safety_.flag(), [this, event = event]() {
RTC_DCHECK_RUN_ON(signaling_thread());
OnTransportControllerCandidateChanged(event);
}));
});
auto ice_config = ParseIceConfig(configuration);
ice_config.dtls_handshake_in_stun =
CanAttemptDtlsStunPiggybacking(configuration);
transport_controller_->SetIceConfig(ice_config);
return transport_controller_.get();
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
"Plan SdpSemantics. Please use GetSenders "
"instead.";
return sdp_handler_->local_streams();
}
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
"Plan SdpSemantics. Please use GetReceivers "
"instead.";
return sdp_handler_->remote_streams();
}
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
"SdpSemantics. Please use AddTrack instead.";
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
if (!ConfiguredForMedia()) {
RTC_LOG(LS_ERROR) << "AddStream: Not configured for media";
return false;
}
return sdp_handler_->AddStream(local_stream);
}
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(ConfiguredForMedia());
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
"Plan SdpSemantics. Please use RemoveTrack "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
sdp_handler_->RemoveStream(local_stream);
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
return AddTrack(std::move(track), stream_ids, nullptr);
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
rtc::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<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
rtc::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() + ".");
}
auto sender_or_error =
rtp_manager()->AddTrack(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(
rtc::scoped_refptr<RtpSenderInterface> sender) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
if (!sender) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
}
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"PeerConnection is closed.");
}
if (IsUnifiedPlan()) {
auto transceiver = FindTransceiverBySender(sender);
if (!transceiver || !sender->track()) {
return RTCError::OK();
}
sender->SetTrack(nullptr);
if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kRecvOnly);
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kInactive);
}
} else {
bool removed;
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
removed = rtp_manager()->GetAudioTransceiver()->internal()->RemoveSender(
sender.get());
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
removed = rtp_manager()->GetVideoTransceiver()->internal()->RemoveSender(
sender.get());
}
if (!removed) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Couldn't find sender " + sender->id() + " to remove.");
}
}
sdp_handler_->UpdateNegotiationNeeded();
return RTCError::OK();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindTransceiverBySender(
rtc::scoped_refptr<RtpSenderInterface> sender) {
return rtp_manager()->transceivers()->FindBySender(sender);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
return AddTransceiver(track, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
RTC_CHECK(IsUnifiedPlan())
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
if (!track) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
}
cricket::MediaType media_type;
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
media_type = cricket::MEDIA_TYPE_AUDIO;
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
media_type = cricket::MEDIA_TYPE_VIDEO;
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Track kind is not audio or video");
}
return AddTransceiver(media_type, track, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
return AddTransceiver(media_type, RtpTransceiverInit());
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(cricket::MediaType media_type,
const RtpTransceiverInit& init) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
RTC_CHECK(IsUnifiedPlan())
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"media type is not audio or video");
}
return AddTransceiver(media_type, nullptr, init);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
cricket::MediaType media_type,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const RtpTransceiverInit& init,
bool update_negotiation_needed) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"Not configured for media");
}
RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO));
if (track) {
RTC_DCHECK_EQ(media_type,
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO));
}
size_t num_rids = absl::c_count_if(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !encoding.rid.empty();
});
if (num_rids > 0 && num_rids != init.send_encodings.size()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"RIDs must be provided for either all or none of the send encodings.");
}
if (num_rids > 0 && absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !IsLegalRsidName(encoding.rid);
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Invalid RID value provided.");
}
if (absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return encoding.ssrc.has_value();
})) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
RtpParameters parameters;
parameters.encodings = init.send_encodings;
// Encodings are dropped from the tail if too many are provided.
size_t max_simulcast_streams =
media_type == cricket::MEDIA_TYPE_VIDEO ? kMaxSimulcastStreams : 1u;
if (parameters.encodings.size() > max_simulcast_streams) {
parameters.encodings.erase(
parameters.encodings.begin() + max_simulcast_streams,
parameters.encodings.end());
}
// Single RID should be removed.
if (parameters.encodings.size() == 1 &&
!parameters.encodings[0].rid.empty()) {
RTC_LOG(LS_INFO) << "Removing RID: " << parameters.encodings[0].rid << ".";
parameters.encodings[0].rid.clear();
}
// If RIDs were not provided, they are generated for simulcast scenario.
if (parameters.encodings.size() > 1 && num_rids == 0) {
rtc::UniqueStringGenerator rid_generator;
for (RtpEncodingParameters& encoding : parameters.encodings) {
encoding.rid = rid_generator.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<cricket::Codec> codecs;
// Gather the current codec capabilities to allow checking scalabilityMode and
// codec selection against supported values.
if (media_type == cricket::MEDIA_TYPE_VIDEO) {
codecs = context_->media_engine()->video().send_codecs(false);
} else {
codecs = context_->media_engine()->voice().send_codecs();
}
auto result = cricket::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 " << cricket::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTransceiver.";
// Set the sender ID equal to the track ID if the track is specified unless
// that sender ID is already in use.
std::string sender_id = (track && !rtp_manager()->FindSenderById(track->id())
? track->id()
: rtc::CreateRandomUuid());
auto sender = rtp_manager()->CreateSender(
media_type, sender_id, track, init.stream_ids, parameters.encodings);
auto receiver =
rtp_manager()->CreateReceiver(media_type, rtc::CreateRandomUuid());
auto transceiver = rtp_manager()->CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_direction(init.direction);
if (update_negotiation_needed) {
sdp_handler_->UpdateNegotiationNeeded();
}
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
}
void PeerConnection::OnNegotiationNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsClosed());
sdp_handler_->UpdateNegotiationNeeded();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
const std::string& kind,
const std::string& stream_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
RTC_LOG(LS_ERROR) << "Not configured for media";
return nullptr;
}
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
"Plan SdpSemantics. Please use AddTransceiver "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
if (IsClosed()) {
return nullptr;
}
// Internally we need to have one stream with Plan B semantics, so we
// generate a random stream ID if not specified.
std::vector<std::string> stream_ids;
if (stream_id.empty()) {
stream_ids.push_back(rtc::CreateRandomUuid());
RTC_LOG(LS_INFO)
<< "No stream_id specified for sender. Generated stream ID: "
<< stream_ids[0];
} else {
stream_ids.push_back(stream_id);
}
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
auto audio_sender =
AudioRtpSender::Create(env_, worker_thread(), rtc::CreateRandomUuid(),
legacy_stats_.get(), rtp_manager());
audio_sender->SetMediaChannel(rtp_manager()->voice_media_send_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), audio_sender);
rtp_manager()->GetAudioTransceiver()->internal()->AddSender(new_sender);
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
auto video_sender = VideoRtpSender::Create(
env_, worker_thread(), rtc::CreateRandomUuid(), rtp_manager());
video_sender->SetMediaChannel(rtp_manager()->video_media_send_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), video_sender);
rtp_manager()->GetVideoTransceiver()->internal()->AddSender(new_sender);
} else {
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
return nullptr;
}
new_sender->internal()->set_stream_ids(stream_ids);
return new_sender;
}
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
if (ConfiguredForMedia()) {
for (const auto& sender : rtp_manager()->GetSendersInternal()) {
ret.push_back(sender);
}
}
return ret;
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
if (ConfiguredForMedia()) {
for (const auto& receiver : rtp_manager()->GetReceiversInternal()) {
ret.push_back(receiver);
}
}
return ret;
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::GetTransceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(IsUnifiedPlan())
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
if (ConfiguredForMedia()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
all_transceivers.push_back(transceiver);
}
}
return all_transceivers;
}
bool PeerConnection::GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats (legacy)");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!observer) {
RTC_LOG(LS_ERROR) << "Legacy GetStats - observer is NULL.";
return false;
}
RTC_LOG_THREAD_BLOCK_COUNT();
legacy_stats_->UpdateStats(level);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(4);
// The LegacyStatsCollector is used to tell if a track is valid because it may
// remember tracks that the PeerConnection previously removed.
if (track && !legacy_stats_->IsValidTrack(track->id())) {
RTC_LOG(LS_WARNING) << "Legacy GetStats is called with an invalid track: "
<< track->id();
return false;
}
message_handler_.PostGetStats(observer, legacy_stats_.get(), track);
return true;
}
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(stats_collector_);
RTC_DCHECK(callback);
RTC_LOG_THREAD_BLOCK_COUNT();
stats_collector_->GetStatsReport(
rtc::scoped_refptr<RTCStatsCollectorCallback>(callback));
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
void PeerConnection::GetStats(
rtc::scoped_refptr<RtpSenderInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_DCHECK(stats_collector_);
RTC_LOG_THREAD_BLOCK_COUNT();
rtc::scoped_refptr<RtpSenderInternal> internal_sender;
if (selector) {
for (const auto& proxy_transceiver :
rtp_manager()->transceivers()->List()) {
for (const auto& proxy_sender :
proxy_transceiver->internal()->senders()) {
if (proxy_sender == selector) {
internal_sender = proxy_sender->internal();
break;
}
}
if (internal_sender)
break;
}
}
// If there is no `internal_sender` then `selector` is either null or does not
// belong to the PeerConnection (in Plan B, senders can be removed from the
// PeerConnection). This means that "all the stats objects representing the
// selector" is an empty set. Invoking GetStatsReport() with a null selector
// produces an empty stats report.
stats_collector_->GetStatsReport(internal_sender, callback);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
void PeerConnection::GetStats(
rtc::scoped_refptr<RtpReceiverInterface> selector,
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(callback);
RTC_DCHECK(stats_collector_);
RTC_LOG_THREAD_BLOCK_COUNT();
rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
if (selector) {
for (const auto& proxy_transceiver :
rtp_manager()->transceivers()->List()) {
for (const auto& proxy_receiver :
proxy_transceiver->internal()->receivers()) {
if (proxy_receiver == selector) {
internal_receiver = proxy_receiver->internal();
break;
}
}
if (internal_receiver)
break;
}
}
// If there is no `internal_receiver` then `selector` is either null or does
// not belong to the PeerConnection (in Plan B, receivers can be removed from
// the PeerConnection). This means that "all the stats objects representing
// the selector" is an empty set. Invoking GetStatsReport() with a null
// selector produces an empty stats report.
stats_collector_->GetStatsReport(internal_receiver, callback);
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(2);
}
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->signaling_state();
}
PeerConnectionInterface::IceConnectionState
PeerConnection::ice_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return ice_connection_state_;
}
PeerConnectionInterface::IceConnectionState
PeerConnection::standardized_ice_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return standardized_ice_connection_state_;
}
PeerConnectionInterface::PeerConnectionState
PeerConnection::peer_connection_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return connection_state_;
}
PeerConnectionInterface::IceGatheringState
PeerConnection::ice_gathering_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return ice_gathering_state_;
}
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().size() < 1) {
return std::nullopt;
}
return description->description()->transport_infos()[0].description.HasOption(
"trickle");
}
RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>>
PeerConnection::CreateDataChannelOrError(const std::string& label,
const DataChannelInit* config) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
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<rtc::scoped_refptr<DataChannelInterface>> ret =
data_channel_controller_.InternalCreateDataChannelWithProxy(
label, internal_config);
if (!ret.ok()) {
return ret.MoveError();
}
rtc::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_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer, desc_ptr);
}
void PeerConnection::SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(std::move(desc), observer);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer);
}
void PeerConnection::SetLocalDescription(
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetLocalDescription(observer);
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetRemoteDescription(observer, desc_ptr);
}
void PeerConnection::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->SetRemoteDescription(std::move(desc), observer);
}
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
RTC_DCHECK_RUN_ON(signaling_thread());
return configuration_;
}
RTCError PeerConnection::SetConfiguration(
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (IsClosed()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
"SetConfiguration: PeerConnection is closed.");
}
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 Ice 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.
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
validate_error = ParseAndValidateIceServersFromConfiguration(
configuration, stun_servers, turn_servers, usage_pattern_);
if (!validate_error.ok()) {
return validate_error;
}
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();
cricket::IceConfig ice_config = ParseIceConfig(modified_config);
ice_config.dtls_handshake_in_stun =
CanAttemptDtlsStunPiggybacking(modified_config);
// 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);
transport_controller_->SetActiveResetSrtpParams(
modified_config.active_reset_srtp_params);
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 IceCandidateInterface* ice_candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
ClearStatsCache();
return sdp_handler_->AddIceCandidate(ice_candidate);
}
void PeerConnection::AddIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate,
std::function<void(RTCError)> callback) {
RTC_DCHECK_RUN_ON(signaling_thread());
sdp_handler_->AddIceCandidate(std::move(candidate),
[this, callback](RTCError result) {
ClearStatsCache();
callback(result);
});
}
bool PeerConnection::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->RemoveIceCandidates(candidates);
}
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->BlockingCall([&]() { return SetBitrate(bitrate); });
}
RTC_DCHECK_RUN_ON(worker_thread());
const bool has_min = bitrate.min_bitrate_bps.has_value();
const bool has_start = bitrate.start_bitrate_bps.has_value();
const bool has_max = bitrate.max_bitrate_bps.has_value();
if (has_min && *bitrate.min_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"min_bitrate_bps <= 0");
}
if (has_start) {
if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"start_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.start_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"curent_bitrate_bps < 0");
}
}
if (has_max) {
if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < start_bitrate_bps");
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.max_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < 0");
}
}
RTC_DCHECK(call_.get());
call_->SetClientBitratePreferences(bitrate);
return RTCError::OK();
}
void PeerConnection::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) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->BlockingCall(
[this, playout] { SetAudioPlayout(playout); });
return;
}
auto audio_state = context_->media_engine()->voice().GetAudioState();
audio_state->SetPlayout(playout);
}
void PeerConnection::SetAudioRecording(bool recording) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->BlockingCall(
[this, recording] { SetAudioRecording(recording); });
return;
}
auto audio_state = context_->media_engine()->voice().GetAudioState();
audio_state->SetRecording(recording);
}
void PeerConnection::AddAdaptationResource(
rtc::scoped_refptr<Resource> resource) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->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_->media_engine();
}
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(); });
}
rtc::scoped_refptr<DtlsTransportInterface>
PeerConnection::LookupDtlsTransportByMid(const std::string& mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
}
rtc::scoped_refptr<DtlsTransport>
PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
// TODO(bugs.webrtc.org/9987): Avoid the thread jump.
// This might be done by caching the value on the signaling thread.
return network_thread()->BlockingCall([this, mid]() {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
});
}
rtc::scoped_refptr<SctpTransportInterface> PeerConnection::GetSctpTransport()
const {
RTC_DCHECK_RUN_ON(network_thread());
if (!sctp_mid_n_)
return nullptr;
return transport_controller_->GetSctpTransport(*sctp_mid_n_);
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->local_description();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->remote_description();
}
const SessionDescriptionInterface* PeerConnection::current_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->current_local_description();
}
const SessionDescriptionInterface* PeerConnection::current_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->current_remote_description();
}
const SessionDescriptionInterface* PeerConnection::pending_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->pending_local_description();
}
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->pending_remote_description();
}
void PeerConnection::Close() {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::Close");
RTC_LOG_THREAD_BLOCK_COUNT();
if (IsClosed()) {
return;
}
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
legacy_stats_->UpdateStats(kStatsOutputLevelStandard);
ice_connection_state_ = PeerConnectionInterface::kIceConnectionClosed;
Observer()->OnIceConnectionChange(ice_connection_state_);
standardized_ice_connection_state_ =
PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
Observer()->OnConnectionChange(connection_state_);
sdp_handler_->Close();
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
if (ConfiguredForMedia()) {
for (const auto& transceiver : rtp_manager()->transceivers()->List()) {
transceiver->internal()->SetPeerConnectionClosed();
if (!transceiver->stopped())
transceiver->StopInternal();
}
}
// Ensure that all asynchronous stats requests are completed before destroying
// the transport controller below.
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
}
// Don't destroy BaseChannels until after stats has been cleaned up so that
// the last stats request can still read from the channels.
// 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.
sdp_handler_->DestroyMediaChannels();
// The event log is used in the transport controller, which must be outlived
// by the former. CreateOffer by the peer connection is implemented
// asynchronously and if the peer connection is closed without resetting the
// WebRTC session description factory, the session description factory would
// call the transport controller.
sdp_handler_->ResetSessionDescFactory();
if (ConfiguredForMedia()) {
rtp_manager_->Close();
}
network_thread()->BlockingCall([this] {
RTC_DCHECK_RUN_ON(network_thread());
TeardownDataChannelTransport_n({});
transport_controller_.reset();
port_allocator_->DiscardCandidatePool();
if (network_thread_safety_) {
network_thread_safety_->SetNotAlive();
}
});
sctp_mid_s_.reset();
SetSctpTransportName("");
worker_thread()->BlockingCall([this] {
RTC_DCHECK_RUN_ON(worker_thread());
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;
Observer()->OnIceConnectionChange(ice_connection_state_);
}
void PeerConnection::SetStandardizedIceConnectionState(
PeerConnectionInterface::IceConnectionState new_state) {
if (standardized_ice_connection_state_ == new_state) {
return;
}
if (IsClosed()) {
return;
}
RTC_LOG(LS_INFO) << "Changing standardized IceConnectionState "
<< standardized_ice_connection_state_ << " => " << new_state;
standardized_ice_connection_state_ = new_state;
Observer()->OnStandardizedIceConnectionChange(new_state);
}
void PeerConnection::SetConnectionState(
PeerConnectionInterface::PeerConnectionState new_state) {
if (connection_state_ == new_state)
return;
if (IsClosed())
return;
connection_state_ = new_state;
Observer()->OnConnectionChange(new_state);
// The first connection state change to connected happens once per
// connection which makes it a good point to report metrics.
if (new_state == PeerConnectionState::kConnected && !was_ever_connected_) {
was_ever_connected_ = true;
ReportFirstConnectUsageMetrics();
}
}
void PeerConnection::ReportFirstConnectUsageMetrics() {
// Record bundle-policy from configuration. Done here from
// connectionStateChange to limit to actually established connections.
BundlePolicyUsage policy = kBundlePolicyUsageMax;
switch (configuration_.bundle_policy) {
case kBundlePolicyBalanced:
policy = kBundlePolicyUsageBalanced;
break;
case kBundlePolicyMaxBundle:
policy = kBundlePolicyUsageMaxBundle;
break;
case kBundlePolicyMaxCompat:
policy = kBundlePolicyUsageMaxCompat;
break;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.BundlePolicy", policy,
kBundlePolicyUsageMax);
// Record 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;
}
}
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;
Observer()->OnIceGatheringChange(ice_gathering_state_);
}
void PeerConnection::OnIceCandidate(
std::unique_ptr<IceCandidateInterface> candidate) {
if (IsClosed()) {
return;
}
ReportIceCandidateCollected(candidate->candidate());
ClearStatsCache();
Observer()->OnIceCandidate(candidate.get());
}
void PeerConnection::OnIceCandidateError(const std::string& address,
int port,
const std::string& url,
int error_code,
const std::string& error_text) {
if (IsClosed()) {
return;
}
Observer()->OnIceCandidateError(address, port, url, error_code, error_text);
}
void PeerConnection::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
if (IsClosed()) {
return;
}
Observer()->OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnSelectedCandidatePairChanged(
const cricket::CandidatePairChangeEvent& event) {
if (IsClosed()) {
return;
}
if (event.selected_candidate_pair.local_candidate().is_local() &&
event.selected_candidate_pair.remote_candidate().is_local()) {
NoteUsageEvent(UsageEvent::DIRECT_CONNECTION_SELECTED);
}
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());
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());
if (stats_collector_)
stats_collector_->OnSctpDataChannelStateChanged(channel_id, state);
}
PeerConnection::InitializePortAllocatorResult
PeerConnection::InitializePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_->Initialize();
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
int port_allocator_flags = port_allocator_->flags();
port_allocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
cricket::PORTALLOCATOR_ENABLE_IPV6 |
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
if (trials().IsDisabled("WebRTC-IPv6Default")) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
}
if (configuration.disable_ipv6_on_wifi) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
}
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
}
if (configuration.candidate_network_policy ==
kCandidateNetworkPolicyLowCost) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
}
if (configuration.disable_link_local_networks) {
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS;
RTC_LOG(LS_INFO) << "Disable candidates on link-local network interfaces.";
}
port_allocator_->set_flags(port_allocator_flags);
// No step delay is used while allocating ports.
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
port_allocator_->SetCandidateFilter(
ConvertIceTransportTypeToCandidateFilter(configuration.type));
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
auto turn_servers_copy = turn_servers;
for (auto& turn_server : turn_servers_copy) {
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
}
// Call this last since it may create pooled allocator sessions using the
// properties set above.
port_allocator_->SetConfiguration(
stun_servers, std::move(turn_servers_copy),
configuration.ice_candidate_pool_size,
configuration.GetTurnPortPrunePolicy(), configuration.turn_customizer,
configuration.stun_candidate_keepalive_interval);
InitializePortAllocatorResult res;
res.enable_ipv6 = port_allocator_flags & cricket::PORTALLOCATOR_ENABLE_IPV6;
return res;
}
bool PeerConnection::ReconfigurePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
IceTransportsType type,
int candidate_pool_size,
PortPrunePolicy turn_port_prune_policy,
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<rtc::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,
rtc::SSLRole* role) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the session.";
return false;
}
auto dtls_role = network_thread()->BlockingCall([this, content_name]() {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->GetDtlsRole(content_name);
});
if (dtls_role) {
*role = *dtls_role;
return true;
}
return false;
}
bool PeerConnection::GetTransportDescription(
const SessionDescription* description,
const std::string& content_name,
cricket::TransportDescription* tdesc) {
if (!description || !tdesc) {
return false;
}
const TransportInfo* transport_info =
description->GetTransportInfoByName(content_name);
if (!transport_info) {
return false;
}
*tdesc = transport_info->description;
return true;
}
std::vector<DataChannelStats> PeerConnection::GetDataChannelStats() const {
RTC_DCHECK_RUN_ON(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();
}
std::optional<std::string> PeerConnection::sctp_mid() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sctp_mid_s_;
}
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive())
return {};
cricket::CandidateStatsList candidate_stats_list;
port_allocator_->GetCandidateStatsFromPooledSessions(&candidate_stats_list);
return candidate_stats_list;
}
std::map<std::string, cricket::TransportStats>
PeerConnection::GetTransportStatsByNames(
const std::set<std::string>& transport_names) {
TRACE_EVENT0("webrtc", "PeerConnection::GetTransportStatsByNames");
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive())
return {};
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, cricket::TransportStats> transport_stats_by_name;
for (const std::string& transport_name : transport_names) {
cricket::TransportStats transport_stats;
bool success =
transport_controller_->GetStats(transport_name, &transport_stats);
if (success) {
transport_stats_by_name[transport_name] = std::move(transport_stats);
} else {
RTC_LOG(LS_ERROR) << "Failed to get transport stats for transport_name="
<< transport_name;
}
}
return transport_stats_by_name;
}
bool PeerConnection::GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
RTC_DCHECK_RUN_ON(network_thread());
if (!network_thread_safety_->alive() || !certificate) {
return false;
}
*certificate = transport_controller_->GetLocalCertificate(transport_name);
return *certificate != nullptr;
}
std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
const std::string& transport_name) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->GetRemoteSSLCertChain(transport_name);
}
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->IceRestartPending(content_name);
}
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
return network_thread()->BlockingCall([this, &content_name] {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_->NeedsIceRestart(content_name);
});
}
void PeerConnection::OnTransportControllerConnectionState(
cricket::IceConnectionState state) {
switch (state) {
case cricket::kIceConnectionConnecting:
// If the current state is Connected or Completed, then there were
// writable channels but now there are not, so the next state must
// be Disconnected.
// kIceConnectionConnecting is currently used as the default,
// un-connected state by the TransportController, so its only use is
// detecting disconnections.
if (ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionConnected ||
ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionCompleted) {
SetIceConnectionState(
PeerConnectionInterface::kIceConnectionDisconnected);
}
break;
case cricket::kIceConnectionFailed:
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
break;
case cricket::kIceConnectionConnected:
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
"all transports are writable.";
{
std::vector<RtpTransceiverProxyRefPtr> transceivers;
if (ConfiguredForMedia()) {
transceivers = rtp_manager()->transceivers()->List();
}
network_thread()->PostTask(
SafeTask(network_thread_safety_,
[this, transceivers = std::move(transceivers)] {
RTC_DCHECK_RUN_ON(network_thread());
ReportTransportStats(std::move(transceivers));
}));
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
break;
case cricket::kIceConnectionCompleted:
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
"all transports are complete.";
if (ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionConnected) {
// If jumping directly from "checking" to "connected",
// signal "connected" first.
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
break;
default:
RTC_DCHECK_NOTREACHED();
}
}
void PeerConnection::OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const cricket::Candidates& candidates) {
// TODO(bugs.webrtc.org/12427): Expect this to come in on the network thread
// (not signaling as it currently does), handle appropriately.
int sdp_mline_index;
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
RTC_LOG(LS_ERROR)
<< "OnTransportControllerCandidatesGathered: content name "
<< transport_name << " not found";
return;
}
for (cricket::Candidates::const_iterator citer = candidates.begin();
citer != candidates.end(); ++citer) {
// Use transport_name as the candidate media id.
std::unique_ptr<JsepIceCandidate> candidate(
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
sdp_handler_->AddLocalIceCandidate(candidate.get());
OnIceCandidate(std::move(candidate));
}
}
void PeerConnection::OnTransportControllerCandidateError(
const cricket::IceCandidateErrorEvent& event) {
OnIceCandidateError(event.address, event.port, event.url, event.error_code,
event.error_text);
}
void PeerConnection::OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
// Sanity check.
for (const cricket::Candidate& candidate : candidates) {
if (candidate.transport_name().empty()) {
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
"empty content name in candidate "
<< candidate.ToString();
return;
}
}
sdp_handler_->RemoveLocalIceCandidates(candidates);
OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnTransportControllerCandidateChanged(
const cricket::CandidatePairChangeEvent& event) {
OnSelectedCandidatePairChanged(event);
}
void PeerConnection::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
}
// Returns the media index for a local ice candidate given the content name.
bool PeerConnection::GetLocalCandidateMediaIndex(
const std::string& content_name,
int* sdp_mline_index) {
if (!local_description() || !sdp_mline_index) {
return false;
}
bool content_found = false;
const ContentInfos& contents = local_description()->description()->contents();
for (size_t index = 0; index < contents.size(); ++index) {
if (contents[index].name == content_name) {
*sdp_mline_index = static_cast<int>(index);
content_found = true;
break;
}
}
return content_found;
}
Call::Stats PeerConnection::GetCallStats() {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->BlockingCall([this] { return GetCallStats(); });
}
RTC_DCHECK_RUN_ON(worker_thread());
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
if (call_) {
return call_->GetStats();
} else {
return Call::Stats();
}
}
std::optional<AudioDeviceModule::Stats> PeerConnection::GetAudioDeviceStats() {
if (context_->media_engine()) {
return context_->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) {
RTC_LOG(LS_ERROR)
<< "Data channel transport is not available for data channels, mid="
<< mid;
sctp_mid_n_ = std::nullopt;
return std::nullopt;
}
std::optional<std::string> transport_name;
cricket::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 std::map<std::string, const cricket::ContentGroup*>&
bundle_groups_by_mid) {
if (bundle_groups_by_mid.empty())
return true;
const cricket::ContentInfos& contents = desc->contents();
for (cricket::ContentInfos::const_iterator citer = contents.begin();
citer != contents.end(); ++citer) {
const cricket::ContentInfo* content = (&*citer);
RTC_DCHECK(content != NULL);
auto it = bundle_groups_by_mid.find(content->name);
if (it != bundle_groups_by_mid.end() &&
!(content->rejected || content->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(cricket::GROUP_TYPE_BUNDLE);
int num_audio_mlines = 0;
int num_video_mlines = 0;
int num_data_mlines = 0;
for (const ContentInfo& content :
remote_description.description()->contents()) {
cricket::MediaType media_type = content.media_description()->type();
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
num_audio_mlines += 1;
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
num_video_mlines += 1;
} else if (media_type == cricket::MEDIA_TYPE_DATA) {
num_data_mlines += 1;
}
}
bool simple = num_audio_mlines <= 1 && num_video_mlines <= 1;
BundleUsage usage = kBundleUsageMax;
if (num_audio_mlines == 0 && num_video_mlines == 0) {
if (num_data_mlines > 0) {
usage = using_bundle ? kBundleUsageBundleDatachannelOnly
: kBundleUsageNoBundleDatachannelOnly;
} else {
usage = kBundleUsageEmpty;
}
} else if (configuration_.sdp_semantics == SdpSemantics::kPlanB_DEPRECATED) {
// In plan-b, simple/complex usage will not show up in the number of
// m-lines or BUNDLE.
usage = using_bundle ? kBundleUsageBundlePlanB : kBundleUsageNoBundlePlanB;
} else {
if (simple) {
usage =
using_bundle ? kBundleUsageBundleSimple : kBundleUsageNoBundleSimple;
} else {
usage = using_bundle ? kBundleUsageBundleComplex
: kBundleUsageNoBundleComplex;
}
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.BundleUsage", usage,
kBundleUsageMax);
}
void PeerConnection::ReportIceCandidateCollected(
const cricket::Candidate& candidate) {
NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
if (candidate.address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
}
if (candidate.address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::MDNS_CANDIDATE_COLLECTED);
}
if (candidate.address().family() == AF_INET6) {
NoteUsageEvent(UsageEvent::IPV6_CANDIDATE_COLLECTED);
}
}
void PeerConnection::NoteUsageEvent(UsageEvent event) {
RTC_DCHECK_RUN_ON(signaling_thread());
usage_pattern_.NoteUsageEvent(event);
}
// Asynchronously adds remote candidates on the network thread.
void PeerConnection::AddRemoteCandidate(const std::string& mid,
const cricket::Candidate& candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (candidate.network_type() != rtc::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.
cricket::Candidate new_candidate(candidate);
new_candidate.set_network_type(rtc::ADAPTER_TYPE_UNKNOWN);
new_candidate.set_relay_protocol("");
new_candidate.set_underlying_type_for_vpn(rtc::ADAPTER_TYPE_UNKNOWN);
network_thread()->PostTask(SafeTask(
network_thread_safety_, [this, mid = mid, candidate = new_candidate] {
RTC_DCHECK_RUN_ON(network_thread());
std::vector<cricket::Candidate> candidates = {candidate};
RTCError error =
transport_controller_->AddRemoteCandidates(mid, candidates);
if (error.ok()) {
signaling_thread()->PostTask(SafeTask(
signaling_thread_safety_.flag(),
[this, candidate = std::move(candidate)] {
ReportRemoteIceCandidateAdded(candidate);
// Candidates successfully submitted for checking.
if (ice_connection_state() ==
PeerConnectionInterface::kIceConnectionNew ||
ice_connection_state() ==
PeerConnectionInterface::kIceConnectionDisconnected) {
// If state is New, then the session has just gotten its first
// remote ICE candidates, so go to Checking. If state is
// Disconnected, the session is re-using old candidates or
// receiving additional ones, so go to Checking. If state is
// Connected, stay Connected.
// TODO(bemasc): If state is Connected, and the new candidates
// are for a newly added transport, then the state actually
// _should_ move to checking. Add a way to distinguish that
// case.
SetIceConnectionState(
PeerConnectionInterface::kIceConnectionChecking);
}
// TODO(bemasc): If state is Completed, go back to Connected.
}));
} else {
RTC_LOG(LS_WARNING) << error.message();
}
}));
}
void PeerConnection::ReportUsagePattern() const {
usage_pattern_.ReportUsagePattern(observer_);
}
void PeerConnection::ReportRemoteIceCandidateAdded(
const cricket::Candidate& candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
if (candidate.address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED);
}
if (candidate.address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED);
}
if (candidate.address().family() == AF_INET6) {
NoteUsageEvent(UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED);
}
}
bool PeerConnection::SrtpRequired() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return dtls_enabled_;
}
void PeerConnection::OnTransportControllerGatheringState(
cricket::IceGatheringState state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (state == cricket::kIceGatheringGathering) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
} else if (state == cricket::kIceGatheringComplete) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
} else if (state == cricket::kIceGatheringNew) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringNew);
} else {
RTC_LOG(LS_ERROR) << "Unknown state received: " << state;
RTC_DCHECK_NOTREACHED();
}
}
// Runs on network_thread().
void PeerConnection::ReportTransportStats(
std::vector<RtpTransceiverProxyRefPtr> transceivers) {
TRACE_EVENT0("webrtc", "PeerConnection::ReportTransportStats");
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
std::map<std::string, std::set<cricket::MediaType>>
media_types_by_transport_name;
for (const auto& transceiver : transceivers) {
if (transceiver->internal()->channel()) {
std::string transport_name(
transceiver->internal()->channel()->transport_name());
media_types_by_transport_name[transport_name].insert(
transceiver->media_type());
}
}
if (sctp_mid_n_) {
cricket::DtlsTransportInternal* dtls_transport =
transport_controller_->GetDtlsTransport(*sctp_mid_n_);
if (dtls_transport) {
media_types_by_transport_name[dtls_transport->transport_name()].insert(
cricket::MEDIA_TYPE_DATA);
}
}
for (const auto& entry : media_types_by_transport_name) {
const std::string& transport_name = entry.first;
const std::set<cricket::MediaType> media_types = entry.second;
cricket::TransportStats stats;
if (transport_controller_->GetStats(transport_name, &stats)) {
ReportBestConnectionState(stats);
ReportNegotiatedCiphers(dtls_enabled_, stats, media_types);
}
}
}
// Walk through the ConnectionInfos to gather best connection usage
// for IPv4 and IPv6.
// static (no member state required)
void PeerConnection::ReportBestConnectionState(
const cricket::TransportStats& stats) {
for (const cricket::TransportChannelStats& channel_stats :
stats.channel_stats) {
for (const cricket::ConnectionInfo& connection_info :
channel_stats.ice_transport_stats.connection_infos) {
if (!connection_info.best_connection) {
continue;
}
const cricket::Candidate& local = connection_info.local_candidate;
const cricket::Candidate& remote = connection_info.remote_candidate;
// Increment the counter for IceCandidatePairType.
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
(local.is_relay() &&
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else {
RTC_CHECK_NOTREACHED();
}
// Increment the counter for IP type.
if (local.address().family() == AF_INET) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
kBestConnections_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
} else if (local.address().family() == AF_INET6) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
kBestConnections_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
RTC_CHECK(!local.address().hostname().empty() &&
local.address().IsUnresolvedIP());
}
return;
}
}
}
// static
void PeerConnection::ReportNegotiatedCiphers(
bool dtls_enabled,
const cricket::TransportStats& stats,
const std::set<cricket::MediaType>& media_types) {
if (!dtls_enabled || stats.channel_stats.empty()) {
return;
}
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
if (srtp_crypto_suite == rtc::kSrtpInvalidCryptoSuite &&
ssl_cipher_suite == rtc::kTlsNullWithNullNull) {
return;
}
if (ssl_cipher_suite != rtc::kTlsNullWithNullNull) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
rtc::kSslCipherSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
rtc::kSslCipherSuiteMaxValue);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
rtc::kSslCipherSuiteMaxValue);
break;
default:
RTC_DCHECK_NOTREACHED();
continue;
}
}
}
uint16_t ssl_peer_signature_algorithm =
stats.channel_stats[0].ssl_peer_signature_algorithm;
if (ssl_peer_signature_algorithm != rtc::kSslSignatureAlgorithmUnknown) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Audio",
ssl_peer_signature_algorithm,
rtc::kSslSignatureAlgorithmMaxValue);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Video",
ssl_peer_signature_algorithm,
rtc::kSslSignatureAlgorithmMaxValue);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslPeerSignatureAlgorithm.Data",
ssl_peer_signature_algorithm,
rtc::kSslSignatureAlgorithmMaxValue);
break;
default:
RTC_DCHECK_NOTREACHED();
continue;
}
}
}
}
bool PeerConnection::OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
DataChannelTransportInterface* data_channel_transport) {
RTC_DCHECK_RUN_ON(network_thread());
bool ret = true;
if (ConfiguredForMedia()) {
for (const auto& transceiver :
rtp_manager()->transceivers()->UnsafeList()) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && channel->mid() == mid) {
ret = channel->SetRtpTransport(rtp_transport);
}
}
}
if (mid == sctp_mid_n_) {
data_channel_controller_.OnTransportChanged(data_channel_transport);
if (dtls_transport) {
signaling_thread()->PostTask(SafeTask(
signaling_thread_safety_.flag(),
[this,
name = std::string(dtls_transport->internal()->transport_name())] {
RTC_DCHECK_RUN_ON(signaling_thread());
SetSctpTransportName(std::move(name));
}));
}
}
return ret;
}
PeerConnectionObserver* PeerConnection::Observer() const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(observer_);
return observer_;
}
void PeerConnection::StartSctpTransport(int local_port,
int remote_port,
int max_message_size) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!sctp_mid_s_)
return;
network_thread()->PostTask(SafeTask(
network_thread_safety_,
[this, mid = *sctp_mid_s_, local_port, remote_port, max_message_size] {
rtc::scoped_refptr<SctpTransport> sctp_transport =
transport_controller_n()->GetSctpTransport(mid);
if (sctp_transport)
sctp_transport->Start(local_port, remote_port, max_message_size);
}));
}
CryptoOptions PeerConnection::GetCryptoOptions() {
RTC_DCHECK_RUN_ON(signaling_thread());
// TODO(bugs.webrtc.org/9891) - Remove PeerConnectionFactory::CryptoOptions
// after it has been removed.
return configuration_.crypto_options.has_value()
? *configuration_.crypto_options
: options_.crypto_options;
}
void PeerConnection::ClearStatsCache() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (legacy_stats_) {
legacy_stats_->InvalidateCache();
}
if (stats_collector_) {
stats_collector_->ClearCachedStatsReport();
}
}
bool PeerConnection::ShouldFireNegotiationNeededEvent(uint32_t event_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
return sdp_handler_->ShouldFireNegotiationNeededEvent(event_id);
}
void PeerConnection::RequestUsagePatternReportForTesting() {
RTC_DCHECK_RUN_ON(signaling_thread());
message_handler_.RequestUsagePatternReport(
[this]() {
RTC_DCHECK_RUN_ON(signaling_thread());
ReportUsagePattern();
},
/* delay_ms= */ 0);
}
int PeerConnection::FeedbackAccordingToRfc8888CountForTesting() const {
return worker_thread()->BlockingCall([this]() {
RTC_DCHECK_RUN_ON(worker_thread());
return call_->FeedbackAccordingToRfc8888Count();
});
}
int PeerConnection::FeedbackAccordingToTransportCcCountForTesting() const {
return worker_thread()->BlockingCall([this]() {
RTC_DCHECK_RUN_ON(worker_thread());
return call_->FeedbackAccordingToTransportCcCount();
});
}
std::function<void(const rtc::CopyOnWriteBuffer& packet,
int64_t packet_time_us)>
PeerConnection::InitializeRtcpCallback() {
RTC_DCHECK_RUN_ON(network_thread());
return [this](const rtc::CopyOnWriteBuffer& packet,
int64_t /*packet_time_us*/) {
worker_thread()->PostTask(SafeTask(worker_thread_safety_, [this, packet]() {
call_ptr_->Receiver()->DeliverRtcpPacket(packet);
}));
};
}
std::function<void(const RtpPacketReceived& parsed_packet)>
PeerConnection::InitializeUnDemuxablePacketHandler() {
RTC_DCHECK_RUN_ON(network_thread());
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(
const RTCConfiguration& configuration) {
// Enable DTLS-in-STUN only if no certificates were passed those
// may be RSA certificates and this feature only works with small
// ECDSA certificates. Determining the type of the key is
// not trivially possible at this point.
return dtls_enabled_ && configuration.certificates.empty() &&
env_.field_trials().IsEnabled("WebRTC-IceHandshakeDtls");
}
} // namespace webrtc
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