diff --git a/p2p/base/p2p_transport_channel.cc b/p2p/base/p2p_transport_channel.cc
index eff79ab9be..836721c151 100644
--- a/p2p/base/p2p_transport_channel.cc
+++ b/p2p/base/p2p_transport_channel.cc
@@ -43,6 +43,7 @@
 #include "rtc_base/task_utils/to_queued_task.h"
 #include "rtc_base/third_party/sigslot/sigslot.h"
 #include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 
@@ -193,6 +194,7 @@ P2PTransportChannel::P2PTransportChannel(
               true /* presume_writable_when_fully_relayed */,
               REGATHER_ON_FAILED_NETWORKS_INTERVAL,
               RECEIVING_SWITCHING_DELAY) {
+  TRACE_EVENT0("webrtc", "P2PTransportChannel::P2PTransportChannel");
   RTC_DCHECK(allocator_ != nullptr);
   weak_ping_interval_ = GetWeakPingIntervalInFieldTrial();
   // Validate IceConfig even for mostly built-in constant default values in case
@@ -247,6 +249,7 @@ P2PTransportChannel::P2PTransportChannel(
           ice_controller_factory) {}
 
 P2PTransportChannel::~P2PTransportChannel() {
+  TRACE_EVENT0("webrtc", "P2PTransportChannel::~P2PTransportChannel");
   RTC_DCHECK_RUN_ON(network_thread_);
   std::vector<Connection*> copy(connections().begin(), connections().end());
   for (Connection* con : copy) {
diff --git a/p2p/base/port.cc b/p2p/base/port.cc
index d24d40f957..0a123b4d93 100644
--- a/p2p/base/port.cc
+++ b/p2p/base/port.cc
@@ -33,6 +33,7 @@
 #include "rtc_base/string_utils.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/third_party/base64/base64.h"
+#include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace {
@@ -836,6 +837,7 @@ void Port::Prune() {
 
 // Call to stop any currently pending operations from running.
 void Port::CancelPendingTasks() {
+  TRACE_EVENT0("webrtc", "Port::CancelPendingTasks");
   RTC_DCHECK_RUN_ON(thread_);
   thread_->Clear(this);
 }
diff --git a/p2p/client/basic_port_allocator.cc b/p2p/client/basic_port_allocator.cc
index 7e1f970fad..a190fb75da 100644
--- a/p2p/client/basic_port_allocator.cc
+++ b/p2p/client/basic_port_allocator.cc
@@ -27,6 +27,7 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/helpers.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 
@@ -268,12 +269,16 @@ BasicPortAllocatorSession::BasicPortAllocatorSession(
       network_manager_started_(false),
       allocation_sequences_created_(false),
       turn_port_prune_policy_(allocator->turn_port_prune_policy()) {
+  TRACE_EVENT0("webrtc",
+               "BasicPortAllocatorSession::BasicPortAllocatorSession");
   allocator_->network_manager()->SignalNetworksChanged.connect(
       this, &BasicPortAllocatorSession::OnNetworksChanged);
   allocator_->network_manager()->StartUpdating();
 }
 
 BasicPortAllocatorSession::~BasicPortAllocatorSession() {
+  TRACE_EVENT0("webrtc",
+               "BasicPortAllocatorSession::~BasicPortAllocatorSession");
   RTC_DCHECK_RUN_ON(network_thread_);
   allocator_->network_manager()->StopUpdating();
   if (network_thread_ != NULL)
@@ -1248,6 +1253,7 @@ void AllocationSequence::Init() {
 }
 
 void AllocationSequence::Clear() {
+  TRACE_EVENT0("webrtc", "AllocationSequence::Clear");
   udp_port_ = NULL;
   relay_ports_.clear();
 }
diff --git a/pc/channel.cc b/pc/channel.cc
index a8ecd4b020..8630703be1 100644
--- a/pc/channel.cc
+++ b/pc/channel.cc
@@ -220,6 +220,7 @@ void BaseChannel::Deinit() {
 }
 
 bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
+  TRACE_EVENT0("webrtc", "BaseChannel::SetRtpTransport");
   RTC_DCHECK_RUN_ON(network_thread());
   if (rtp_transport == rtp_transport_) {
     return true;
@@ -524,6 +525,7 @@ void BaseChannel::DisableMedia_w() {
 }
 
 void BaseChannel::UpdateWritableState_n() {
+  TRACE_EVENT0("webrtc", "BaseChannel::UpdateWritableState_n");
   if (rtp_transport_->IsWritable(/*rtcp=*/true) &&
       rtp_transport_->IsWritable(/*rtcp=*/false)) {
     ChannelWritable_n();
@@ -533,6 +535,7 @@ void BaseChannel::UpdateWritableState_n() {
 }
 
 void BaseChannel::ChannelWritable_n() {
+  TRACE_EVENT0("webrtc", "BaseChannel::ChannelWritable_n");
   if (writable_) {
     return;
   }
@@ -552,6 +555,7 @@ void BaseChannel::ChannelWritable_n() {
 }
 
 void BaseChannel::ChannelNotWritable_n() {
+  TRACE_EVENT0("webrtc", "BaseChannel::ChannelNotWritable_n");
   if (!writable_) {
     return;
   }
diff --git a/pc/jsep_transport.cc b/pc/jsep_transport.cc
index dc4649bf11..e72088885f 100644
--- a/pc/jsep_transport.cc
+++ b/pc/jsep_transport.cc
@@ -26,6 +26,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/ref_counted_object.h"
 #include "rtc_base/strings/string_builder.h"
+#include "rtc_base/trace_event.h"
 
 using webrtc::SdpType;
 
@@ -104,6 +105,7 @@ JsepTransport::JsepTransport(
                           ? rtc::make_ref_counted<webrtc::SctpTransport>(
                                 std::move(sctp_transport))
                           : nullptr) {
+  TRACE_EVENT0("webrtc", "JsepTransport::JsepTransport");
   RTC_DCHECK(ice_transport_);
   RTC_DCHECK(rtp_dtls_transport_);
   // |rtcp_ice_transport_| must be present iff |rtcp_dtls_transport_| is
@@ -129,6 +131,7 @@ JsepTransport::JsepTransport(
 }
 
 JsepTransport::~JsepTransport() {
+  TRACE_EVENT0("webrtc", "JsepTransport::~JsepTransport");
   if (sctp_transport_) {
     sctp_transport_->Clear();
   }
@@ -147,7 +150,7 @@ webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription(
     const JsepTransportDescription& jsep_description,
     SdpType type) {
   webrtc::RTCError error;
-
+  TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription");
   RTC_DCHECK_RUN_ON(network_thread_);
 
   IceParameters ice_parameters =
@@ -233,6 +236,7 @@ webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription(
 webrtc::RTCError JsepTransport::SetRemoteJsepTransportDescription(
     const JsepTransportDescription& jsep_description,
     webrtc::SdpType type) {
+  TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription");
   webrtc::RTCError error;
 
   RTC_DCHECK_RUN_ON(network_thread_);
@@ -344,6 +348,7 @@ absl::optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const {
 }
 
 bool JsepTransport::GetStats(TransportStats* stats) {
+  TRACE_EVENT0("webrtc", "JsepTransport::GetStats");
   RTC_DCHECK_RUN_ON(network_thread_);
   stats->transport_name = mid();
   stats->channel_stats.clear();
@@ -362,6 +367,7 @@ bool JsepTransport::GetStats(TransportStats* stats) {
 webrtc::RTCError JsepTransport::VerifyCertificateFingerprint(
     const rtc::RTCCertificate* certificate,
     const rtc::SSLFingerprint* fingerprint) const {
+  TRACE_EVENT0("webrtc", "JsepTransport::VerifyCertificateFingerprint");
   RTC_DCHECK_RUN_ON(network_thread_);
   if (!fingerprint) {
     return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
@@ -400,6 +406,7 @@ void JsepTransport::SetActiveResetSrtpParams(bool active_reset_srtp_params) {
 void JsepTransport::SetRemoteIceParameters(
     const IceParameters& ice_parameters,
     IceTransportInternal* ice_transport) {
+  TRACE_EVENT0("webrtc", "JsepTransport::SetRemoteIceParameters");
   RTC_DCHECK_RUN_ON(network_thread_);
   RTC_DCHECK(ice_transport);
   RTC_DCHECK(remote_description_);
diff --git a/pc/jsep_transport_controller.cc b/pc/jsep_transport_controller.cc
index 78aa032c52..a12438956b 100644
--- a/pc/jsep_transport_controller.cc
+++ b/pc/jsep_transport_controller.cc
@@ -32,6 +32,7 @@
 #include "rtc_base/net_helper.h"
 #include "rtc_base/socket_address.h"
 #include "rtc_base/thread.h"
+#include "rtc_base/trace_event.h"
 
 using webrtc::SdpType;
 
@@ -77,6 +78,7 @@ JsepTransportController::~JsepTransportController() {
 RTCError JsepTransportController::SetLocalDescription(
     SdpType type,
     const cricket::SessionDescription* description) {
+  TRACE_EVENT0("webrtc", "JsepTransportController::SetLocalDescription");
   if (!network_thread_->IsCurrent()) {
     return network_thread_->Invoke<RTCError>(
         RTC_FROM_HERE, [=] { return SetLocalDescription(type, description); });
@@ -97,6 +99,7 @@ RTCError JsepTransportController::SetLocalDescription(
 RTCError JsepTransportController::SetRemoteDescription(
     SdpType type,
     const cricket::SessionDescription* description) {
+  TRACE_EVENT0("webrtc", "JsepTransportController::SetRemoteDescription");
   if (!network_thread_->IsCurrent()) {
     return network_thread_->Invoke<RTCError>(
         RTC_FROM_HERE, [=] { return SetRemoteDescription(type, description); });
@@ -539,6 +542,7 @@ RTCError JsepTransportController::ApplyDescription_n(
     bool local,
     SdpType type,
     const cricket::SessionDescription* description) {
+  TRACE_EVENT0("webrtc", "JsepTransportController::ApplyDescription_n");
   RTC_DCHECK(description);
 
   if (local) {
@@ -866,6 +870,7 @@ void JsepTransportController::HandleRejectedContent(
 bool JsepTransportController::HandleBundledContent(
     const cricket::ContentInfo& content_info,
     const cricket::ContentGroup& bundle_group) {
+  TRACE_EVENT0("webrtc", "JsepTransportController::HandleBundledContent");
   RTC_DCHECK(bundle_group.FirstContentName());
   auto jsep_transport =
       GetJsepTransportByName(*bundle_group.FirstContentName());
@@ -887,6 +892,7 @@ bool JsepTransportController::HandleBundledContent(
 bool JsepTransportController::SetTransportForMid(
     const std::string& mid,
     cricket::JsepTransport* jsep_transport) {
+  TRACE_EVENT0("webrtc", "JsepTransportController::SetTransportForMid");
   RTC_DCHECK_RUN_ON(network_thread_);
   RTC_DCHECK(jsep_transport);
 
@@ -924,6 +930,8 @@ JsepTransportController::CreateJsepTransportDescription(
     const cricket::TransportInfo& transport_info,
     const std::vector<int>& encrypted_extension_ids,
     int rtp_abs_sendtime_extn_id) {
+  TRACE_EVENT0("webrtc",
+               "JsepTransportController::CreateJsepTransportDescription");
   const cricket::MediaContentDescription* content_desc =
       content_info.media_description();
   RTC_DCHECK(content_desc);
@@ -1127,6 +1135,7 @@ RTCError JsepTransportController::MaybeCreateJsepTransport(
 
 void JsepTransportController::MaybeDestroyJsepTransport(
     const std::string& mid) {
+  TRACE_EVENT0("webrtc", "JsepTransportController::MaybeDestroyJsepTransport");
   auto jsep_transport = GetJsepTransportByName(mid);
   if (!jsep_transport) {
     return;
@@ -1280,6 +1289,7 @@ void JsepTransportController::OnTransportStateChanged_n(
 }
 
 void JsepTransportController::UpdateAggregateStates_n() {
+  TRACE_EVENT0("webrtc", "JsepTransportController::UpdateAggregateStates_n");
   auto dtls_transports = GetDtlsTransports();
   cricket::IceConnectionState new_connection_state =
       cricket::kIceConnectionConnecting;
diff --git a/pc/peer_connection.cc b/pc/peer_connection.cc
index 2ef00deee7..54b49340ba 100644
--- a/pc/peer_connection.cc
+++ b/pc/peer_connection.cc
@@ -2188,6 +2188,7 @@ cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
 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 {};
@@ -2636,6 +2637,7 @@ void PeerConnection::OnTransportControllerGatheringState(
 
 // Runs on network_thread().
 void PeerConnection::ReportTransportStats() {
+  TRACE_EVENT0("webrtc", "PeerConnection::ReportTransportStats");
   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
   std::map<std::string, std::set<cricket::MediaType>>
       media_types_by_transport_name;
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index d7b7ba7802..4b38abc073 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -1273,6 +1273,8 @@ void RTCStatsCollector::ProducePartialResultsOnSignalingThreadImpl(
 void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
     int64_t timestamp_us,
     absl::optional<std::string> sctp_transport_name) {
+  TRACE_EVENT0("webrtc",
+               "RTCStatsCollector::ProducePartialResultsOnNetworkThread");
   RTC_DCHECK_RUN_ON(network_thread_);
   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 761ae7cfae..8aa194781e 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -1251,6 +1251,7 @@ RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
     std::unique_ptr<SessionDescriptionInterface> desc,
     const std::map<std::string, const cricket::ContentGroup*>&
         bundle_groups_by_mid) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyLocalDescription");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(desc);
 
@@ -1540,6 +1541,7 @@ RTCError SdpOfferAnswerHandler::ApplyRemoteDescription(
     std::unique_ptr<SessionDescriptionInterface> desc,
     const std::map<std::string, const cricket::ContentGroup*>&
         bundle_groups_by_mid) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyRemoteDescription");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(desc);
 
@@ -2025,6 +2027,7 @@ void SdpOfferAnswerHandler::DoCreateOffer(
 void SdpOfferAnswerHandler::CreateAnswer(
     CreateSessionDescriptionObserver* observer,
     const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateAnswer");
   RTC_DCHECK_RUN_ON(signaling_thread());
   // Chain this operation. If asynchronous operations are pending on the chain,
   // this operation will be queued to be invoked, otherwise the contents of the
@@ -2325,6 +2328,7 @@ AddIceCandidateResult SdpOfferAnswerHandler::AddIceCandidateInternal(
 void SdpOfferAnswerHandler::AddIceCandidate(
     std::unique_ptr<IceCandidateInterface> candidate,
     std::function<void(RTCError)> callback) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AddIceCandidate");
   RTC_DCHECK_RUN_ON(signaling_thread());
   // Chain this operation. If asynchronous operations are pending on the chain,
   // this operation will be queued to be invoked, otherwise the contents of the
@@ -2456,6 +2460,7 @@ PeerConnectionInterface::SignalingState SdpOfferAnswerHandler::signaling_state()
 
 void SdpOfferAnswerHandler::ChangeSignalingState(
     PeerConnectionInterface::SignalingState signaling_state) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ChangeSignalingState");
   RTC_DCHECK_RUN_ON(signaling_thread());
   if (signaling_state_ == signaling_state) {
     return;
@@ -2660,6 +2665,7 @@ void SdpOfferAnswerHandler::OnVideoTrackRemoved(VideoTrackInterface* track,
 }
 
 RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::Rollback");
   auto state = signaling_state();
   if (state != PeerConnectionInterface::kHaveLocalOffer &&
       state != PeerConnectionInterface::kHaveRemoteOffer) {
@@ -3119,6 +3125,8 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
     const SessionDescriptionInterface* old_remote_description,
     const std::map<std::string, const cricket::ContentGroup*>&
         bundle_groups_by_mid) {
+  TRACE_EVENT0("webrtc",
+               "SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(IsUnifiedPlan());
 
@@ -3205,6 +3213,7 @@ SdpOfferAnswerHandler::AssociateTransceiver(
     const ContentInfo& content,
     const ContentInfo* old_local_content,
     const ContentInfo* old_remote_content) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AssociateTransceiver");
   RTC_DCHECK(IsUnifiedPlan());
 #if RTC_DCHECK_IS_ON
   // If this is an offer then the m= section might be recycled. If the m=
@@ -3338,6 +3347,7 @@ RTCError SdpOfferAnswerHandler::UpdateTransceiverChannel(
         transceiver,
     const cricket::ContentInfo& content,
     const cricket::ContentGroup* bundle_group) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateTransceiverChannel");
   RTC_DCHECK(IsUnifiedPlan());
   RTC_DCHECK(transceiver);
   cricket::ChannelInterface* channel = transceiver->internal()->channel();
@@ -4022,6 +4032,7 @@ void SdpOfferAnswerHandler::RemoveSenders(cricket::MediaType media_type) {
 void SdpOfferAnswerHandler::UpdateLocalSenders(
     const std::vector<cricket::StreamParams>& streams,
     cricket::MediaType media_type) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateLocalSenders");
   RTC_DCHECK_RUN_ON(signaling_thread());
   std::vector<RtpSenderInfo>* current_senders =
       rtp_manager()->GetLocalSenderInfos(media_type);
@@ -4064,6 +4075,7 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
     bool default_sender_needed,
     cricket::MediaType media_type,
     StreamCollection* new_streams) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateRemoteSendersList");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(!IsUnifiedPlan());
 
@@ -4163,6 +4175,7 @@ void SdpOfferAnswerHandler::UpdateRemoteSendersList(
 }
 
 void SdpOfferAnswerHandler::EnableSending() {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::EnableSending");
   RTC_DCHECK_RUN_ON(signaling_thread());
   for (const auto& transceiver : transceivers()->ListInternal()) {
     cricket::ChannelInterface* channel = transceiver->channel();
@@ -4177,6 +4190,7 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
     cricket::ContentSource source,
     const std::map<std::string, const cricket::ContentGroup*>&
         bundle_groups_by_mid) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownMediaDescription");
   const SessionDescriptionInterface* sdesc =
       (source == cricket::CS_LOCAL ? local_description()
                                    : remote_description());
@@ -4270,6 +4284,7 @@ RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
 RTCError SdpOfferAnswerHandler::PushdownTransportDescription(
     cricket::ContentSource source,
     SdpType type) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownTransportDescription");
   RTC_DCHECK_RUN_ON(signaling_thread());
 
   if (source == cricket::CS_LOCAL) {
@@ -4286,6 +4301,7 @@ RTCError SdpOfferAnswerHandler::PushdownTransportDescription(
 }
 
 void SdpOfferAnswerHandler::RemoveStoppedTransceivers() {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::RemoveStoppedTransceivers");
   RTC_DCHECK_RUN_ON(signaling_thread());
   // 3.2.10.1: For each transceiver in the connection's set of transceivers
   //           run the following steps:
@@ -4505,6 +4521,7 @@ RTCErrorOr<const cricket::ContentInfo*> SdpOfferAnswerHandler::FindContentInfo(
 }
 
 RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateChannels");
   // Creating the media channels. Transports should already have been created
   // at this point.
   RTC_DCHECK_RUN_ON(signaling_thread());
@@ -4545,6 +4562,7 @@ RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
 // TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
 cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
     const std::string& mid) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateVoiceChannel");
   RTC_DCHECK_RUN_ON(signaling_thread());
   if (!channel_manager()->media_engine())
     return nullptr;
@@ -4563,6 +4581,7 @@ cricket::VoiceChannel* SdpOfferAnswerHandler::CreateVoiceChannel(
 // TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
 cricket::VideoChannel* SdpOfferAnswerHandler::CreateVideoChannel(
     const std::string& mid) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateVideoChannel");
   RTC_DCHECK_RUN_ON(signaling_thread());
   if (!channel_manager()->media_engine())
     return nullptr;
@@ -4600,6 +4619,7 @@ bool SdpOfferAnswerHandler::CreateDataChannel(const std::string& mid) {
 void SdpOfferAnswerHandler::DestroyTransceiverChannel(
     rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
         transceiver) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DestroyTransceiverChannel");
   RTC_DCHECK(transceiver);
   RTC_LOG_THREAD_BLOCK_COUNT();
 
@@ -4648,6 +4668,7 @@ void SdpOfferAnswerHandler::DestroyDataChannelTransport() {
 
 void SdpOfferAnswerHandler::DestroyChannelInterface(
     cricket::ChannelInterface* channel) {
+  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DestroyChannelInterface");
   RTC_DCHECK_RUN_ON(signaling_thread());
   RTC_DCHECK(channel_manager()->media_engine());
   RTC_DCHECK(channel);
@@ -4802,6 +4823,8 @@ bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
     cricket::ContentSource source,
     const std::map<std::string, const cricket::ContentGroup*>&
         bundle_groups_by_mid) {
+  TRACE_EVENT0("webrtc",
+               "SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState");
   RTC_DCHECK_RUN_ON(signaling_thread());
   // We may need to delete any created default streams and disable creation of
   // new ones on the basis of payload type. This is needed to avoid SSRC
diff --git a/pc/srtp_transport.cc b/pc/srtp_transport.cc
index ee073497e7..c90b3fa227 100644
--- a/pc/srtp_transport.cc
+++ b/pc/srtp_transport.cc
@@ -201,12 +201,12 @@ bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
 
 void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
                                         int64_t packet_time_us) {
+  TRACE_EVENT0("webrtc", "SrtpTransport::OnRtpPacketReceived");
   if (!IsSrtpActive()) {
     RTC_LOG(LS_WARNING)
         << "Inactive SRTP transport received an RTP packet. Drop it.";
     return;
   }
-  TRACE_EVENT0("webrtc", "SRTP Decode");
   char* data = packet.MutableData<char>();
   int len = rtc::checked_cast<int>(packet.size());
   if (!UnprotectRtp(data, len, &len)) {
@@ -233,12 +233,12 @@ void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet,
 
 void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet,
                                          int64_t packet_time_us) {
+  TRACE_EVENT0("webrtc", "SrtpTransport::OnRtcpPacketReceived");
   if (!IsSrtpActive()) {
     RTC_LOG(LS_WARNING)
         << "Inactive SRTP transport received an RTCP packet. Drop it.";
     return;
   }
-  TRACE_EVENT0("webrtc", "SRTP Decode");
   char* data = packet.MutableData<char>();
   int len = rtc::checked_cast<int>(packet.size());
   if (!UnprotectRtcp(data, len, &len)) {
diff --git a/pc/stats_collector.cc b/pc/stats_collector.cc
index 6d4c224cb6..1728cb4d5d 100644
--- a/pc/stats_collector.cc
+++ b/pc/stats_collector.cc
@@ -50,6 +50,7 @@
 #include "rtc_base/string_encode.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
@@ -849,6 +850,7 @@ StatsReport* StatsCollector::AddCandidateReport(
 }
 
 std::map<std::string, std::string> StatsCollector::ExtractSessionInfo() {
+  TRACE_EVENT0("webrtc", "StatsCollector::ExtractSessionInfo");
   RTC_DCHECK_RUN_ON(pc_->signaling_thread());
 
   SessionStats stats;
@@ -870,6 +872,7 @@ StatsCollector::SessionStats StatsCollector::ExtractSessionInfo_n(
         RtpTransceiverProxyWithInternal<RtpTransceiver>>>& transceivers,
     absl::optional<std::string> sctp_transport_name,
     absl::optional<std::string> sctp_mid) {
+  TRACE_EVENT0("webrtc", "StatsCollector::ExtractSessionInfo_n");
   RTC_DCHECK_RUN_ON(pc_->network_thread());
   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
   SessionStats stats;