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webrtc / src / 34cd4858e3997f7e93834c4580a5e820020a3ba1 / . / pc / peer_connection.cc
blob: 14c868325607650bd3326de40bf0add3d5ef35fa [file] [log] [blame]
/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/peer_connection.h"
#include <algorithm>
#include <limits>
#include <queue>
#include <set>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "api/jsep_ice_candidate.h"
#include "api/jsep_session_description.h"
#include "api/media_stream_proxy.h"
#include "api/media_stream_track_proxy.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log_output_file.h"
#include "api/rtp_parameters.h"
#include "api/uma_metrics.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "call/call.h"
#include "logging/rtc_event_log/ice_logger.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/rid_description.h"
#include "media/sctp/sctp_transport.h"
#include "pc/audio_rtp_receiver.h"
#include "pc/audio_track.h"
#include "pc/channel.h"
#include "pc/channel_manager.h"
#include "pc/dtmf_sender.h"
#include "pc/media_stream.h"
#include "pc/media_stream_observer.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_media_utils.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_sender.h"
#include "pc/sctp_transport.h"
#include "pc/sctp_utils.h"
#include "pc/sdp_utils.h"
#include "pc/stream_collection.h"
#include "pc/video_rtp_receiver.h"
#include "pc/video_track.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
using cricket::ContentInfo;
using cricket::ContentInfos;
using cricket::MediaContentDescription;
using cricket::MediaProtocolType;
using cricket::RidDescription;
using cricket::RidDirection;
using cricket::SessionDescription;
using cricket::SimulcastDescription;
using cricket::SimulcastLayer;
using cricket::SimulcastLayerList;
using cricket::StreamParams;
using cricket::TransportInfo;
using cricket::LOCAL_PORT_TYPE;
using cricket::PRFLX_PORT_TYPE;
using cricket::RELAY_PORT_TYPE;
using cricket::STUN_PORT_TYPE;
namespace webrtc {
// Error messages
const char kBundleWithoutRtcpMux[] =
"rtcp-mux must be enabled when BUNDLE "
"is enabled.";
const char kInvalidCandidates[] = "Description contains invalid candidates.";
const char kInvalidSdp[] = "Invalid session description.";
const char kMlineMismatchInAnswer[] =
"The order of m-lines in answer doesn't match order in offer. Rejecting "
"answer.";
const char kMlineMismatchInSubsequentOffer[] =
"The order of m-lines in subsequent offer doesn't match order from "
"previous offer/answer.";
const char kSdpWithoutDtlsFingerprint[] =
"Called with SDP without DTLS fingerprint.";
const char kSdpWithoutSdesCrypto[] = "Called with SDP without SDES crypto.";
const char kSdpWithoutIceUfragPwd[] =
"Called with SDP without ice-ufrag and ice-pwd.";
const char kSessionError[] = "Session error code: ";
const char kSessionErrorDesc[] = "Session error description: ";
const char kDtlsSrtpSetupFailureRtp[] =
"Couldn't set up DTLS-SRTP on RTP channel.";
const char kDtlsSrtpSetupFailureRtcp[] =
"Couldn't set up DTLS-SRTP on RTCP channel.";
namespace {
// UMA metric names.
const char kSimulcastVersionApplyLocalDescription[] =
"WebRTC.PeerConnection.Simulcast.ApplyLocalDescription";
const char kSimulcastVersionApplyRemoteDescription[] =
"WebRTC.PeerConnection.Simulcast.ApplyRemoteDescription";
const char kSimulcastNumberOfEncodings[] =
"WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings";
const char kSimulcastDisabled[] = "WebRTC.PeerConnection.Simulcast.Disabled";
static const char kDefaultStreamId[] = "default";
static const char kDefaultAudioSenderId[] = "defaulta0";
static const char kDefaultVideoSenderId[] = "defaultv0";
// The length of RTCP CNAMEs.
static const int kRtcpCnameLength = 16;
enum {
MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
MSG_SET_SESSIONDESCRIPTION_FAILED,
MSG_CREATE_SESSIONDESCRIPTION_FAILED,
MSG_GETSTATS,
MSG_FREE_DATACHANNELS,
MSG_REPORT_USAGE_PATTERN,
};
static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
struct SetSessionDescriptionMsg : public rtc::MessageData {
explicit SetSessionDescriptionMsg(
webrtc::SetSessionDescriptionObserver* observer)
: observer(observer) {}
rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
RTCError error;
};
struct CreateSessionDescriptionMsg : public rtc::MessageData {
explicit CreateSessionDescriptionMsg(
webrtc::CreateSessionDescriptionObserver* observer)
: observer(observer) {}
rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
RTCError error;
};
struct GetStatsMsg : public rtc::MessageData {
GetStatsMsg(webrtc::StatsObserver* observer,
webrtc::MediaStreamTrackInterface* track)
: observer(observer), track(track) {}
rtc::scoped_refptr<webrtc::StatsObserver> observer;
rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
};
// Check if we can send |new_stream| on a PeerConnection.
bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
webrtc::MediaStreamInterface* new_stream) {
if (!new_stream || !current_streams) {
return false;
}
if (current_streams->find(new_stream->id()) != nullptr) {
RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
<< " is already added.";
return false;
}
return true;
}
// If the direction is "recvonly" or "inactive", treat the description
// as containing no streams.
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
std::vector<cricket::StreamParams> GetActiveStreams(
const cricket::MediaContentDescription* desc) {
return RtpTransceiverDirectionHasSend(desc->direction())
? desc->streams()
: std::vector<cricket::StreamParams>();
}
bool IsValidOfferToReceiveMedia(int value) {
typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
return (value >= Options::kUndefined) &&
(value <= Options::kMaxOfferToReceiveMedia);
}
// Add options to |[audio/video]_media_description_options| from |senders|.
void AddPlanBRtpSenderOptions(
const std::vector<rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
cricket::MediaDescriptionOptions* audio_media_description_options,
cricket::MediaDescriptionOptions* video_media_description_options,
int num_sim_layers) {
for (const auto& sender : senders) {
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
if (audio_media_description_options) {
audio_media_description_options->AddAudioSender(
sender->id(), sender->internal()->stream_ids());
}
} else {
RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO);
if (video_media_description_options) {
video_media_description_options->AddVideoSender(
sender->id(), sender->internal()->stream_ids(), {},
SimulcastLayerList(), num_sim_layers);
}
}
}
}
// Add options to |session_options| from |rtp_data_channels|.
void AddRtpDataChannelOptions(
const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
rtp_data_channels,
cricket::MediaDescriptionOptions* data_media_description_options) {
if (!data_media_description_options) {
return;
}
// Check for data channels.
for (const auto& kv : rtp_data_channels) {
const DataChannel* channel = kv.second;
if (channel->state() == DataChannel::kConnecting ||
channel->state() == DataChannel::kOpen) {
// Legacy RTP data channels are signaled with the track/stream ID set to
// the data channel's label.
data_media_description_options->AddRtpDataChannel(channel->label(),
channel->label());
}
}
}
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_NOTREACHED();
}
return cricket::CF_NONE;
}
// Helper to set an error and return from a method.
bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
if (error) {
error->set_type(type);
}
return type == webrtc::RTCErrorType::NONE;
}
bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
bool ok = error.ok();
if (error_out) {
*error_out = std::move(error);
}
return ok;
}
std::string GetSignalingStateString(
PeerConnectionInterface::SignalingState state) {
switch (state) {
case PeerConnectionInterface::kStable:
return "kStable";
case PeerConnectionInterface::kHaveLocalOffer:
return "kHaveLocalOffer";
case PeerConnectionInterface::kHaveLocalPrAnswer:
return "kHavePrAnswer";
case PeerConnectionInterface::kHaveRemoteOffer:
return "kHaveRemoteOffer";
case PeerConnectionInterface::kHaveRemotePrAnswer:
return "kHaveRemotePrAnswer";
case PeerConnectionInterface::kClosed:
return "kClosed";
}
RTC_NOTREACHED();
return "";
}
IceCandidatePairType GetIceCandidatePairCounter(
const cricket::Candidate& local,
const cricket::Candidate& remote) {
const auto& l = local.type();
const auto& r = remote.type();
const auto& host = LOCAL_PORT_TYPE;
const auto& srflx = STUN_PORT_TYPE;
const auto& relay = RELAY_PORT_TYPE;
const auto& prflx = PRFLX_PORT_TYPE;
if (l == host && r == host) {
bool local_hostname =
!local.address().hostname().empty() && local.address().IsUnresolvedIP();
bool remote_hostname = !remote.address().hostname().empty() &&
remote.address().IsUnresolvedIP();
bool local_private = IPIsPrivate(local.address().ipaddr());
bool remote_private = IPIsPrivate(remote.address().ipaddr());
if (local_hostname) {
if (remote_hostname) {
return kIceCandidatePairHostNameHostName;
} else if (remote_private) {
return kIceCandidatePairHostNameHostPrivate;
} else {
return kIceCandidatePairHostNameHostPublic;
}
} else if (local_private) {
if (remote_hostname) {
return kIceCandidatePairHostPrivateHostName;
} else if (remote_private) {
return kIceCandidatePairHostPrivateHostPrivate;
} else {
return kIceCandidatePairHostPrivateHostPublic;
}
} else {
if (remote_hostname) {
return kIceCandidatePairHostPublicHostName;
} else if (remote_private) {
return kIceCandidatePairHostPublicHostPrivate;
} else {
return kIceCandidatePairHostPublicHostPublic;
}
}
}
if (l == host && r == srflx)
return kIceCandidatePairHostSrflx;
if (l == host && r == relay)
return kIceCandidatePairHostRelay;
if (l == host && r == prflx)
return kIceCandidatePairHostPrflx;
if (l == srflx && r == host)
return kIceCandidatePairSrflxHost;
if (l == srflx && r == srflx)
return kIceCandidatePairSrflxSrflx;
if (l == srflx && r == relay)
return kIceCandidatePairSrflxRelay;
if (l == srflx && r == prflx)
return kIceCandidatePairSrflxPrflx;
if (l == relay && r == host)
return kIceCandidatePairRelayHost;
if (l == relay && r == srflx)
return kIceCandidatePairRelaySrflx;
if (l == relay && r == relay)
return kIceCandidatePairRelayRelay;
if (l == relay && r == prflx)
return kIceCandidatePairRelayPrflx;
if (l == prflx && r == host)
return kIceCandidatePairPrflxHost;
if (l == prflx && r == srflx)
return kIceCandidatePairPrflxSrflx;
if (l == prflx && r == relay)
return kIceCandidatePairPrflxRelay;
return kIceCandidatePairMax;
}
// Logic to decide if an m= section can be recycled. This means that the new
// m= section is not rejected, but the old local or remote m= section is
// rejected. |old_content_one| and |old_content_two| refer to the m= section
// of the old remote and old local descriptions in no particular order.
// We need to check both the old local and remote because either
// could be the most current from the latest negotation.
bool IsMediaSectionBeingRecycled(SdpType type,
const ContentInfo& content,
const ContentInfo* old_content_one,
const ContentInfo* old_content_two) {
return type == SdpType::kOffer && !content.rejected &&
((old_content_one && old_content_one->rejected) ||
(old_content_two && old_content_two->rejected));
}
// Verify that the order of media sections in |new_desc| matches
// |current_desc|. The number of m= sections in |new_desc| should be no
// less than |current_desc|. In the case of checking an answer's
// |new_desc|, the |current_desc| is the last offer that was set as the
// local or remote. In the case of checking an offer's |new_desc| we
// check against the local and remote descriptions stored from the last
// negotiation, because either of these could be the most up to date for
// possible rejected m sections. These are the |current_desc| and
// |secondary_current_desc|.
bool MediaSectionsInSameOrder(const SessionDescription& current_desc,
const SessionDescription* secondary_current_desc,
const SessionDescription& new_desc,
const SdpType type) {
if (current_desc.contents().size() > new_desc.contents().size()) {
return false;
}
for (size_t i = 0; i < current_desc.contents().size(); ++i) {
const cricket::ContentInfo* secondary_content_info = nullptr;
if (secondary_current_desc &&
i < secondary_current_desc->contents().size()) {
secondary_content_info = &secondary_current_desc->contents()[i];
}
if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i],
&current_desc.contents()[i],
secondary_content_info)) {
// For new offer descriptions, if the media section can be recycled, it's
// valid for the MID and media type to change.
continue;
}
if (new_desc.contents()[i].name != current_desc.contents()[i].name) {
return false;
}
const MediaContentDescription* new_desc_mdesc =
new_desc.contents()[i].media_description();
const MediaContentDescription* current_desc_mdesc =
current_desc.contents()[i].media_description();
if (new_desc_mdesc->type() != current_desc_mdesc->type()) {
return false;
}
}
return true;
}
bool MediaSectionsHaveSameCount(const SessionDescription& desc1,
const SessionDescription& desc2) {
return desc1.contents().size() == desc2.contents().size();
}
void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
cricket::MediaType media_type) {
// Array of structs needed to map {KeyExchangeProtocolType,
// cricket::MediaType} to KeyExchangeProtocolMedia without using std::map in
// order to avoid -Wglobal-constructors and -Wexit-time-destructors.
static constexpr struct {
KeyExchangeProtocolType protocol_type;
cricket::MediaType media_type;
KeyExchangeProtocolMedia protocol_media;
} kEnumCounterKeyProtocolMediaMap[] = {
{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_AUDIO,
kEnumCounterKeyProtocolMediaTypeDtlsAudio},
{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_VIDEO,
kEnumCounterKeyProtocolMediaTypeDtlsVideo},
{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_DATA,
kEnumCounterKeyProtocolMediaTypeDtlsData},
{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_AUDIO,
kEnumCounterKeyProtocolMediaTypeSdesAudio},
{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_VIDEO,
kEnumCounterKeyProtocolMediaTypeSdesVideo},
{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_DATA,
kEnumCounterKeyProtocolMediaTypeSdesData},
};
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
kEnumCounterKeyProtocolMax);
for (const auto& i : kEnumCounterKeyProtocolMediaMap) {
if (i.protocol_type == protocol_type && i.media_type == media_type) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
i.protocol_media,
kEnumCounterKeyProtocolMediaTypeMax);
}
}
}
void NoteAddIceCandidateResult(int result) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result,
kAddIceCandidateMax);
}
// Checks that each non-rejected content has SDES crypto keys or a DTLS
// fingerprint, unless it's in a BUNDLE group, in which case only the
// BUNDLE-tag section (first media section/description in the BUNDLE group)
// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
// by Channel's |srtp_required| check.
RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
for (const cricket::ContentInfo& content_info : desc->contents()) {
if (content_info.rejected) {
continue;
}
// Note what media is used with each crypto protocol, for all sections.
NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
: webrtc::kEnumCounterKeyProtocolSdes,
content_info.media_description()->type());
const std::string& mid = content_info.name;
if (bundle && bundle->HasContentName(mid) &&
mid != *(bundle->FirstContentName())) {
// This isn't the first media section in the BUNDLE group, so it's not
// required to have crypto attributes, since only the crypto attributes
// from the first section actually get used.
continue;
}
// If the content isn't rejected or bundled into another m= section, crypto
// must be present.
const MediaContentDescription* media = content_info.media_description();
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
if (!media || !tinfo) {
// Something is not right.
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
}
if (dtls_enabled) {
if (!tinfo->description.identity_fingerprint) {
RTC_LOG(LS_WARNING)
<< "Session description must have DTLS fingerprint if "
"DTLS enabled.";
return RTCError(RTCErrorType::INVALID_PARAMETER,
kSdpWithoutDtlsFingerprint);
}
} else {
if (media->cryptos().empty()) {
RTC_LOG(LS_WARNING)
<< "Session description must have SDES when DTLS disabled.";
return RTCError(RTCErrorType::INVALID_PARAMETER, kSdpWithoutSdesCrypto);
}
}
}
return RTCError::OK();
}
// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
// media section/description in the BUNDLE group) needs a ufrag and pwd.
bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
for (const cricket::ContentInfo& content_info : desc->contents()) {
if (content_info.rejected) {
continue;
}
const std::string& mid = content_info.name;
if (bundle && bundle->HasContentName(mid) &&
mid != *(bundle->FirstContentName())) {
// This isn't the first media section in the BUNDLE group, so it's not
// required to have ufrag/password, since only the ufrag/password from
// the first section actually get used.
continue;
}
// If the content isn't rejected or bundled into another m= section,
// ice-ufrag and ice-pwd must be present.
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
if (!tinfo) {
// Something is not right.
RTC_LOG(LS_ERROR) << kInvalidSdp;
return false;
}
if (tinfo->description.ice_ufrag.empty() ||
tinfo->description.ice_pwd.empty()) {
RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
return false;
}
}
return true;
}
// Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
const SessionDescriptionInterface* new_desc,
const std::string& content_name) {
if (!old_desc) {
return false;
}
const SessionDescription* new_sd = new_desc->description();
const SessionDescription* old_sd = old_desc->description();
const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
if (!cinfo || cinfo->rejected) {
return false;
}
// If the content isn't rejected, check if ufrag and password has changed.
const cricket::TransportDescription* new_transport_desc =
new_sd->GetTransportDescriptionByName(content_name);
const cricket::TransportDescription* old_transport_desc =
old_sd->GetTransportDescriptionByName(content_name);
if (!new_transport_desc || !old_transport_desc) {
// No transport description exists. This is not an ICE restart.
return false;
}
if (cricket::IceCredentialsChanged(
old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
<< ".";
return true;
}
return false;
}
// Generates a string error message for SetLocalDescription/SetRemoteDescription
// from an RTCError.
std::string GetSetDescriptionErrorMessage(cricket::ContentSource source,
SdpType type,
const RTCError& error) {
rtc::StringBuilder oss;
oss << "Failed to set " << (source == cricket::CS_LOCAL ? "local" : "remote")
<< " " << SdpTypeToString(type) << " sdp: " << error.message();
return oss.Release();
}
std::string GetStreamIdsString(rtc::ArrayView<const std::string> stream_ids) {
std::string output = "streams=[";
const char* separator = "";
for (const auto& stream_id : stream_ids) {
output.append(separator).append(stream_id);
separator = ", ";
}
output.append("]");
return output;
}
absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
int rtc_configuration_parameter) {
if (rtc_configuration_parameter ==
webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
return absl::nullopt;
}
return rtc_configuration_parameter;
}
cricket::DataMessageType ToCricketDataMessageType(DataMessageType type) {
switch (type) {
case DataMessageType::kText:
return cricket::DMT_TEXT;
case DataMessageType::kBinary:
return cricket::DMT_BINARY;
case DataMessageType::kControl:
return cricket::DMT_CONTROL;
default:
return cricket::DMT_NONE;
}
return cricket::DMT_NONE;
}
DataMessageType ToWebrtcDataMessageType(cricket::DataMessageType type) {
switch (type) {
case cricket::DMT_TEXT:
return DataMessageType::kText;
case cricket::DMT_BINARY:
return DataMessageType::kBinary;
case cricket::DMT_CONTROL:
return DataMessageType::kControl;
case cricket::DMT_NONE:
default:
RTC_NOTREACHED();
}
return DataMessageType::kControl;
}
void ReportSimulcastApiVersion(const char* name,
const SessionDescription& session) {
bool has_legacy = false;
bool has_spec_compliant = false;
for (const ContentInfo& content : session.contents()) {
if (!content.description) {
continue;
}
has_spec_compliant |= content.description->HasSimulcast();
for (const StreamParams& sp : content.description->streams()) {
has_legacy |= sp.has_ssrc_group(cricket::kSimSsrcGroupSemantics);
}
}
if (has_legacy) {
RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionLegacy,
kSimulcastApiVersionMax);
}
if (has_spec_compliant) {
RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionSpecCompliant,
kSimulcastApiVersionMax);
}
if (!has_legacy && !has_spec_compliant) {
RTC_HISTOGRAM_ENUMERATION(name, kSimulcastApiVersionNone,
kSimulcastApiVersionMax);
}
}
const ContentInfo* FindTransceiverMSection(
RtpTransceiverProxyWithInternal<RtpTransceiver>* transceiver,
const SessionDescriptionInterface* session_description) {
return transceiver->mid()
? session_description->description()->GetContentByName(
*transceiver->mid())
: nullptr;
}
} // namespace
// Upon completion, posts a task to execute the callback of the
// SetSessionDescriptionObserver asynchronously on the same thread. At this
// point, the state of the peer connection might no longer reflect the effects
// of the SetRemoteDescription operation, as the peer connection could have been
// modified during the post.
// TODO(hbos): Remove this class once we remove the version of
// PeerConnectionInterface::SetRemoteDescription() that takes a
// SetSessionDescriptionObserver as an argument.
class PeerConnection::SetRemoteDescriptionObserverAdapter
: public rtc::RefCountedObject<SetRemoteDescriptionObserverInterface> {
public:
SetRemoteDescriptionObserverAdapter(
rtc::scoped_refptr<PeerConnection> pc,
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper)
: pc_(std::move(pc)), wrapper_(std::move(wrapper)) {}
// SetRemoteDescriptionObserverInterface implementation.
void OnSetRemoteDescriptionComplete(RTCError error) override {
if (error.ok())
pc_->PostSetSessionDescriptionSuccess(wrapper_);
else
pc_->PostSetSessionDescriptionFailure(wrapper_, std::move(error));
}
private:
rtc::scoped_refptr<PeerConnection> pc_;
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper_;
};
bool PeerConnectionInterface::RTCConfiguration::operator==(
const PeerConnectionInterface::RTCConfiguration& o) const {
// This static_assert prevents us from accidentally breaking operator==.
// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
struct stuff_being_tested_for_equality {
IceServers servers;
IceTransportsType type;
BundlePolicy bundle_policy;
RtcpMuxPolicy rtcp_mux_policy;
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
int ice_candidate_pool_size;
bool disable_ipv6;
bool disable_ipv6_on_wifi;
int max_ipv6_networks;
bool disable_link_local_networks;
bool enable_rtp_data_channel;
absl::optional<int> screencast_min_bitrate;
absl::optional<bool> combined_audio_video_bwe;
absl::optional<bool> enable_dtls_srtp;
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;
bool audio_jitter_buffer_enable_rtx_handling;
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;
bool presume_writable_when_fully_relayed;
bool enable_ice_renomination;
bool redetermine_role_on_ice_restart;
absl::optional<int> ice_check_interval_strong_connectivity;
absl::optional<int> ice_check_interval_weak_connectivity;
absl::optional<int> ice_check_min_interval;
absl::optional<int> ice_unwritable_timeout;
absl::optional<int> ice_unwritable_min_checks;
absl::optional<int> ice_inactive_timeout;
absl::optional<int> stun_candidate_keepalive_interval;
absl::optional<rtc::IntervalRange> ice_regather_interval_range;
webrtc::TurnCustomizer* turn_customizer;
SdpSemantics sdp_semantics;
absl::optional<rtc::AdapterType> network_preference;
bool active_reset_srtp_params;
bool use_media_transport;
bool use_media_transport_for_data_channels;
bool use_datagram_transport;
absl::optional<CryptoOptions> crypto_options;
bool offer_extmap_allow_mixed;
};
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 &&
audio_jitter_buffer_enable_rtx_handling ==
o.audio_jitter_buffer_enable_rtx_handling &&
ice_connection_receiving_timeout ==
o.ice_connection_receiving_timeout &&
ice_backup_candidate_pair_ping_interval ==
o.ice_backup_candidate_pair_ping_interval &&
continual_gathering_policy == o.continual_gathering_policy &&
certificates == o.certificates &&
prioritize_most_likely_ice_candidate_pairs ==
o.prioritize_most_likely_ice_candidate_pairs &&
media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
max_ipv6_networks == o.max_ipv6_networks &&
disable_link_local_networks == o.disable_link_local_networks &&
enable_rtp_data_channel == o.enable_rtp_data_channel &&
screencast_min_bitrate == o.screencast_min_bitrate &&
combined_audio_video_bwe == o.combined_audio_video_bwe &&
enable_dtls_srtp == o.enable_dtls_srtp &&
ice_candidate_pool_size == o.ice_candidate_pool_size &&
prune_turn_ports == o.prune_turn_ports &&
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 &&
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 &&
ice_regather_interval_range == o.ice_regather_interval_range &&
turn_customizer == o.turn_customizer &&
sdp_semantics == o.sdp_semantics &&
network_preference == o.network_preference &&
active_reset_srtp_params == o.active_reset_srtp_params &&
use_media_transport == o.use_media_transport &&
use_media_transport_for_data_channels ==
o.use_media_transport_for_data_channels &&
use_datagram_transport == o.use_datagram_transport &&
crypto_options == o.crypto_options &&
offer_extmap_allow_mixed == o.offer_extmap_allow_mixed;
}
bool PeerConnectionInterface::RTCConfiguration::operator!=(
const PeerConnectionInterface::RTCConfiguration& o) const {
return !(*this == o);
}
// Generate a RTCP CNAME when a PeerConnection is created.
std::string GenerateRtcpCname() {
std::string cname;
if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
RTC_NOTREACHED();
}
return cname;
}
bool ValidateOfferAnswerOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
}
// From |rtc_options|, fill parts of |session_options| shared by all generated
// m= sections (in other words, nothing that involves a map/array).
void ExtractSharedMediaSessionOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
cricket::MediaSessionOptions* session_options) {
session_options->vad_enabled = rtc_options.voice_activity_detection;
session_options->bundle_enabled = rtc_options.use_rtp_mux;
}
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call)
: factory_(factory),
event_log_(std::move(event_log)),
event_log_ptr_(event_log_.get()),
rtcp_cname_(GenerateRtcpCname()),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()),
call_(std::move(call)),
call_ptr_(call_.get()) {}
PeerConnection::~PeerConnection() {
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
RTC_DCHECK_RUN_ON(signaling_thread());
// Need to stop transceivers before destroying the stats collector because
// AudioRtpSender has a reference to the StatsCollector it will update when
// stopping.
for (const auto& transceiver : transceivers_) {
transceiver->Stop();
}
stats_.reset(nullptr);
if (stats_collector_) {
stats_collector_->WaitForPendingRequest();
stats_collector_ = nullptr;
}
// Don't destroy BaseChannels until after stats has been cleaned up so that
// the last stats request can still read from the channels.
DestroyAllChannels();
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
webrtc_session_desc_factory_.reset();
sctp_invoker_.reset();
sctp_factory_.reset();
media_transport_invoker_.reset();
transport_controller_.reset();
// port_allocator_ lives on the network thread and should be destroyed there.
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_.reset();
});
// call_ and event_log_ must be destroyed on the worker thread.
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
call_.reset();
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
});
}
void PeerConnection::DestroyAllChannels() {
// Destroy video channels first since they may have a pointer to a voice
// channel.
for (const auto& transceiver : transceivers_) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
DestroyTransceiverChannel(transceiver);
}
}
for (const auto& transceiver : transceivers_) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
DestroyTransceiverChannel(transceiver);
}
}
DestroyDataChannel();
}
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
RTCError config_error = ValidateConfiguration(configuration);
if (!config_error.ok()) {
RTC_LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
return false;
}
if (!dependencies.allocator) {
RTC_LOG(LS_ERROR)
<< "PeerConnection initialized without a PortAllocator? "
"This shouldn't happen if using PeerConnectionFactory.";
return false;
}
if (!dependencies.observer) {
// TODO(deadbeef): Why do we do this?
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
"PeerConnectionObserver";
return false;
}
observer_ = dependencies.observer;
async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
port_allocator_ = std::move(dependencies.allocator);
tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
RTCErrorType parse_error =
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
if (parse_error != RTCErrorType::NONE) {
return false;
}
// The port allocator lives on the network thread and should be initialized
// there.
const auto pa_result =
network_thread()->Invoke<InitializePortAllocatorResult>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::InitializePortAllocator_n, this,
stun_servers, turn_servers, configuration));
// If initialization was successful, note if STUN or TURN servers
// were supplied.
if (!stun_servers.empty()) {
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
// Send information about IPv4/IPv6 status.
PeerConnectionAddressFamilyCounter address_family;
if (pa_result.enable_ipv6) {
address_family = kPeerConnection_IPv6;
} else {
address_family = kPeerConnection_IPv4;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
kPeerConnectionAddressFamilyCounter_Max);
const PeerConnectionFactoryInterface::Options& options = factory_->options();
// 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);
JsepTransportController::Config config;
config.redetermine_role_on_ice_restart =
configuration.redetermine_role_on_ice_restart;
config.ssl_max_version = factory_->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.event_log = event_log_ptr_;
#if defined(ENABLE_EXTERNAL_AUTH)
config.enable_external_auth = true;
#endif
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
if (configuration.use_datagram_transport ||
configuration.use_media_transport ||
configuration.use_media_transport_for_data_channels) {
if (!factory_->media_transport_factory()) {
RTC_DCHECK(false)
<< "PeerConnecton is initialized with use_media_transport = true or "
<< "use_media_transport_for_data_channels = true "
<< "but media transport factory is not set in PeerConnectionFactory";
return false;
}
if (configuration.use_media_transport ||
configuration.use_media_transport_for_data_channels) {
// TODO(bugs.webrtc.org/9719): This check will eventually go away, when
// RTP media transport is introduced. But until then, we require SDES to
// be enabled.
if (configuration.enable_dtls_srtp.has_value() &&
configuration.enable_dtls_srtp.value()) {
RTC_LOG(LS_WARNING)
<< "When media transport is used, SDES must be enabled. Set "
"configuration.enable_dtls_srtp to false. use_media_transport="
<< configuration.use_media_transport
<< ", use_media_transport_for_data_channels="
<< configuration.use_media_transport_for_data_channels;
return false;
}
}
config.use_media_transport_for_media = configuration.use_media_transport;
config.use_media_transport_for_data_channels =
configuration.use_media_transport_for_data_channels;
config.use_datagram_transport = configuration.use_datagram_transport;
config.media_transport_factory = factory_->media_transport_factory();
}
transport_controller_.reset(new JsepTransportController(
signaling_thread(), network_thread(), port_allocator_.get(),
async_resolver_factory_.get(), config));
transport_controller_->SignalIceConnectionState.connect(
this, &PeerConnection::OnTransportControllerConnectionState);
transport_controller_->SignalStandardizedIceConnectionState.connect(
this, &PeerConnection::SetStandardizedIceConnectionState);
transport_controller_->SignalConnectionState.connect(
this, &PeerConnection::SetConnectionState);
transport_controller_->SignalIceGatheringState.connect(
this, &PeerConnection::OnTransportControllerGatheringState);
transport_controller_->SignalIceCandidatesGathered.connect(
this, &PeerConnection::OnTransportControllerCandidatesGathered);
transport_controller_->SignalIceCandidatesRemoved.connect(
this, &PeerConnection::OnTransportControllerCandidatesRemoved);
transport_controller_->SignalDtlsHandshakeError.connect(
this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
sctp_factory_ = factory_->CreateSctpTransportInternalFactory();
stats_.reset(new StatsCollector(this));
stats_collector_ = RTCStatsCollector::Create(this);
configuration_ = configuration;
use_media_transport_ = configuration.use_media_transport;
// Obtain a certificate from RTCConfiguration if any were provided (optional).
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
if (!configuration.certificates.empty()) {
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
// just picking the first one. The decision should be made based on the DTLS
// handshake. The DTLS negotiations need to know about all certificates.
certificate = configuration.certificates[0];
}
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
if (options.disable_encryption) {
dtls_enabled_ = false;
} else {
// Enable DTLS by default if we have an identity store or a certificate.
dtls_enabled_ = (dependencies.cert_generator || certificate);
// |configuration| can override the default |dtls_enabled_| value.
if (configuration.enable_dtls_srtp) {
dtls_enabled_ = *(configuration.enable_dtls_srtp);
}
}
if (configuration.use_media_transport_for_data_channels) {
if (configuration.enable_rtp_data_channel) {
RTC_LOG(LS_ERROR) << "enable_rtp_data_channel and "
"use_media_transport_for_data_channels are "
"incompatible and cannot both be set to true";
return false;
}
data_channel_type_ = cricket::DCT_MEDIA_TRANSPORT;
} else if (configuration.enable_rtp_data_channel) {
// Enable creation of RTP data channels if the kEnableRtpDataChannels is
// set. It takes precendence over the disable_sctp_data_channels
// PeerConnectionFactoryInterface::Options.
data_channel_type_ = cricket::DCT_RTP;
} else {
// DTLS has to be enabled to use SCTP.
if (!options.disable_sctp_data_channels && dtls_enabled_) {
data_channel_type_ = cricket::DCT_SCTP;
}
}
video_options_.screencast_min_bitrate_kbps =
configuration.screencast_min_bitrate;
audio_options_.combined_audio_video_bwe =
configuration.combined_audio_video_bwe;
audio_options_.audio_jitter_buffer_max_packets =
configuration.audio_jitter_buffer_max_packets;
audio_options_.audio_jitter_buffer_fast_accelerate =
configuration.audio_jitter_buffer_fast_accelerate;
audio_options_.audio_jitter_buffer_min_delay_ms =
configuration.audio_jitter_buffer_min_delay_ms;
audio_options_.audio_jitter_buffer_enable_rtx_handling =
configuration.audio_jitter_buffer_enable_rtx_handling;
// Whether the certificate generator/certificate is null or not determines
// what PeerConnectionDescriptionFactory will do, so make sure that we give it
// the right instructions by clearing the variables if needed.
if (!dtls_enabled_) {
dependencies.cert_generator.reset();
certificate = nullptr;
} else if (certificate) {
// Favor generated certificate over the certificate generator.
dependencies.cert_generator.reset();
}
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager(), this, session_id(),
std::move(dependencies.cert_generator), certificate, &ssrc_generator_));
webrtc_session_desc_factory_->SignalCertificateReady.connect(
this, &PeerConnection::OnCertificateReady);
if (options.disable_encryption) {
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
webrtc_session_desc_factory_->set_is_unified_plan(IsUnifiedPlan());
// Add default audio/video transceivers for Plan B SDP.
if (!IsUnifiedPlan()) {
transceivers_.push_back(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
transceivers_.push_back(
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
}
int delay_ms =
return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
MSG_REPORT_USAGE_PATTERN, nullptr);
if (dependencies.video_bitrate_allocator_factory) {
video_bitrate_allocator_factory_ =
std::move(dependencies.video_bitrate_allocator_factory);
} else {
video_bitrate_allocator_factory_ =
CreateBuiltinVideoBitrateAllocatorFactory();
}
return true;
}
RTCError PeerConnection::ValidateConfiguration(
const RTCConfiguration& config) const {
if (config.ice_regather_interval_range &&
config.continual_gathering_policy == GATHER_ONCE) {
return RTCError(RTCErrorType::INVALID_PARAMETER,
"ice_regather_interval_range specified but continual "
"gathering policy is GATHER_ONCE");
}
auto result =
cricket::P2PTransportChannel::ValidateIceConfig(ParseIceConfig(config));
return result;
}
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 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 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 (IsClosed()) {
return false;
}
if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
return false;
}
local_streams_->AddStream(local_stream);
MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
observer->SignalAudioTrackAdded.connect(this,
&PeerConnection::OnAudioTrackAdded);
observer->SignalAudioTrackRemoved.connect(
this, &PeerConnection::OnAudioTrackRemoved);
observer->SignalVideoTrackAdded.connect(this,
&PeerConnection::OnVideoTrackAdded);
observer->SignalVideoTrackRemoved.connect(
this, &PeerConnection::OnVideoTrackRemoved);
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
for (const auto& track : local_stream->GetAudioTracks()) {
AddAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
AddVideoTrack(track.get(), local_stream);
}
stats_->AddStream(local_stream);
UpdateNegotiationNeeded();
return true;
}
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
"Plan SdpSemantics. Please use RemoveTrack "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
if (!IsClosed()) {
for (const auto& track : local_stream->GetAudioTracks()) {
RemoveAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
RemoveVideoTrack(track.get(), local_stream);
}
}
local_streams_->RemoveStream(local_stream);
stream_observers_.erase(
std::remove_if(
stream_observers_.begin(), stream_observers_.end(),
[local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
return observer->stream()->id().compare(local_stream->id()) == 0;
}),
stream_observers_.end());
if (IsClosed()) {
return;
}
UpdateNegotiationNeeded();
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
if (!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 (FindSenderForTrack(track)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Sender already exists for track " + track->id() + ".");
}
auto sender_or_error =
(IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids)
: AddTrackPlanB(track, stream_ids));
if (sender_or_error.ok()) {
UpdateNegotiationNeeded();
stats_->AddTrack(track);
}
return sender_or_error;
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackPlanB(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
if (stream_ids.size() > 1u) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
"AddTrack with more than one stream is not "
"supported with Plan B semantics.");
}
std::vector<std::string> adjusted_stream_ids = stream_ids;
if (adjusted_stream_ids.empty()) {
adjusted_stream_ids.push_back(rtc::CreateRandomUuid());
}
cricket::MediaType media_type =
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO);
auto new_sender =
CreateSender(media_type, track->id(), track, adjusted_stream_ids, {});
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
new_sender->internal()->SetMediaChannel(voice_media_channel());
GetAudioTransceiver()->internal()->AddSender(new_sender);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_audio_sender_infos_,
new_sender->internal()->stream_ids()[0], track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
} else {
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
new_sender->internal()->SetMediaChannel(video_media_channel());
GetVideoTransceiver()->internal()->AddSender(new_sender);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_video_sender_infos_,
new_sender->internal()->stream_ids()[0], track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
}
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
}
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
PeerConnection::AddTrackUnifiedPlan(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids) {
auto transceiver = FindFirstTransceiverForAddedTrack(track);
if (transceiver) {
RTC_LOG(LS_INFO) << "Reusing an existing "
<< cricket::MediaTypeToString(transceiver->media_type())
<< " transceiver for AddTrack.";
if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kSendRecv);
} else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
transceiver->internal()->set_direction(
RtpTransceiverDirection::kSendOnly);
}
transceiver->sender()->SetTrack(track);
transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
} else {
cricket::MediaType media_type =
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO);
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTrack.";
std::string sender_id = track->id();
// Avoid creating a sender with an existing ID by generating a random ID.
// This can happen if this is the second time AddTrack has created a sender
// for this track.
if (FindSenderById(sender_id)) {
sender_id = rtc::CreateRandomUuid();
}
auto sender = CreateSender(media_type, sender_id, track, stream_ids, {});
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
transceiver = CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_created_by_addtrack(true);
transceiver->internal()->set_direction(RtpTransceiverDirection::kSendRecv);
}
return transceiver->sender();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindFirstTransceiverForAddedTrack(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
RTC_DCHECK(track);
for (auto transceiver : transceivers_) {
if (!transceiver->sender()->track() &&
cricket::MediaTypeToString(transceiver->media_type()) ==
track->kind() &&
!transceiver->internal()->has_ever_been_used_to_send() &&
!transceiver->stopped()) {
return transceiver;
}
}
return nullptr;
}
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
return RemoveTrackNew(sender).ok();
}
RTCError PeerConnection::RemoveTrackNew(
rtc::scoped_refptr<RtpSenderInterface> sender) {
RTC_DCHECK_RUN_ON(signaling_thread());
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 = GetAudioTransceiver()->internal()->RemoveSender(sender);
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
removed = GetVideoTransceiver()->internal()->RemoveSender(sender);
}
if (!removed) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Couldn't find sender " + sender->id() + " to remove.");
}
}
UpdateNegotiationNeeded();
return RTCError::OK();
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindTransceiverBySender(
rtc::scoped_refptr<RtpSenderInterface> sender) {
for (auto transceiver : transceivers_) {
if (transceiver->sender() == sender) {
return transceiver;
}
}
return nullptr;
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
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());
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());
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((media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO));
if (track) {
RTC_DCHECK_EQ(media_type,
(track->kind() == MediaStreamTrackInterface::kAudioKind
? cricket::MEDIA_TYPE_AUDIO
: cricket::MEDIA_TYPE_VIDEO));
}
RTC_HISTOGRAM_COUNTS_LINEAR(kSimulcastNumberOfEncodings,
init.send_encodings.size(), 0, 7, 8);
size_t num_rids = absl::c_count_if(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !encoding.rid.empty();
});
if (num_rids > 0 && num_rids != init.send_encodings.size()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"RIDs must be provided for either all or none of the send encodings.");
}
if (num_rids > 0 && absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return !IsLegalRsidName(encoding.rid);
})) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Invalid RID value provided.");
}
if (absl::c_any_of(init.send_encodings,
[](const RtpEncodingParameters& encoding) {
return encoding.ssrc.has_value();
})) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
RtpParameters parameters;
parameters.encodings = init.send_encodings;
// Encodings are dropped from the tail if too many are provided.
if (parameters.encodings.size() > kMaxSimulcastStreams) {
parameters.encodings.erase(
parameters.encodings.begin() + kMaxSimulcastStreams,
parameters.encodings.end());
}
// Single RID should be removed.
if (parameters.encodings.size() == 1 &&
!parameters.encodings[0].rid.empty()) {
RTC_LOG(LS_INFO) << "Removing RID: " << parameters.encodings[0].rid << ".";
parameters.encodings[0].rid.clear();
}
// If RIDs were not provided, they are generated for simulcast scenario.
if (parameters.encodings.size() > 1 && num_rids == 0) {
rtc::UniqueStringGenerator rid_generator;
for (RtpEncodingParameters& encoding : parameters.encodings) {
encoding.rid = rid_generator();
}
}
if (UnimplementedRtpParameterHasValue(parameters)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::UNSUPPORTED_PARAMETER,
"Attempted to set an unimplemented parameter of RtpParameters.");
}
auto result = cricket::CheckRtpParametersValues(parameters);
if (!result.ok()) {
LOG_AND_RETURN_ERROR(result.type(), result.message());
}
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
<< " transceiver in response to a call to AddTransceiver.";
// Set the sender ID equal to the track ID if the track is specified unless
// that sender ID is already in use.
std::string sender_id =
(track && !FindSenderById(track->id()) ? track->id()
: rtc::CreateRandomUuid());
auto sender = CreateSender(media_type, sender_id, track, init.stream_ids,
parameters.encodings);
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
auto transceiver = CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_direction(init.direction);
if (update_negotiation_needed) {
UpdateNegotiationNeeded();
}
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::CreateSender(
cricket::MediaType media_type,
const std::string& id,
rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_ids,
const std::vector<RtpEncodingParameters>& send_encodings) {
RTC_DCHECK_RUN_ON(signaling_thread());
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kAudioKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(),
AudioRtpSender::Create(worker_thread(), id, stats_.get(), this));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
RTC_DCHECK(!track ||
(track->kind() == MediaStreamTrackInterface::kVideoKind));
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), VideoRtpSender::Create(worker_thread(), id, this));
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
bool set_track_succeeded = sender->SetTrack(track);
RTC_DCHECK(set_track_succeeded);
sender->internal()->set_stream_ids(stream_ids);
sender->internal()->set_init_send_encodings(send_encodings);
return sender;
}
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
PeerConnection::CreateReceiver(cricket::MediaType media_type,
const std::string& receiver_id) {
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
std::vector<std::string>({})));
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
} else {
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
std::vector<std::string>({})));
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
return receiver;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::CreateAndAddTransceiver(
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
receiver) {
// Ensure that the new sender does not have an ID that is already in use by
// another sender.
// Allow receiver IDs to conflict since those come from remote SDP (which
// could be invalid, but should not cause a crash).
RTC_DCHECK(!FindSenderById(sender->id()));
auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
signaling_thread(),
new RtpTransceiver(sender, receiver, channel_manager()));
transceivers_.push_back(transceiver);
transceiver->internal()->SignalNegotiationNeeded.connect(
this, &PeerConnection::OnNegotiationNeeded);
return transceiver;
}
void PeerConnection::OnNegotiationNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsClosed());
UpdateNegotiationNeeded();
}
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
const std::string& kind,
const std::string& stream_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
"Plan SdpSemantics. Please use AddTransceiver "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
if (IsClosed()) {
return nullptr;
}
// Internally we need to have one stream with Plan B semantics, so we
// generate a random stream ID if not specified.
std::vector<std::string> stream_ids;
if (stream_id.empty()) {
stream_ids.push_back(rtc::CreateRandomUuid());
RTC_LOG(LS_INFO)
<< "No stream_id specified for sender. Generated stream ID: "
<< stream_ids[0];
} else {
stream_ids.push_back(stream_id);
}
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
if (kind == MediaStreamTrackInterface::kAudioKind) {
auto audio_sender = AudioRtpSender::Create(
worker_thread(), rtc::CreateRandomUuid(), stats_.get(), this);
audio_sender->SetMediaChannel(voice_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), audio_sender);
GetAudioTransceiver()->internal()->AddSender(new_sender);
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
auto video_sender =
VideoRtpSender::Create(worker_thread(), rtc::CreateRandomUuid(), this);
video_sender->SetMediaChannel(video_media_channel());
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
signaling_thread(), video_sender);
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;
for (const auto& sender : GetSendersInternal()) {
ret.push_back(sender);
}
return ret;
}
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
PeerConnection::GetSendersInternal() const {
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
all_senders;
for (const auto& transceiver : transceivers_) {
auto senders = transceiver->internal()->senders();
all_senders.insert(all_senders.end(), senders.begin(), senders.end());
}
return all_senders;
}
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
PeerConnection::GetReceivers() const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
for (const auto& receiver : GetReceiversInternal()) {
ret.push_back(receiver);
}
return ret;
}
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
PeerConnection::GetReceiversInternal() const {
std::vector<
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
all_receivers;
for (const auto& transceiver : transceivers_) {
auto receivers = transceiver->internal()->receivers();
all_receivers.insert(all_receivers.end(), receivers.begin(),
receivers.end());
}
return all_receivers;
}
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;
for (const auto& transceiver : transceivers_) {
all_transceivers.push_back(transceiver);
}
return all_transceivers;
}
bool PeerConnection::GetStats(StatsObserver* observer,
MediaStreamTrackInterface* track,
StatsOutputLevel level) {
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!observer) {
RTC_LOG(LS_ERROR) << "GetStats - observer is NULL.";
return false;
}
stats_->UpdateStats(level);
// The StatsCollector is used to tell if a track is valid because it may
// remember tracks that the PeerConnection previously removed.
if (track && !stats_->IsValidTrack(track->id())) {
RTC_LOG(LS_WARNING) << "GetStats is called with an invalid track: "
<< track->id();
return false;
}
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
new GetStatsMsg(observer, 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);
stats_collector_->GetStatsReport(callback);
}
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::scoped_refptr<RtpSenderInternal> internal_sender;
if (selector) {
for (const auto& proxy_transceiver : transceivers_) {
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);
}
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::scoped_refptr<RtpReceiverInternal> internal_receiver;
if (selector) {
for (const auto& proxy_transceiver : transceivers_) {
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);
}
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
RTC_DCHECK_RUN_ON(signaling_thread());
return 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_;
}
rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
const std::string& label,
const DataChannelInit* config) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
bool first_datachannel = !HasDataChannels();
std::unique_ptr<InternalDataChannelInit> internal_config;
if (config) {
internal_config.reset(new InternalDataChannelInit(*config));
}
rtc::scoped_refptr<DataChannelInterface> channel(
InternalCreateDataChannel(label, internal_config.get()));
if (!channel.get()) {
return nullptr;
}
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
// the first SCTP DataChannel.
if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
UpdateNegotiationNeeded();
}
NoteUsageEvent(UsageEvent::DATA_ADDED);
return DataChannelProxy::Create(signaling_thread(), channel.get());
}
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
if (!observer) {
RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
if (IsClosed()) {
std::string error = "CreateOffer called when PeerConnection is closed.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
return;
}
if (!ValidateOfferAnswerOptions(options)) {
std::string error = "CreateOffer called with invalid options.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(
observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
return;
}
// Legacy handling for offer_to_receive_audio and offer_to_receive_video.
// Specified in WebRTC section 4.4.3.2 "Legacy configuration extensions".
if (IsUnifiedPlan()) {
RTCError error = HandleLegacyOfferOptions(options);
if (!error.ok()) {
PostCreateSessionDescriptionFailure(observer, std::move(error));
return;
}
}
cricket::MediaSessionOptions session_options;
GetOptionsForOffer(options, &session_options);
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
}
RTCError PeerConnection::HandleLegacyOfferOptions(
const RTCOfferAnswerOptions& options) {
RTC_DCHECK(IsUnifiedPlan());
if (options.offer_to_receive_audio == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MEDIA_TYPE_AUDIO);
} else if (options.offer_to_receive_audio == 1) {
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
} else if (options.offer_to_receive_audio > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_audio > 1 is not supported.");
}
if (options.offer_to_receive_video == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MEDIA_TYPE_VIDEO);
} else if (options.offer_to_receive_video == 1) {
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
} else if (options.offer_to_receive_video > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_video > 1 is not supported.");
}
return RTCError::OK();
}
void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
cricket::MediaType media_type) {
for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
RtpTransceiverDirection new_direction =
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
if (new_direction != transceiver->direction()) {
RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
<< " transceiver (MID="
<< transceiver->mid().value_or("<not set>") << ") from "
<< RtpTransceiverDirectionToString(
transceiver->direction())
<< " to "
<< RtpTransceiverDirectionToString(new_direction)
<< " since CreateOffer specified offer_to_receive=0";
transceiver->internal()->set_direction(new_direction);
}
}
}
void PeerConnection::AddUpToOneReceivingTransceiverOfType(
cricket::MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (GetReceivingTransceiversOfType(media_type).empty()) {
RTC_LOG(LS_INFO)
<< "Adding one recvonly " << cricket::MediaTypeToString(media_type)
<< " transceiver since CreateOffer specified offer_to_receive=1";
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kRecvOnly;
AddTransceiver(media_type, nullptr, init,
/*update_negotiation_needed=*/false);
}
}
std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
std::vector<
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
receiving_transceivers;
for (const auto& transceiver : transceivers_) {
if (!transceiver->stopped() && transceiver->media_type() == media_type &&
RtpTransceiverDirectionHasRecv(transceiver->direction())) {
receiving_transceivers.push_back(transceiver);
}
}
return receiving_transceivers;
}
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
const RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
if (!observer) {
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
if (!(signaling_state_ == kHaveRemoteOffer ||
signaling_state_ == kHaveLocalPrAnswer)) {
std::string error =
"PeerConnection cannot create an answer in a state other than "
"have-remote-offer or have-local-pranswer.";
RTC_LOG(LS_ERROR) << error;
PostCreateSessionDescriptionFailure(
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
return;
}
// The remote description should be set if we're in the right state.
RTC_DCHECK(remote_description());
if (IsUnifiedPlan()) {
if (options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
"supported with Unified Plan semantics. Use the "
"RtpTransceiver API instead.";
}
if (options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
"supported with Unified Plan semantics. Use the "
"RtpTransceiver API instead.";
}
}
cricket::MediaSessionOptions session_options;
GetOptionsForAnswer(options, &session_options);
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
}
void PeerConnection::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
// The SetLocalDescription contract is that we take ownership of the session
// description regardless of the outcome, so wrap it in a unique_ptr right
// away. Ideally, SetLocalDescription's signature will be changed to take the
// description as a unique_ptr argument to formalize this agreement.
std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr);
if (!observer) {
RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
return;
}
if (!desc) {
PostSetSessionDescriptionFailure(
observer,
RTCError(RTCErrorType::INTERNAL_ERROR, "SessionDescription is NULL."));
return;
}
// If a session error has occurred the PeerConnection is in a possibly
// inconsistent state so fail right away.
if (session_error() != SessionError::kNone) {
std::string error_message = GetSessionErrorMsg();
RTC_LOG(LS_ERROR) << "SetLocalDescription: " << error_message;
PostSetSessionDescriptionFailure(
observer,
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
if (!error.ok()) {
std::string error_message = GetSetDescriptionErrorMessage(
cricket::CS_LOCAL, desc->GetType(), error);
RTC_LOG(LS_ERROR) << error_message;
PostSetSessionDescriptionFailure(
observer,
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
// Grab the description type before moving ownership to ApplyLocalDescription,
// which may destroy it before returning.
const SdpType type = desc->GetType();
error = ApplyLocalDescription(std::move(desc));
// |desc| may be destroyed at this point.
if (!error.ok()) {
// If ApplyLocalDescription fails, the PeerConnection could be in an
// inconsistent state, so act conservatively here and set the session error
// so that future calls to SetLocalDescription/SetRemoteDescription fail.
SetSessionError(SessionError::kContent, error.message());
std::string error_message =
GetSetDescriptionErrorMessage(cricket::CS_LOCAL, type, error);
RTC_LOG(LS_ERROR) << error_message;
PostSetSessionDescriptionFailure(
observer,
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
RTC_DCHECK(local_description());
PostSetSessionDescriptionSuccess(observer);
// MaybeStartGathering needs to be called after posting
// MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
// before signaling that SetLocalDescription completed.
transport_controller_->MaybeStartGathering();
if (local_description()->GetType() == SdpType::kAnswer) {
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
// Make UMA notes about what was agreed to.
ReportNegotiatedSdpSemantics(*local_description());
}
if (IsUnifiedPlan()) {
bool was_negotiation_needed = is_negotiation_needed_;
UpdateNegotiationNeeded();
if (signaling_state() == kStable && was_negotiation_needed &&
is_negotiation_needed_) {
Observer()->OnRenegotiationNeeded();
}
}
NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED);
}
RTCError PeerConnection::ApplyLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
// Take a reference to the old local description since it's used below to
// compare against the new local description. When setting the new local
// description, grab ownership of the replaced session description in case it
// is the same as |old_local_description|, to keep it alive for the duration
// of the method.
const SessionDescriptionInterface* old_local_description =
local_description();
std::unique_ptr<SessionDescriptionInterface> replaced_local_description;
SdpType type = desc->GetType();
if (type == SdpType::kAnswer) {
replaced_local_description = pending_local_description_
? std::move(pending_local_description_)
: std::move(current_local_description_);
current_local_description_ = std::move(desc);
pending_local_description_ = nullptr;
current_remote_description_ = std::move(pending_remote_description_);
} else {
replaced_local_description = std::move(pending_local_description_);
pending_local_description_ = std::move(desc);
}
// The session description to apply now must be accessed by
// |local_description()|.
RTC_DCHECK(local_description());
// Report statistics about any use of simulcast.
ReportSimulcastApiVersion(kSimulcastVersionApplyLocalDescription,
*local_description()->description());
if (!is_caller_) {
if (remote_description()) {
// Remote description was applied first, so this PC is the callee.
is_caller_ = false;
} else {
// Local description is applied first, so this PC is the caller.
is_caller_ = true;
}
}
RTCError error = PushdownTransportDescription(cricket::CS_LOCAL, type);
if (!error.ok()) {
return error;
}
if (IsUnifiedPlan()) {
RTCError error = UpdateTransceiversAndDataChannels(
cricket::CS_LOCAL, *local_description(), old_local_description,
remote_description());
if (!error.ok()) {
return error;
}
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
for (const auto& transceiver : transceivers_) {
// 2.2.7.1.1.(6-9): Set sender and receiver's transport slots.
// Note that code paths that don't set MID won't be able to use
// information about DTLS transports.
if (transceiver->mid()) {
auto dtls_transport =
LookupDtlsTransportByMidInternal(*transceiver->mid());
transceiver->internal()->sender_internal()->set_transport(
dtls_transport);
transceiver->internal()->receiver_internal()->set_transport(
dtls_transport);
}
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
const MediaContentDescription* media_desc = content->media_description();
// 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
// the following steps:
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
// 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
// transceiver's [[FiredDirection]] slot is either "sendrecv" or
// "recvonly", process the removal of a remote track for the media
// description, given transceiver, removeList, and muteTracks.
if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
(transceiver->internal()->fired_direction() &&
RtpTransceiverDirectionHasRecv(
*transceiver->internal()->fired_direction()))) {
ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
&removed_streams);
}
// 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
// [[FiredDirection]] slots to direction.
transceiver->internal()->set_current_direction(media_desc->direction());
transceiver->internal()->set_fired_direction(media_desc->direction());
}
}
auto observer = Observer();
for (const auto& transceiver : remove_list) {
observer->OnRemoveTrack(transceiver->receiver());
}
for (const auto& stream : removed_streams) {
observer->OnRemoveStream(stream);
}
} else {
// Media channels will be created only when offer is set. These may use new
// transports just created by PushdownTransportDescription.
if (type == SdpType::kOffer) {
// TODO(bugs.webrtc.org/4676) - Handle CreateChannel failure, as new local
// description is applied. Restore back to old description.
RTCError error = CreateChannels(*local_description()->description());
if (!error.ok()) {
return error;
}
}
// Remove unused channels if MediaContentDescription is rejected.
RemoveUnusedChannels(local_description()->description());
}
error = UpdateSessionState(type, cricket::CS_LOCAL,
local_description()->description());
if (!error.ok()) {
return error;
}
if (remote_description()) {
// Now that we have a local description, we can push down remote candidates.
UseCandidatesInSessionDescription(remote_description());
}
pending_ice_restarts_.clear();
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (DataChannel::IsSctpLike(data_channel_type_) && GetSctpSslRole(&role)) {
AllocateSctpSids(role);
}
if (IsUnifiedPlan()) {
for (const auto& transceiver : transceivers_) {
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (content->rejected || !channel || channel->local_streams().empty()) {
// 0 is a special value meaning "this sender has no associated send
// stream". Need to call this so the sender won't attempt to configure
// a no longer existing stream and run into DCHECKs in the lower
// layers.
transceiver->internal()->sender_internal()->SetSsrc(0);
} else {
// Get the StreamParams from the channel which could generate SSRCs.
const std::vector<StreamParams>& streams = channel->local_streams();
transceiver->internal()->sender_internal()->set_stream_ids(
streams[0].stream_ids());
transceiver->internal()->sender_internal()->SetSsrc(
streams[0].first_ssrc());
}
}
} else {
// Plan B semantics.
// Update state and SSRC of local MediaStreams and DataChannels based on the
// local session description.
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(local_description()->description());
if (audio_content) {
if (audio_content->rejected) {
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
} else {
const cricket::AudioContentDescription* audio_desc =
audio_content->media_description()->as_audio();
UpdateLocalSenders(audio_desc->streams(), audio_desc->type());
}
}
const cricket::ContentInfo* video_content =
GetFirstVideoContent(local_description()->description());
if (video_content) {
if (video_content->rejected) {
RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
} else {
const cricket::VideoContentDescription* video_desc =
video_content->media_description()->as_video();
UpdateLocalSenders(video_desc->streams(), video_desc->type());
}
}
}
const cricket::ContentInfo* data_content =
GetFirstDataContent(local_description()->description());
if (data_content) {
const cricket::RtpDataContentDescription* rtp_data_desc =
data_content->media_description()->as_rtp_data();
// rtp_data_desc will be null if this is an SCTP description.
if (rtp_data_desc) {
UpdateLocalRtpDataChannels(rtp_data_desc->streams());
}
}
return RTCError::OK();
}
// The SDP parser used to populate these values by default for the 'content
// name' if an a=mid line was absent.
static absl::string_view GetDefaultMidForPlanB(cricket::MediaType media_type) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
return cricket::CN_AUDIO;
case cricket::MEDIA_TYPE_VIDEO:
return cricket::CN_VIDEO;
case cricket::MEDIA_TYPE_DATA:
return cricket::CN_DATA;
}
RTC_NOTREACHED();
return "";
}
void PeerConnection::FillInMissingRemoteMids(
cricket::SessionDescription* new_remote_description) {
RTC_DCHECK(new_remote_description);
const cricket::ContentInfos& local_contents =
(local_description() ? local_description()->description()->contents()
: cricket::ContentInfos());
const cricket::ContentInfos& remote_contents =
(remote_description() ? remote_description()->description()->contents()
: cricket::ContentInfos());
for (size_t i = 0; i < new_remote_description->contents().size(); ++i) {
cricket::ContentInfo& content = new_remote_description->contents()[i];
if (!content.name.empty()) {
continue;
}
std::string new_mid;
absl::string_view source_explanation;
if (IsUnifiedPlan()) {
if (i < local_contents.size()) {
new_mid = local_contents[i].name;
source_explanation = "from the matching local media section";
} else if (i < remote_contents.size()) {
new_mid = remote_contents[i].name;
source_explanation = "from the matching previous remote media section";
} else {
new_mid = mid_generator_();
source_explanation = "generated just now";
}
} else {
new_mid = std::string(
GetDefaultMidForPlanB(content.media_description()->type()));
source_explanation = "to match pre-existing behavior";
}
RTC_DCHECK(!new_mid.empty());
content.name = new_mid;
new_remote_description->transport_infos()[i].content_name = new_mid;
RTC_LOG(LS_INFO) << "SetRemoteDescription: Remote media section at i=" << i
<< " is missing an a=mid line. Filling in the value '"
<< new_mid << "' " << source_explanation << ".";
}
}
void PeerConnection::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc) {
SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface>(desc),
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>(
new SetRemoteDescriptionObserverAdapter(this, observer)));
}
void PeerConnection::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
if (!observer) {
RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
return;
}
if (!desc) {
observer->OnSetRemoteDescriptionComplete(RTCError(
RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL."));
return;
}
// If a session error has occurred the PeerConnection is in a possibly
// inconsistent state so fail right away.
if (session_error() != SessionError::kNone) {
std::string error_message = GetSessionErrorMsg();
RTC_LOG(LS_ERROR) << "SetRemoteDescription: " << error_message;
observer->OnSetRemoteDescriptionComplete(
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
if (desc->GetType() == SdpType::kOffer) {
// Report to UMA the format of the received offer.
ReportSdpFormatReceived(*desc);
}
// Handle remote descriptions missing a=mid lines for interop with legacy end
// points.
FillInMissingRemoteMids(desc->description());
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
if (!error.ok()) {
std::string error_message = GetSetDescriptionErrorMessage(
cricket::CS_REMOTE, desc->GetType(), error);
RTC_LOG(LS_ERROR) << error_message;
observer->OnSetRemoteDescriptionComplete(
RTCError(error.type(), std::move(error_message)));
return;
}
// Grab the description type before moving ownership to
// ApplyRemoteDescription, which may destroy it before returning.
const SdpType type = desc->GetType();
error = ApplyRemoteDescription(std::move(desc));
// |desc| may be destroyed at this point.
if (!error.ok()) {
// If ApplyRemoteDescription fails, the PeerConnection could be in an
// inconsistent state, so act conservatively here and set the session error
// so that future calls to SetLocalDescription/SetRemoteDescription fail.
SetSessionError(SessionError::kContent, error.message());
std::string error_message =
GetSetDescriptionErrorMessage(cricket::CS_REMOTE, type, error);
RTC_LOG(LS_ERROR) << error_message;
observer->OnSetRemoteDescriptionComplete(
RTCError(error.type(), std::move(error_message)));
return;
}
RTC_DCHECK(remote_description());
if (type == SdpType::kAnswer) {
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
// Make UMA notes about what was agreed to.
ReportNegotiatedSdpSemantics(*remote_description());
}
if (IsUnifiedPlan()) {
bool was_negotiation_needed = is_negotiation_needed_;
UpdateNegotiationNeeded();
if (signaling_state() == kStable && was_negotiation_needed &&
is_negotiation_needed_) {
Observer()->OnRenegotiationNeeded();
}
}
observer->OnSetRemoteDescriptionComplete(RTCError::OK());
NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED);
}
RTCError PeerConnection::ApplyRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
// Update stats here so that we have the most recent stats for tracks and
// streams that might be removed by updating the session description.
stats_->UpdateStats(kStatsOutputLevelStandard);
// Take a reference to the old remote description since it's used below to
// compare against the new remote description. When setting the new remote
// description, grab ownership of the replaced session description in case it
// is the same as |old_remote_description|, to keep it alive for the duration
// of the method.
const SessionDescriptionInterface* old_remote_description =
remote_description();
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
SdpType type = desc->GetType();
if (type == SdpType::kAnswer) {
replaced_remote_description = pending_remote_description_
? std::move(pending_remote_description_)
: std::move(current_remote_description_);
current_remote_description_ = std::move(desc);
pending_remote_description_ = nullptr;
current_local_description_ = std::move(pending_local_description_);
} else {
replaced_remote_description = std::move(pending_remote_description_);
pending_remote_description_ = std::move(desc);
}
// The session description to apply now must be accessed by
// |remote_description()|.
RTC_DCHECK(remote_description());
// Report statistics about any use of simulcast.
ReportSimulcastApiVersion(kSimulcastVersionApplyRemoteDescription,
*remote_description()->description());
RTCError error = PushdownTransportDescription(cricket::CS_REMOTE, type);
if (!error.ok()) {
return error;
}
// Transport and Media channels will be created only when offer is set.
if (IsUnifiedPlan()) {
RTCError error = UpdateTransceiversAndDataChannels(
cricket::CS_REMOTE, *remote_description(), local_description(),
old_remote_description);
if (!error.ok()) {
return error;
}
} else {
// Media channels will be created only when offer is set. These may use new
// transports just created by PushdownTransportDescription.
if (type == SdpType::kOffer) {
// TODO(mallinath) - Handle CreateChannel failure, as new local
// description is applied. Restore back to old description.
RTCError error = CreateChannels(*remote_description()->description());
if (!error.ok()) {
return error;
}
}
// Remove unused channels if MediaContentDescription is rejected.
RemoveUnusedChannels(remote_description()->description());
}
// NOTE: Candidates allocation will be initiated only when
// SetLocalDescription is called.
error = UpdateSessionState(type, cricket::CS_REMOTE,
remote_description()->description());
if (!error.ok()) {
return error;
}
if (local_description() &&
!UseCandidatesInSessionDescription(remote_description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidCandidates);
}
if (old_remote_description) {
for (const cricket::ContentInfo& content :
old_remote_description->description()->contents()) {
// Check if this new SessionDescription contains new ICE ufrag and
// password that indicates the remote peer requests an ICE restart.
// TODO(deadbeef): When we start storing both the current and pending
// remote description, this should reset pending_ice_restarts and compare
// against the current description.
if (CheckForRemoteIceRestart(old_remote_description, remote_description(),
content.name)) {
if (type == SdpType::kOffer) {
pending_ice_restarts_.insert(content.name);
}
} else {
// We retain all received candidates only if ICE is not restarted.
// When ICE is restarted, all previous candidates belong to an old
// generation and should not be kept.
// TODO(deadbeef): This goes against the W3C spec which says the remote
// description should only contain candidates from the last set remote
// description plus any candidates added since then. We should remove
// this once we're sure it won't break anything.
WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
old_remote_description, content.name, mutable_remote_description());
}
}
}
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
// Set the the ICE connection state to connecting since the connection may
// become writable with peer reflexive candidates before any remote candidate
// is signaled.
// TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
// is to have a new signal the indicates a change in checking state from the
// transport and expose a new checking() member from transport that can be
// read to determine the current checking state. The existing SignalConnecting
// actually means "gathering candidates", so cannot be be used here.
if (remote_description()->GetType() != SdpType::kOffer &&
remote_description()->number_of_mediasections() > 0u &&
ice_connection_state() == PeerConnectionInterface::kIceConnectionNew) {
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
rtc::SSLRole role;
if (DataChannel::IsSctpLike(data_channel_type_) && GetSctpSslRole(&role)) {
AllocateSctpSids(role);
}
if (IsUnifiedPlan()) {
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
now_receiving_transceivers;
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
for (const auto& transceiver : transceivers_) {
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, remote_description());
if (!content) {
continue;
}
const MediaContentDescription* media_desc = content->media_description();
RtpTransceiverDirection local_direction =
RtpTransceiverDirectionReversed(media_desc->direction());
// Roughly the same as steps 2.2.8.6 of section 4.4.1.6 "Set the
// RTCSessionDescription: Set the associated remote streams given
// transceiver.[[Receiver]], msids, addList, and removeList".
// https://w3c.github.io/webrtc-pc/#set-the-rtcsessiondescription
if (RtpTransceiverDirectionHasRecv(local_direction)) {
std::vector<std::string> stream_ids;
if (!media_desc->streams().empty()) {
// The remote description has signaled the stream IDs.
stream_ids = media_desc->streams()[0].stream_ids();
}
RTC_LOG(LS_INFO) << "Processing the MSIDs for MID=" << content->name
<< " (" << GetStreamIdsString(stream_ids) << ").";
SetAssociatedRemoteStreams(transceiver->internal()->receiver_internal(),
stream_ids, &added_streams,
&removed_streams);
// From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6
// "Set the RTCSessionDescription: If direction is sendrecv or recvonly,
// and transceiver's current direction is neither sendrecv nor recvonly,
// process the addition of a remote track for the media description.
if (!transceiver->fired_direction() ||
!RtpTransceiverDirectionHasRecv(*transceiver->fired_direction())) {
RTC_LOG(LS_INFO)
<< "Processing the addition of a remote track for MID="
<< content->name << ".";
now_receiving_transceivers.push_back(transceiver);
}
}
// 2.2.8.1.9: If direction is "sendonly" or "inactive", and transceiver's
// [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the
// removal of a remote track for the media description, given transceiver,
// removeList, and muteTracks.
if (!RtpTransceiverDirectionHasRecv(local_direction) &&
(transceiver->fired_direction() &&
RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
&removed_streams);
}
// 2.2.8.1.10: Set transceiver's [[FiredDirection]] slot to direction.
transceiver->internal()->set_fired_direction(local_direction);
// 2.2.8.1.11: If description is of type "answer" or "pranswer", then run
// the following steps:
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
// 2.2.8.1.11.1: Set transceiver's [[CurrentDirection]] slot to
// direction.
transceiver->internal()->set_current_direction(local_direction);
// 2.2.8.1.11.[3-6]: Set the transport internal slots.
if (transceiver->mid()) {
auto dtls_transport =
LookupDtlsTransportByMidInternal(*transceiver->mid());
transceiver->internal()->sender_internal()->set_transport(
dtls_transport);
transceiver->internal()->receiver_internal()->set_transport(
dtls_transport);
}
}
// 2.2.8.1.12: If the media description is rejected, and transceiver is
// not already stopped, stop the RTCRtpTransceiver transceiver.
if (content->rejected && !transceiver->stopped()) {
RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->name
<< " since the media section was rejected.";
transceiver->Stop();
}
if (!content->rejected &&
RtpTransceiverDirectionHasRecv(local_direction)) {
// Set ssrc to 0 in the case of an unsignalled ssrc.
uint32_t ssrc = 0;
if (!media_desc->streams().empty() &&
media_desc->streams()[0].has_ssrcs()) {
ssrc = media_desc->streams()[0].first_ssrc();
}
transceiver->internal()->receiver_internal()->SetupMediaChannel(ssrc);
}
}
// Once all processing has finished, fire off callbacks.
auto observer = Observer();
for (const auto& transceiver : now_receiving_transceivers) {
stats_->AddTrack(transceiver->receiver()->track());
observer->OnTrack(transceiver);
observer->OnAddTrack(transceiver->receiver(),
transceiver->receiver()->streams());
}
for (const auto& stream : added_streams) {
observer->OnAddStream(stream);
}
for (const auto& transceiver : remove_list) {
observer->OnRemoveTrack(transceiver->receiver());
}
for (const auto& stream : removed_streams) {
observer->OnRemoveStream(stream);
}
}
const cricket::ContentInfo* audio_content =
GetFirstAudioContent(remote_description()->description());
const cricket::ContentInfo* video_content =
GetFirstVideoContent(remote_description()->description());
const cricket::AudioContentDescription* audio_desc =
GetFirstAudioContentDescription(remote_description()->description());
const cricket::VideoContentDescription* video_desc =
GetFirstVideoContentDescription(remote_description()->description());
const cricket::RtpDataContentDescription* rtp_data_desc =
GetFirstRtpDataContentDescription(remote_description()->description());
// Check if the descriptions include streams, just in case the peer supports
// MSID, but doesn't indicate so with "a=msid-semantic".
if (remote_description()->description()->msid_supported() ||
(audio_desc && !audio_desc->streams().empty()) ||
(video_desc && !video_desc->streams().empty())) {
remote_peer_supports_msid_ = true;
}
// We wait to signal new streams until we finish processing the description,
// since only at that point will new streams have all their tracks.
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
if (!IsUnifiedPlan()) {
// TODO(steveanton): When removing RTP senders/receivers in response to a
// rejected media section, there is some cleanup logic that expects the
// voice/ video channel to still be set. But in this method the voice/video
// channel would have been destroyed by the SetRemoteDescription caller
// above so the cleanup that relies on them fails to run. The RemoveSenders
// calls should be moved to right before the DestroyChannel calls to fix
// this.
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
if (audio_content) {
if (audio_content->rejected) {
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
} else {
bool default_audio_track_needed =
!remote_peer_supports_msid_ &&
RtpTransceiverDirectionHasSend(audio_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(audio_desc),
default_audio_track_needed, audio_desc->type(),
new_streams);
}
}
// Find all video rtp streams and create corresponding remote VideoTracks
// and MediaStreams.
if (video_content) {
if (video_content->rejected) {
RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
} else {
bool default_video_track_needed =
!remote_peer_supports_msid_ &&
RtpTransceiverDirectionHasSend(video_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(video_desc),
default_video_track_needed, video_desc->type(),
new_streams);
}
}
// If this is an RTP data transport, update the DataChannels with the
// information from the remote peer.
if (rtp_data_desc) {
UpdateRemoteRtpDataChannels(GetActiveStreams(rtp_data_desc));
}
// Iterate new_streams and notify the observer about new MediaStreams.
auto observer = Observer();
for (size_t i = 0; i < new_streams->count(); ++i) {
MediaStreamInterface* new_stream = new_streams->at(i);
stats_->AddStream(new_stream);
observer->OnAddStream(
rtc::scoped_refptr<MediaStreamInterface>(new_stream));
}
UpdateEndedRemoteMediaStreams();
}
return RTCError::OK();
}
void PeerConnection::SetAssociatedRemoteStreams(
rtc::scoped_refptr<RtpReceiverInternal> receiver,
const std::vector<std::string>& stream_ids,
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* added_streams,
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams;
for (const std::string& stream_id : stream_ids) {
rtc::scoped_refptr<MediaStreamInterface> stream =
remote_streams_->find(stream_id);
if (!stream) {
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
MediaStream::Create(stream_id));
remote_streams_->AddStream(stream);
added_streams->push_back(stream);
}
media_streams.push_back(stream);
}
// Special case: "a=msid" missing, use random stream ID.
if (media_streams.empty() &&
!(remote_description()->description()->msid_signaling() &
cricket::kMsidSignalingMediaSection)) {
if (!missing_msid_default_stream_) {
missing_msid_default_stream_ = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(rtc::CreateRandomUuid()));
added_streams->push_back(missing_msid_default_stream_);
}
media_streams.push_back(missing_msid_default_stream_);
}
std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams =
receiver->streams();
// SetStreams() will add/remove the receiver's track to/from the streams. This
// differs from the spec - the spec uses an "addList" and "removeList" to
// update the stream-track relationships in a later step. We do this earlier,
// changing the order of things, but the end-result is the same.
// TODO(hbos): When we remove remote_streams(), use set_stream_ids()
// instead. https://crbug.com/webrtc/9480
receiver->SetStreams(media_streams);
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
}
void PeerConnection::ProcessRemovalOfRemoteTrack(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
RTC_DCHECK(transceiver->mid());
RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
<< *transceiver->mid();
std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams =
transceiver->internal()->receiver_internal()->streams();
// This will remove the remote track from the streams.
transceiver->internal()->receiver_internal()->set_stream_ids({});
remove_list->push_back(transceiver);
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
}
void PeerConnection::RemoveRemoteStreamsIfEmpty(
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& remote_streams,
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
// TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead of
// streams, see if the stream was removed by checking if this was the last
// receiver with that stream ID.
for (const auto& remote_stream : remote_streams) {
if (remote_stream->GetAudioTracks().empty() &&
remote_stream->GetVideoTracks().empty()) {
remote_streams_->RemoveStream(remote_stream);
removed_streams->push_back(remote_stream);
}
}
}
RTCError PeerConnection::UpdateTransceiversAndDataChannels(
cricket::ContentSource source,
const SessionDescriptionInterface& new_session,
const SessionDescriptionInterface* old_local_description,
const SessionDescriptionInterface* old_remote_description) {
RTC_DCHECK(IsUnifiedPlan());
const cricket::ContentGroup* bundle_group = nullptr;
if (new_session.GetType() == SdpType::kOffer) {
auto bundle_group_or_error =
GetEarlyBundleGroup(*new_session.description());
if (!bundle_group_or_error.ok()) {
return bundle_group_or_error.MoveError();
}
bundle_group = bundle_group_or_error.MoveValue();
}
const ContentInfos& new_contents = new_session.description()->contents();
for (size_t i = 0; i < new_contents.size(); ++i) {
const cricket::ContentInfo& new_content = new_contents[i];
cricket::MediaType media_type = new_content.media_description()->type();
mid_generator_.AddKnownId(new_content.name);
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO) {
const cricket::ContentInfo* old_local_content = nullptr;
if (old_local_description &&
i < old_local_description->description()->contents().size()) {
old_local_content =
&old_local_description->description()->contents()[i];
}
const cricket::ContentInfo* old_remote_content = nullptr;
if (old_remote_description &&
i < old_remote_description->description()->contents().size()) {
old_remote_content =
&old_remote_description->description()->contents()[i];
}
auto transceiver_or_error =
AssociateTransceiver(source, new_session.GetType(), i, new_content,
old_local_content, old_remote_content);
if (!transceiver_or_error.ok()) {
return transceiver_or_error.MoveError();
}
auto transceiver = transceiver_or_error.MoveValue();
RTCError error =
UpdateTransceiverChannel(transceiver, new_content, bundle_group);
if (!error.ok()) {
return error;
}
} else if (media_type == cricket::MEDIA_TYPE_DATA) {
if (GetDataMid() && new_content.name != *GetDataMid()) {
// Ignore all but the first data section.
RTC_LOG(LS_INFO) << "Ignoring data media section with MID="
<< new_content.name;
continue;
}
RTCError error = UpdateDataChannel(source, new_content, bundle_group);
if (!error.ok()) {
return error;
}
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Unknown section type.");
}
}
return RTCError::OK();
}
RTCError PeerConnection::UpdateTransceiverChannel(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
const cricket::ContentInfo& content,
const cricket::ContentGroup* bundle_group) {
RTC_DCHECK(IsUnifiedPlan());
RTC_DCHECK(transceiver);
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (content.rejected) {
if (channel) {
transceiver->internal()->SetChannel(nullptr);
DestroyChannelInterface(channel);
}
} else {
if (!channel) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
channel = CreateVoiceChannel(content.name);
} else {
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
channel = CreateVideoChannel(content.name);
}
if (!channel) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create channel for mid=" + content.name);
}
transceiver->internal()->SetChannel(channel);
}
}
return RTCError::OK();
}
RTCError PeerConnection::UpdateDataChannel(
cricket::ContentSource source,
const cricket::ContentInfo& content,
const cricket::ContentGroup* bundle_group) {
if (data_channel_type_ == cricket::DCT_NONE) {
// If data channels are disabled, ignore this media section. CreateAnswer
// will take care of rejecting it.
return RTCError::OK();
}
if (content.rejected) {
DestroyDataChannel();
} else {
if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
if (!CreateDataChannel(content.name)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create data channel.");
}
}
if (source == cricket::CS_REMOTE) {
const MediaContentDescription* data_desc = content.media_description();
if (data_desc && cricket::IsRtpProtocol(data_desc->protocol())) {
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
}
}
}
return RTCError::OK();
}
// This method will extract any send encodings that were sent by the remote
// connection. This is currently only relevant for Simulcast scenario (where
// the number of layers may be communicated by the server).
static std::vector<RtpEncodingParameters> GetSendEncodingsFromRemoteDescription(
const MediaContentDescription& desc) {
if (!desc.HasSimulcast()) {
return {};
}
std::vector<RtpEncodingParameters> result;
const SimulcastDescription& simulcast = desc.simulcast_description();
// This is a remote description, the parameters we are after should appear
// as receive streams.
for (const auto& alternatives : simulcast.receive_layers()) {
RTC_DCHECK(!alternatives.empty());
// There is currently no way to specify or choose from alternatives.
// We will always use the first alternative, which is the most preferred.
const SimulcastLayer& layer = alternatives[0];
RtpEncodingParameters parameters;
parameters.rid = layer.rid;
parameters.active = !layer.is_paused;
result.push_back(parameters);
}
return result;
}
static RTCError UpdateSimulcastLayerStatusInSender(
const std::vector<SimulcastLayer>& layers,
rtc::scoped_refptr<RtpSenderInternal> sender) {
RTC_DCHECK(sender);
RtpParameters parameters = sender->GetParametersInternal();
std::vector<std::string> disabled_layers;
// The simulcast envelope cannot be changed, only the status of the streams.
// So we will iterate over the send encodings rather than the layers.
for (RtpEncodingParameters& encoding : parameters.encodings) {
auto iter = std::find_if(layers.begin(), layers.end(),
[&encoding](const SimulcastLayer& layer) {
return layer.rid == encoding.rid;
});
// A layer that cannot be found may have been removed by the remote party.
if (iter == layers.end()) {
disabled_layers.push_back(encoding.rid);
continue;
}
encoding.active = !iter->is_paused;
}
RTCError result = sender->SetParametersInternal(parameters);
if (result.ok()) {
result = sender->DisableEncodingLayers(disabled_layers);
}
return result;
}
static bool SimulcastIsRejected(
const ContentInfo* local_content,
const MediaContentDescription& answer_media_desc) {
bool simulcast_offered = local_content &&
local_content->media_description() &&
local_content->media_description()->HasSimulcast();
bool simulcast_answered = answer_media_desc.HasSimulcast();
bool rids_supported = RtpExtension::FindHeaderExtensionByUri(
answer_media_desc.rtp_header_extensions(), RtpExtension::kRidUri);
return simulcast_offered && (!simulcast_answered || !rids_supported);
}
static RTCError DisableSimulcastInSender(
rtc::scoped_refptr<RtpSenderInternal> sender) {
RTC_DCHECK(sender);
RtpParameters parameters = sender->GetParametersInternal();
if (parameters.encodings.size() <= 1) {
return RTCError::OK();
}
std::vector<std::string> disabled_layers;
std::transform(
parameters.encodings.begin() + 1, parameters.encodings.end(),
std::back_inserter(disabled_layers),
[](const RtpEncodingParameters& encoding) { return encoding.rid; });
return sender->DisableEncodingLayers(disabled_layers);
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
PeerConnection::AssociateTransceiver(cricket::ContentSource source,
SdpType type,
size_t mline_index,
const ContentInfo& content,
const ContentInfo* old_local_content,
const ContentInfo* old_remote_content) {
RTC_DCHECK(IsUnifiedPlan());
// If this is an offer then the m= section might be recycled. If the m=
// section is being recycled (defined as: rejected in the current local or
// remote description and not rejected in new description), dissociate the
// currently associated RtpTransceiver by setting its mid property to null,
// and discard the mapping between the transceiver and its m= section index.
if (IsMediaSectionBeingRecycled(type, content, old_local_content,
old_remote_content)) {
// We want to dissociate the transceiver that has the rejected mid.
const std::string& old_mid =
(old_local_content && old_local_content->rejected)
? old_local_content->name
: old_remote_content->name;
auto old_transceiver = GetAssociatedTransceiver(old_mid);
if (old_transceiver) {
RTC_LOG(LS_INFO) << "Dissociating transceiver for MID=" << old_mid
<< " since the media section is being recycled.";
old_transceiver->internal()->set_mid(absl::nullopt);
old_transceiver->internal()->set_mline_index(absl::nullopt);
}
}
const MediaContentDescription* media_desc = content.media_description();
auto transceiver = GetAssociatedTransceiver(content.name);
if (source == cricket::CS_LOCAL) {
// Find the RtpTransceiver that corresponds to this m= section, using the
// mapping between transceivers and m= section indices established when
// creating the offer.
if (!transceiver) {
transceiver = GetTransceiverByMLineIndex(mline_index);
}
if (!transceiver) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Unknown transceiver");
}
} else {
RTC_DCHECK_EQ(source, cricket::CS_REMOTE);
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers
// of the same type...
// When simulcast is requested, a transceiver cannot be associated because
// AddTrack cannot be called to initialize it.
if (!transceiver &&
RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
!media_desc->HasSimulcast()) {
transceiver = FindAvailableTransceiverToReceive(media_desc->type());
}
// If no RtpTransceiver was found in the previous step, create one with a
// recvonly direction.
if (!transceiver) {
RTC_LOG(LS_INFO) << "Adding "
<< cricket::MediaTypeToString(media_desc->type())
<< " transceiver for MID=" << content.name
<< " at i=" << mline_index
<< " in response to the remote description.";
std::string sender_id = rtc::CreateRandomUuid();
std::vector<RtpEncodingParameters> send_encodings =
GetSendEncodingsFromRemoteDescription(*media_desc);
auto sender = CreateSender(media_desc->type(), sender_id, nullptr, {},
send_encodings);
std::string receiver_id;
if (!media_desc->streams().empty()) {
receiver_id = media_desc->streams()[0].id;
} else {
receiver_id = rtc::CreateRandomUuid();
}
auto receiver = CreateReceiver(media_desc->type(), receiver_id);
transceiver = CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_direction(
RtpTransceiverDirection::kRecvOnly);
}
// Check if the offer indicated simulcast but the answer rejected it.
// This can happen when simulcast is not supported on the remote party.
if (SimulcastIsRejected(old_local_content, *media_desc)) {
RTC_HISTOGRAM_BOOLEAN(kSimulcastDisabled, true);
RTCError error =
DisableSimulcastInSender(transceiver->internal()->sender_internal());
if (!error.ok()) {
RTC_LOG(LS_ERROR) << "Failed to remove rejected simulcast.";
return std::move(error);
}
}
}
RTC_DCHECK(transceiver);
if (transceiver->media_type() != media_desc->type()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Transceiver type does not match media description type.");
}
if (media_desc->HasSimulcast()) {
std::vector<SimulcastLayer> layers =
source == cricket::CS_LOCAL
? media_desc->simulcast_description().send_layers().GetAllLayers()
: media_desc->simulcast_description()
.receive_layers()
.GetAllLayers();
RTCError error = UpdateSimulcastLayerStatusInSender(
layers, transceiver->internal()->sender_internal());
if (!error.ok()) {
RTC_LOG(LS_ERROR) << "Failed updating status for simulcast layers.";
return std::move(error);
}
}
// Associate the found or created RtpTransceiver with the m= section by
// setting the value of the RtpTransceiver's mid property to the MID of the m=
// section, and establish a mapping between the transceiver and the index of
// the m= section.
transceiver->internal()->set_mid(content.name);
transceiver->internal()->set_mline_index(mline_index);
return std::move(transceiver);
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetAssociatedTransceiver(const std::string& mid) const {
RTC_DCHECK(IsUnifiedPlan());
for (auto transceiver : transceivers_) {
if (transceiver->mid() == mid) {
return transceiver;
}
}
return nullptr;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetTransceiverByMLineIndex(size_t mline_index) const {
RTC_DCHECK(IsUnifiedPlan());
for (auto transceiver : transceivers_) {
if (transceiver->internal()->mline_index() == mline_index) {
return transceiver;
}
}
return nullptr;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::FindAvailableTransceiverToReceive(
cricket::MediaType media_type) const {
RTC_DCHECK(IsUnifiedPlan());
// From JSEP section 5.10 (Applying a Remote Description):
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers of
// the same type that were added to the PeerConnection by addTrack and are not
// associated with any m= section and are not stopped, find the first such
// RtpTransceiver.
for (auto transceiver : transceivers_) {
if (transceiver->media_type() == media_type &&
transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
!transceiver->stopped()) {
return transceiver;
}
}
return nullptr;
}
const cricket::ContentInfo* PeerConnection::FindMediaSectionForTransceiver(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
const SessionDescriptionInterface* sdesc) const {
RTC_DCHECK(transceiver);
RTC_DCHECK(sdesc);
if (IsUnifiedPlan()) {
if (!transceiver->internal()->mid()) {
// This transceiver is not associated with a media section yet.
return nullptr;
}
return sdesc->description()->GetContentByName(
*transceiver->internal()->mid());
} else {
// Plan B only allows at most one audio and one video section, so use the
// first media section of that type.
return cricket::GetFirstMediaContent(sdesc->description()->contents(),
transceiver->media_type());
}
}
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
RTC_DCHECK_RUN_ON(signaling_thread());
return configuration_;
}
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
RTCError* error) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (IsClosed()) {
RTC_LOG(LS_ERROR) << "SetConfiguration: PeerConnection is closed.";
return SafeSetError(RTCErrorType::INVALID_STATE, error);
}
// According to JSEP, after setLocalDescription, changing the candidate pool
// size is not allowed, and changing the set of ICE servers will not result
// in new candidates being gathered.
if (local_description() && configuration.ice_candidate_pool_size !=
configuration_.ice_candidate_pool_size) {
RTC_LOG(LS_ERROR) << "Can't change candidate pool size after calling "
"SetLocalDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (local_description() &&
configuration.use_media_transport != configuration_.use_media_transport) {
RTC_LOG(LS_ERROR) << "Can't change media_transport after calling "
"SetLocalDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (remote_description() &&
configuration.use_media_transport != configuration_.use_media_transport) {
RTC_LOG(LS_ERROR) << "Can't change media_transport after calling "
"SetRemoteDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (local_description() &&
configuration.use_media_transport_for_data_channels !=
configuration_.use_media_transport_for_data_channels) {
RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels "
"after calling SetLocalDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (remote_description() &&
configuration.use_media_transport_for_data_channels !=
configuration_.use_media_transport_for_data_channels) {
RTC_LOG(LS_ERROR) << "Can't change media_transport_for_data_channels "
"after calling SetRemoteDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (local_description() &&
configuration.crypto_options != configuration_.crypto_options) {
RTC_LOG(LS_ERROR) << "Can't change crypto_options after calling "
"SetLocalDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (local_description() && configuration.use_datagram_transport !=
configuration_.use_datagram_transport) {
RTC_LOG(LS_ERROR) << "Can't change use_datagram_transport "
"after calling SetLocalDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (remote_description() && configuration.use_datagram_transport !=
configuration_.use_datagram_transport) {
RTC_LOG(LS_ERROR) << "Can't change use_datagram_transport "
"after calling SetRemoteDescription.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
if (configuration.use_media_transport_for_data_channels ||
configuration.use_media_transport ||
configuration.use_datagram_transport) {
RTC_CHECK(configuration.bundle_policy == kBundlePolicyMaxBundle)
<< "Media transport requires MaxBundle policy.";
}
// The simplest (and most future-compatible) way to tell if the config was
// modified in an invalid way is to copy each property we do support
// modifying, then use operator==. There are far more properties we don't
// support modifying than those we do, and more could be added.
RTCConfiguration modified_config = configuration_;
modified_config.servers = configuration.servers;
modified_config.type = configuration.type;
modified_config.ice_candidate_pool_size =
configuration.ice_candidate_pool_size;
modified_config.prune_turn_ports = configuration.prune_turn_ports;
modified_config.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.use_media_transport = configuration.use_media_transport;
modified_config.use_media_transport_for_data_channels =
configuration.use_media_transport_for_data_channels;
if (configuration != modified_config) {
RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
// Validate the modified configuration.
RTCError validate_error = ValidateConfiguration(modified_config);
if (!validate_error.ok()) {
return SafeSetError(std::move(validate_error), error);
}
// Note that this isn't possible through chromium, since it's an unsigned
// short in WebIDL.
if (configuration.ice_candidate_pool_size < 0 ||
configuration.ice_candidate_pool_size > static_cast<int>(UINT16_MAX)) {
return SafeSetError(RTCErrorType::INVALID_RANGE, error);
}
// Parse ICE servers before hopping to network thread.
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
RTCErrorType parse_error =
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
if (parse_error != RTCErrorType::NONE) {
return SafeSetError(parse_error, error);
}
// Note if STUN or TURN servers were supplied.
if (!stun_servers.empty()) {
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
}
if (!turn_servers.empty()) {
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
// In theory this shouldn't fail.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
stun_servers, turn_servers, modified_config.type,
modified_config.ice_candidate_pool_size,
modified_config.prune_turn_ports,
modified_config.turn_customizer,
modified_config.stun_candidate_keepalive_interval,
static_cast<bool>(local_description())))) {
RTC_LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
}
// 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 (modified_config.servers != configuration_.servers ||
modified_config.type != configuration_.type ||
modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
transport_controller_->SetNeedsIceRestartFlag();
}
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
transport_controller_->SetMediaTransportSettings(
modified_config.use_media_transport,
modified_config.use_media_transport_for_data_channels,
modified_config.use_datagram_transport);
if (configuration_.active_reset_srtp_params !=
modified_config.active_reset_srtp_params) {
transport_controller_->SetActiveResetSrtpParams(
modified_config.active_reset_srtp_params);
}
configuration_ = modified_config;
use_media_transport_ = configuration.use_media_transport;
return SafeSetError(RTCErrorType::NONE, error);
}
bool PeerConnection::AddIceCandidate(
const IceCandidateInterface* ice_candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
if (IsClosed()) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: PeerConnection is closed.";
NoteAddIceCandidateResult(kAddIceCandidateFailClosed);
return false;
}
if (!remote_description()) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: ICE candidates can't be added "
"without any remote session description.";
NoteAddIceCandidateResult(kAddIceCandidateFailNoRemoteDescription);
return false;
}
if (!ice_candidate) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate is null.";
NoteAddIceCandidateResult(kAddIceCandidateFailNullCandidate);
return false;
}
bool valid = false;
bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
if (!valid) {
NoteAddIceCandidateResult(kAddIceCandidateFailNotValid);
return false;
}
// Add this candidate to the remote session description.
if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate cannot be used.";
NoteAddIceCandidateResult(kAddIceCandidateFailInAddition);
return false;
}
if (ready) {
bool result = UseCandidate(ice_candidate);
if (result) {
NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
if (ice_candidate->candidate().address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED);
}
if (ice_candidate->candidate().address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED);
}
NoteAddIceCandidateResult(kAddIceCandidateSuccess);
} else {
NoteAddIceCandidateResult(kAddIceCandidateFailNotUsable);
}
return result;
} else {
RTC_LOG(LS_INFO) << "AddIceCandidate: Not ready to use candidate.";
NoteAddIceCandidateResult(kAddIceCandidateFailNotReady);
return true;
}
}
bool PeerConnection::RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
RTC_DCHECK_RUN_ON(signaling_thread());
if (IsClosed()) {
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: PeerConnection is closed.";
return false;
}
if (!remote_description()) {
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: ICE candidates can't be removed "
"without any remote session description.";
return false;
}
if (candidates.empty()) {
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: candidates are empty.";
return false;
}
size_t number_removed =
mutable_remote_description()->RemoveCandidates(candidates);
if (number_removed != candidates.size()) {
RTC_LOG(LS_ERROR)
<< "RemoveIceCandidates: Failed to remove candidates. Requested "
<< candidates.size() << " but only " << number_removed
<< " are removed.";
}
// Remove the candidates from the transport controller.
RTCError error = transport_controller_->RemoveRemoteCandidates(candidates);
if (!error.ok()) {
RTC_LOG(LS_ERROR)
<< "RemoveIceCandidates: Error when removing remote candidates: "
<< error.message();
}
return true;
}
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<RTCError>(
RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
}
RTC_DCHECK_RUN_ON(worker_thread());
const bool has_min = bitrate.min_bitrate_bps.has_value();
const bool has_start = bitrate.start_bitrate_bps.has_value();
const bool has_max = bitrate.max_bitrate_bps.has_value();
if (has_min && *bitrate.min_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"min_bitrate_bps <= 0");
}
if (has_start) {
if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"start_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.start_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"curent_bitrate_bps < 0");
}
}
if (has_max) {
if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < start_bitrate_bps");
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < min_bitrate_bps");
} else if (*bitrate.max_bitrate_bps < 0) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max_bitrate_bps < 0");
}
}
RTC_DCHECK(call_.get());
call_->SetClientBitratePreferences(bitrate);
return RTCError::OK();
}
void PeerConnection::SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) {
if (!worker_thread()->IsCurrent()) {
// TODO(kwiberg): Use a lambda instead when C++14 makes it possible to
// move-capture values in lambdas.
struct Task {
PeerConnection* const pc;
std::unique_ptr<rtc::BitrateAllocationStrategy> strategy;
void operator()() {
RTC_DCHECK_RUN_ON(pc->worker_thread());
pc->call_->SetBitrateAllocationStrategy(std::move(strategy));
}
};
worker_thread()->Invoke<void>(
RTC_FROM_HERE, Task{this, std::move(bitrate_allocation_strategy)});
return;
}
RTC_DCHECK_RUN_ON(worker_thread());
RTC_DCHECK(call_.get());
call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
}
void PeerConnection::SetAudioPlayout(bool playout) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout));
return;
}
auto audio_state =
factory_->channel_manager()->media_engine()->voice().GetAudioState();
audio_state->SetPlayout(playout);
}
void PeerConnection::SetAudioRecording(bool recording) {
if (!worker_thread()->IsCurrent()) {
worker_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetAudioRecording, this, recording));
return;
}
auto audio_state =
factory_->channel_manager()->media_engine()->voice().GetAudioState();
audio_state->SetRecording(recording);
}
std::unique_ptr<rtc::SSLCertificate>
PeerConnection::GetRemoteAudioSSLCertificate() {
std::unique_ptr<rtc::SSLCertChain> chain = GetRemoteAudioSSLCertChain();
if (!chain || !chain->GetSize()) {
return nullptr;
}
return chain->Get(0).Clone();
}
std::unique_ptr<rtc::SSLCertChain>
PeerConnection::GetRemoteAudioSSLCertChain() {
RTC_DCHECK_RUN_ON(signaling_thread());
auto audio_transceiver = GetFirstAudioTransceiver();
if (!audio_transceiver || !audio_transceiver->internal()->channel()) {
return nullptr;
}
return transport_controller_->GetRemoteSSLCertChain(
audio_transceiver->internal()->channel()->transport_name());
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetFirstAudioTransceiver() const {
for (auto transceiver : transceivers_) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
return transceiver;
}
}
return nullptr;
}
bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) {
// TODO(eladalon): It would be better to not allow negative values into PC.
const size_t max_size = (max_size_bytes < 0)
? RtcEventLog::kUnlimitedOutput
: rtc::saturated_cast<size_t>(max_size_bytes);
int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
if (field_trial::IsEnabled("WebRTC-RtcEventLogNewFormat")) {
output_period_ms = 5000;
}
return StartRtcEventLog(
absl::make_unique<RtcEventLogOutputFile>(file, max_size),
output_period_ms);
}
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
// TODO(eladalon): In C++14, this can be done with a lambda.
struct Functor {
bool operator()() {
return pc->StartRtcEventLog_w(std::move(output), output_period_ms);
}
PeerConnection* const pc;
std::unique_ptr<RtcEventLogOutput> output;
const int64_t output_period_ms;
};
return worker_thread()->Invoke<bool>(
RTC_FROM_HERE, Functor{this, std::move(output), output_period_ms});
}
bool PeerConnection::StartRtcEventLog(
std::unique_ptr<RtcEventLogOutput> output) {
return StartRtcEventLog(std::move(output),
webrtc::RtcEventLog::kImmediateOutput);
}
void PeerConnection::StopRtcEventLog() {
worker_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
}
rtc::scoped_refptr<DtlsTransportInterface>
PeerConnection::LookupDtlsTransportByMid(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
}
rtc::scoped_refptr<DtlsTransport>
PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_->LookupDtlsTransportByMid(mid);
}
rtc::scoped_refptr<SctpTransportInterface> PeerConnection::GetSctpTransport()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return sctp_transport_;
}
const SessionDescriptionInterface* PeerConnection::local_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_local_description_ ? pending_local_description_.get()
: current_local_description_.get();
}
const SessionDescriptionInterface* PeerConnection::remote_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_remote_description_ ? pending_remote_description_.get()
: current_remote_description_.get();
}
const SessionDescriptionInterface* PeerConnection::current_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return current_local_description_.get();
}
const SessionDescriptionInterface* PeerConnection::current_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return current_remote_description_.get();
}
const SessionDescriptionInterface* PeerConnection::pending_local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_local_description_.get();
}
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_remote_description_.get();
}
void PeerConnection::Close() {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "PeerConnection::Close");
// Update stats here so that we have the most recent stats for tracks and
// streams before the channels are closed.
stats_->UpdateStats(kStatsOutputLevelStandard);
ChangeSignalingState(PeerConnectionInterface::kClosed);
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
for (const auto& transceiver : transceivers_) {
transceiver->Stop();
}
// 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.
DestroyAllChannels();
// The event log is used in the transport controller, which must be outlived
// by the former. CreateOffer by the peer connection is implemented
// asynchronously and if the peer connection is closed without resetting the
// WebRTC session description factory, the session description factory would
// call the transport controller.
webrtc_session_desc_factory_.reset();
transport_controller_.reset();
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
port_allocator_.get()));
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(worker_thread());
call_.reset();
// The event log must outlive call (and any other object that uses it).
event_log_.reset();
});
ReportUsagePattern();
// The .h file says that observer can be discarded after close() returns.
// Make sure this is true.
observer_ = nullptr;
}
void PeerConnection::OnMessage(rtc::Message* msg) {
RTC_DCHECK_RUN_ON(signaling_thread());
switch (msg->message_id) {
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnSuccess();
delete param;
break;
}
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
SetSessionDescriptionMsg* param =
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(std::move(param->error));
delete param;
break;
}
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
CreateSessionDescriptionMsg* param =
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
param->observer->OnFailure(std::move(param->error));
delete param;
break;
}
case MSG_GETSTATS: {
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
StatsReports reports;
stats_->GetStats(param->track, &reports);
param->observer->OnComplete(reports);
delete param;
break;
}
case MSG_FREE_DATACHANNELS: {
sctp_data_channels_to_free_.clear();
break;
}
case MSG_REPORT_USAGE_PATTERN: {
ReportUsagePattern();
break;
}
default:
RTC_NOTREACHED() << "Not implemented";
break;
}
}
cricket::VoiceMediaChannel* PeerConnection::voice_media_channel() const {
RTC_DCHECK(!IsUnifiedPlan());
auto* voice_channel = static_cast<cricket::VoiceChannel*>(
GetAudioTransceiver()->internal()->channel());
if (voice_channel) {
return voice_channel->media_channel();
} else {
return nullptr;
}
}
cricket::VideoMediaChannel* PeerConnection::video_media_channel() const {
RTC_DCHECK(!IsUnifiedPlan());
auto* video_channel = static_cast<cricket::VideoChannel*>(
GetVideoTransceiver()->internal()->channel());
if (video_channel) {
return video_channel->media_channel();
} else {
return nullptr;
}
}
void PeerConnection::CreateAudioReceiver(
MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* audio_receiver = new AudioRtpReceiver(
worker_thread(), remote_sender_info.sender_id, streams);
audio_receiver->SetMediaChannel(voice_media_channel());
audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), audio_receiver);
GetAudioTransceiver()->internal()->AddReceiver(receiver);
Observer()->OnAddTrack(receiver, streams);
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
}
void PeerConnection::CreateVideoReceiver(
MediaStreamInterface* stream,
const RtpSenderInfo& remote_sender_info) {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
// the constructor taking stream IDs instead.
auto* video_receiver = new VideoRtpReceiver(
worker_thread(), remote_sender_info.sender_id, streams);
video_receiver->SetMediaChannel(video_media_channel());
video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
signaling_thread(), video_receiver);
GetVideoTransceiver()->internal()->AddReceiver(receiver);
Observer()->OnAddTrack(receiver, streams);
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
}
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
// description.
rtc::scoped_refptr<RtpReceiverInterface> PeerConnection::RemoveAndStopReceiver(
const RtpSenderInfo& remote_sender_info) {
auto receiver = FindReceiverById(remote_sender_info.sender_id);
if (!receiver) {
RTC_LOG(LS_WARNING) << "RtpReceiver for track with id "
<< remote_sender_info.sender_id << " doesn't exist.";
return nullptr;
}
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
GetAudioTransceiver()->internal()->RemoveReceiver(receiver);
} else {
GetVideoTransceiver()->internal()->RemoveReceiver(receiver);
}
return receiver;
}
void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
RTC_DCHECK(track);
RTC_DCHECK(stream);
auto sender = FindSenderForTrack(track);
if (sender) {
// We already have a sender for this track, so just change the stream_id
// so that it's correct in the next call to CreateOffer.
sender->internal()->set_stream_ids({stream->id()});
return;
}
// Normal case; we've never seen this track before.
auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(), track,
{stream->id()}, {});
new_sender->internal()->SetMediaChannel(voice_media_channel());
GetAudioTransceiver()->internal()->AddSender(new_sender);
// If the sender has already been configured in SDP, we call SetSsrc,
// which will connect the sender to the underlying transport. This can
// occur if a local session description that contains the ID of the sender
// is set before AddStream is called. It can also occur if the local
// session description is not changed and RemoveStream is called, and
// later AddStream is called again with the same stream.
const RtpSenderInfo* sender_info =
FindSenderInfo(local_audio_sender_infos_, stream->id(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
}
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
// indefinitely, when we have unified plan SDP.
void PeerConnection::RemoveAudioTrack(AudioTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
auto sender = FindSenderForTrack(track);
if (!sender) {
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
return;
}
GetAudioTransceiver()->internal()->RemoveSender(sender);
}
void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
RTC_DCHECK(track);
RTC_DCHECK(stream);
auto sender = FindSenderForTrack(track);
if (sender) {
// We already have a sender for this track, so just change the stream_id
// so that it's correct in the next call to CreateOffer.
sender->internal()->set_stream_ids({stream->id()});
return;
}
// Normal case; we've never seen this track before.
auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(), track,
{stream->id()}, {});
new_sender->internal()->SetMediaChannel(video_media_channel());
GetVideoTransceiver()->internal()->AddSender(new_sender);
const RtpSenderInfo* sender_info =
FindSenderInfo(local_video_sender_infos_, stream->id(), track->id());
if (sender_info) {
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
}
}
void PeerConnection::RemoveVideoTrack(VideoTrackInterface* track,
MediaStreamInterface* stream) {
RTC_DCHECK(!IsClosed());
auto sender = FindSenderForTrack(track);
if (!sender) {
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
<< " doesn't exist.";
return;
}
GetVideoTransceiver()->internal()->RemoveSender(sender);
}
void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
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;
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);
}
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;
}
NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
if (candidate->candidate().type() == LOCAL_PORT_TYPE &&
candidate->candidate().address().IsPrivateIP()) {
NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
}
if (candidate->candidate().type() == LOCAL_PORT_TYPE &&
candidate->candidate().address().IsUnresolvedIP()) {
NoteUsageEvent(UsageEvent::MDNS_CANDIDATE_COLLECTED);
}
Observer()->OnIceCandidate(candidate.get());
}
void PeerConnection::OnIceCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
if (IsClosed()) {
return;
}
Observer()->OnIceCandidatesRemoved(candidates);
}
void PeerConnection::ChangeSignalingState(
PeerConnectionInterface::SignalingState signaling_state) {
if (signaling_state_ == signaling_state) {
return;
}
RTC_LOG(LS_INFO) << "Session: " << session_id() << " Old state: "
<< GetSignalingStateString(signaling_state_)
<< " New state: "
<< GetSignalingStateString(signaling_state);
signaling_state_ = signaling_state;
if (signaling_state == kClosed) {
ice_connection_state_ = kIceConnectionClosed;
Observer()->OnIceConnectionChange(ice_connection_state_);
standardized_ice_connection_state_ =
PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
Observer()->OnConnectionChange(connection_state_);
if (ice_gathering_state_ != kIceGatheringComplete) {
ice_gathering_state_ = kIceGatheringComplete;
Observer()->OnIceGatheringChange(ice_gathering_state_);
}
}
Observer()->OnSignalingChange(signaling_state_);
}
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
AddAudioTrack(track, stream);
UpdateNegotiationNeeded();
}
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveAudioTrack(track, stream);
UpdateNegotiationNeeded();
}
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
AddVideoTrack(track, stream);
UpdateNegotiationNeeded();
}
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (IsClosed()) {
return;
}
RemoveVideoTrack(track, stream);
UpdateNegotiationNeeded();
}
void PeerConnection::PostSetSessionDescriptionSuccess(
SetSessionDescriptionObserver* observer) {
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
}
void PeerConnection::PostSetSessionDescriptionFailure(
SetSessionDescriptionObserver* observer,
RTCError&& error) {
RTC_DCHECK(!error.ok());
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
msg->error = std::move(error);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::PostCreateSessionDescriptionFailure(
CreateSessionDescriptionObserver* observer,
RTCError error) {
RTC_DCHECK(!error.ok());
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
msg->error = std::move(error);
signaling_thread()->Post(RTC_FROM_HERE, this,
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
}
void PeerConnection::GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
if (IsUnifiedPlan()) {
GetOptionsForUnifiedPlanOffer(offer_answer_options, session_options);
} else {
GetOptionsForPlanBOffer(offer_answer_options, session_options);
}
// Intentionally unset the data channel type for RTP data channel with the
// second condition. Otherwise the RTP data channels would be successfully
// negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
// when building with chromium. We want to leave RTP data channels broken, so
// people won't try to use them.
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
session_options->data_channel_type = data_channel_type();
}
// Apply ICE restart flag and renomination flag.
for (auto& options : session_options->media_description_options) {
options.transport_options.ice_restart = offer_answer_options.ice_restart;
options.transport_options.enable_ice_renomination =
configuration_.enable_ice_renomination;
}
session_options->rtcp_cname = rtcp_cname_;
session_options->crypto_options = GetCryptoOptions();
session_options->pooled_ice_credentials =
network_thread()->Invoke<std::vector<cricket::IceParameters>>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
port_allocator_.get()));
session_options->offer_extmap_allow_mixed =
configuration_.offer_extmap_allow_mixed;
if (configuration_.enable_dtls_srtp &&
!configuration_.enable_dtls_srtp.value()) {
session_options->media_transport_settings =
transport_controller_->GenerateOrGetLastMediaTransportOffer();
}
// Allow fallback for using obsolete SCTP syntax.
// Note that the default in |session_options| is true, while
// the default in |options| is false.
session_options->use_obsolete_sctp_sdp =
offer_answer_options.use_obsolete_sctp_sdp;
}
void PeerConnection::GetOptionsForPlanBOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
// Figure out transceiver directional preferences.
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
// By default, generate sendrecv/recvonly m= sections.
bool recv_audio = true;
bool recv_video = true;
// By default, only offer a new m= section if we have media to send with it.
bool offer_new_audio_description = send_audio;
bool offer_new_video_description = send_video;
bool offer_new_data_description = HasDataChannels();
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
offer_new_audio_description =
offer_new_audio_description ||
(offer_answer_options.offer_to_receive_audio > 0);
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
offer_new_video_description =
offer_new_video_description ||
(offer_answer_options.offer_to_receive_video > 0);
}
absl::optional<size_t> audio_index;
absl::optional<size_t> video_index;
absl::optional<size_t> data_index;
// If a current description exists, generate m= sections in the same order,
// using the first audio/video/data section that appears and rejecting
// extraneous ones.
if (local_description()) {
GenerateMediaDescriptionOptions(
local_description(),
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
&audio_index, &video_index, &data_index, session_options);
}
// Add audio/video/data m= sections to the end if needed.
if (!audio_index && offer_new_audio_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
false));
audio_index = session_options->media_description_options.size() - 1;
}
if (!video_index && offer_new_video_description) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
false));
video_index = session_options->media_description_options.size() - 1;
}
if (!data_index && offer_new_data_description) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(cricket::CN_DATA));
data_index = session_options->media_description_options.size() - 1;
}
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
AddPlanBRtpSenderOptions(GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
}
static cricket::MediaDescriptionOptions
GetMediaDescriptionOptionsForTransceiver(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver,
const std::string& mid) {
cricket::MediaDescriptionOptions media_description_options(
transceiver->media_type(), mid, transceiver->direction(),
transceiver->stopped());
media_description_options.codec_preferences =
transceiver->codec_preferences();
// This behavior is specified in JSEP. The gist is that:
// 1. The MSID is included if the RtpTransceiver's direction is sendonly or
// sendrecv.
// 2. If the MSID is included, then it must be included in any subsequent
// offer/answer exactly the same until the RtpTransceiver is stopped.
if (transceiver->stopped() ||
(!RtpTransceiverDirectionHasSend(transceiver->direction()) &&
!transceiver->internal()->has_ever_been_used_to_send())) {
return media_description_options;
}
cricket::SenderOptions sender_options;
sender_options.track_id = transceiver->sender()->id();
sender_options.stream_ids = transceiver->sender()->stream_ids();
// The following sets up RIDs and Simulcast.
// RIDs are included if Simulcast is requested or if any RID was specified.
RtpParameters send_parameters =
transceiver->internal()->sender_internal()->GetParametersInternal();
bool has_rids = std::any_of(send_parameters.encodings.begin(),
send_parameters.encodings.end(),
[](const RtpEncodingParameters& encoding) {
return !encoding.rid.empty();
});
std::vector<RidDescription> send_rids;
SimulcastLayerList send_layers;
for (const RtpEncodingParameters& encoding : send_parameters.encodings) {
if (encoding.rid.empty()) {
continue;
}
send_rids.push_back(RidDescription(encoding.rid, RidDirection::kSend));
send_layers.AddLayer(SimulcastLayer(encoding.rid, !encoding.active));
}
if (has_rids) {
sender_options.rids = send_rids;
}
sender_options.simulcast_layers = send_layers;
// When RIDs are configured, we must set num_sim_layers to 0 to.
// Otherwise, num_sim_layers must be 1 because either there is no
// simulcast, or simulcast is acheived by munging the SDP.
sender_options.num_sim_layers = has_rids ? 0 : 1;
media_description_options.sender_options.push_back(sender_options);
return media_description_options;
}
// Returns the ContentInfo at mline index |i|, or null if none exists.
static const ContentInfo* GetContentByIndex(
const SessionDescriptionInterface* sdesc,
size_t i) {
if (!sdesc) {
return nullptr;
}
const ContentInfos& contents = sdesc->description()->contents();
return (i < contents.size() ? &contents[i] : nullptr);
}
void PeerConnection::GetOptionsForUnifiedPlanOffer(
const RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
// Rules for generating an offer are dictated by JSEP sections 5.2.1 (Initial
// Offers) and 5.2.2 (Subsequent Offers).
RTC_DCHECK_EQ(session_options->media_description_options.size(), 0);
const ContentInfos& local_contents =
(local_description() ? local_description()->description()->contents()
: ContentInfos());
const ContentInfos& remote_contents =
(remote_description() ? remote_description()->description()->contents()
: ContentInfos());
// The mline indices that can be recycled. New transceivers should reuse these
// slots first.
std::queue<size_t> recycleable_mline_indices;
// First, go through each media section that exists in either the local or
// remote description and generate a media section in this offer for the
// associated transceiver. If a media section can be recycled, generate a
// default, rejected media section here that can be later overwritten.
for (size_t i = 0;
i < std::max(local_contents.size(), remote_contents.size()); ++i) {
// Either |local_content| or |remote_content| is non-null.
const ContentInfo* local_content =
(i < local_contents.size() ? &local_contents[i] : nullptr);
const ContentInfo* current_local_content =
GetContentByIndex(current_local_description(), i);
const ContentInfo* remote_content =
(i < remote_contents.size() ? &remote_contents[i] : nullptr);
const ContentInfo* current_remote_content =
GetContentByIndex(current_remote_description(), i);
bool had_been_rejected =
(current_local_content && current_local_content->rejected) ||
(current_remote_content && current_remote_content->rejected);
const std::string& mid =
(local_content ? local_content->name : remote_content->name);
cricket::MediaType media_type =
(local_content ? local_content->media_description()->type()
: remote_content->media_description()->type());
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO) {
auto transceiver = GetAssociatedTransceiver(mid);
RTC_CHECK(transceiver);
// A media section is considered eligible for recycling if it is marked as
// rejected in either the current local or current remote description.
if (had_been_rejected && transceiver->stopped()) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(transceiver->media_type(), mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
recycleable_mline_indices.push(i);
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(transceiver, mid));
// CreateOffer shouldn't really cause any state changes in
// PeerConnection, but we need a way to match new transceivers to new
// media sections in SetLocalDescription and JSEP specifies this is done
// by recording the index of the media section generated for the
// transceiver in the offer.
transceiver->internal()->set_mline_index(i);
}
} else {
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
RTC_CHECK(GetDataMid());
if (had_been_rejected || mid != *GetDataMid()) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(mid));
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(mid));
}
}
}
// Next, look for transceivers that are newly added (that is, are not stopped
// and not associated). Reuse media sections marked as recyclable first,
// otherwise append to the end of the offer. New media sections should be
// added in the order they were added to the PeerConnection.
for (const auto& transceiver : transceivers_) {
if (transceiver->mid() || transceiver->stopped()) {
continue;
}
size_t mline_index;
if (!recycleable_mline_indices.empty()) {
mline_index = recycleable_mline_indices.front();
recycleable_mline_indices.pop();
session_options->media_description_options[mline_index] =
GetMediaDescriptionOptionsForTransceiver(transceiver,
mid_generator_());
} else {
mline_index = session_options->media_description_options.size();
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(transceiver,
mid_generator_()));
}
// See comment above for why CreateOffer changes the transceiver's state.
transceiver->internal()->set_mline_index(mline_index);
}
// Lastly, add a m-section if we have local data channels and an m section
// does not already exist.
if (!GetDataMid() && HasDataChannels()) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(mid_generator_()));
}
}
void PeerConnection::GetOptionsForAnswer(
const RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
if (IsUnifiedPlan()) {
GetOptionsForUnifiedPlanAnswer(offer_answer_options, session_options);
} else {
GetOptionsForPlanBAnswer(offer_answer_options, session_options);
}
// Intentionally unset the data channel type for RTP data channel. Otherwise
// the RTP data channels would be successfully negotiated by default and the
// unit tests in WebRtcDataBrowserTest will fail when building with chromium.
// We want to leave RTP data channels broken, so people won't try to use them.
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
session_options->data_channel_type = data_channel_type();
}
// Apply ICE renomination flag.
for (auto& options : session_options->media_description_options) {
options.transport_options.enable_ice_renomination =
configuration_.enable_ice_renomination;
}
session_options->rtcp_cname = rtcp_cname_;
session_options->crypto_options = GetCryptoOptions();
session_options->pooled_ice_credentials =
network_thread()->Invoke<std::vector<cricket::IceParameters>>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
port_allocator_.get()));
}
void PeerConnection::GetOptionsForPlanBAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
// Figure out transceiver directional preferences.
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
// By default, generate sendrecv/recvonly m= sections. The direction is also
// restricted by the direction in the offer.
bool recv_audio = true;
bool recv_video = true;
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
}
absl::optional<size_t> audio_index;
absl::optional<size_t> video_index;
absl::optional<size_t> data_index;
// Generate m= sections that match those in the offer.
// Note that mediasession.cc will handle intersection our preferred
// direction with the offered direction.
GenerateMediaDescriptionOptions(
remote_description(),
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), &audio_index,
&video_index, &data_index, session_options);
cricket::MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
cricket::MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
AddPlanBRtpSenderOptions(GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
}
void PeerConnection::GetOptionsForUnifiedPlanAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
cricket::MediaSessionOptions* session_options) {
// Rules for generating an answer are dictated by JSEP sections 5.3.1 (Initial
// Answers) and 5.3.2 (Subsequent Answers).
RTC_DCHECK(remote_description());
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
for (const ContentInfo& content :
remote_description()->description()->contents()) {
cricket::MediaType media_type = content.media_description()->type();
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO) {
auto transceiver = GetAssociatedTransceiver(content.name);
RTC_CHECK(transceiver);
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(transceiver, content.name));
} else {
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
// Reject all data sections if data channels are disabled.
// Reject a data section if it has already been rejected.
// Reject all data sections except for the first one.
if (data_channel_type_ == cricket::DCT_NONE || content.rejected ||
content.name != *GetDataMid()) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(content.name));
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(content.name));
}
}
}
}
void PeerConnection::GenerateMediaDescriptionOptions(
const SessionDescriptionInterface* session_desc,
RtpTransceiverDirection audio_direction,
RtpTransceiverDirection video_direction,
absl::optional<size_t>* audio_index,
absl::optional<size_t>* video_index,
absl::optional<size_t>* data_index,
cricket::MediaSessionOptions* session_options) {
for (const cricket::ContentInfo& content :
session_desc->description()->contents()) {
if (IsAudioContent(&content)) {
// If we already have an audio m= section, reject this extra one.
if (*audio_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_AUDIO, content.name,
RtpTransceiverDirection::kInactive, /*stopped=*/true));
} else {
bool stopped = (audio_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO,
content.name, audio_direction,
stopped));
*audio_index = session_options->media_description_options.size() - 1;
}
} else if (IsVideoContent(&content)) {
// If we already have an video m= section, reject this extra one.
if (*video_index) {
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(
cricket::MEDIA_TYPE_VIDEO, content.name,
RtpTransceiverDirection::kInactive, /*stopped=*/true));
} else {
bool stopped = (video_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO,
content.name, video_direction,
stopped));
*video_index = session_options->media_description_options.size() - 1;
}
} else {
RTC_DCHECK(IsDataContent(&content));
// If we already have an data m= section, reject this extra one.
if (*data_index) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(content.name));
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(content.name));
*data_index = session_options->media_description_options.size() - 1;
}
}
}
}
cricket::MediaDescriptionOptions
PeerConnection::GetMediaDescriptionOptionsForActiveData(
const std::string& mid) const {
// Direction for data sections is meaningless, but legacy endpoints might
// expect sendrecv.
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
RtpTransceiverDirection::kSendRecv,
/*stopped=*/false);
AddRtpDataChannelOptions(rtp_data_channels_, &options);
return options;
}
cricket::MediaDescriptionOptions
PeerConnection::GetMediaDescriptionOptionsForRejectedData(
const std::string& mid) const {
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true);
AddRtpDataChannelOptions(rtp_data_channels_, &options);
return options;
}
absl::optional<std::string> PeerConnection::GetDataMid() const {
switch (data_channel_type_) {
case cricket::DCT_RTP:
if (!rtp_data_channel_) {
return absl::nullopt;
}
return rtp_data_channel_->content_name();
case cricket::DCT_SCTP:
return sctp_mid_;
case cricket::DCT_MEDIA_TRANSPORT:
return media_transport_data_mid_;
default:
return absl::nullopt;
}
}
void PeerConnection::RemoveSenders(cricket::MediaType media_type) {
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
media_type, nullptr);
}
void PeerConnection::UpdateRemoteSendersList(
const cricket::StreamParamsVec& streams,
bool default_sender_needed,
cricket::MediaType media_type,
StreamCollection* new_streams) {
RTC_DCHECK(!IsUnifiedPlan());
std::vector<RtpSenderInfo>* current_senders =
GetRemoteSenderInfos(media_type);
// Find removed senders. I.e., senders where the sender id or ssrc don't match
// the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.first_ssrc);
std::string params_stream_id;
if (params) {
params_stream_id =
(!params->first_stream_id().empty() ? params->first_stream_id()
: kDefaultStreamId);
}
bool sender_exists = params && params->id == info.sender_id &&
params_stream_id == info.stream_id;
// If this is a default track, and we still need it, don't remove it.
if ((info.stream_id == kDefaultStreamId && default_sender_needed) ||
sender_exists) {
++sender_it;
} else {
OnRemoteSenderRemoved(info, media_type);
sender_it = current_senders->erase(sender_it);
}
}
// Find new and active senders.
for (const cricket::StreamParams& params : streams) {
if (!params.has_ssrcs()) {
// The remote endpoint has streams, but didn't signal ssrcs. For an active
// sender, this means it is coming from a Unified Plan endpoint,so we just
// create a default.
default_sender_needed = true;
break;
}
// |params.id| is the sender id and the stream id uses the first of
// |params.stream_ids|. The remote description could come from a Unified
// Plan endpoint, with multiple or no stream_ids() signaled. Since this is
// not supported in Plan B, we just take the first here and create the
// default stream ID if none is specified.
const std::string& stream_id =
(!params.first_stream_id().empty() ? params.first_stream_id()
: kDefaultStreamId);
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
rtc::scoped_refptr<MediaStreamInterface> stream =
remote_streams_->find(stream_id);
if (!stream) {
// This is a new MediaStream. Create a new remote MediaStream.
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
MediaStream::Create(stream_id));
remote_streams_->AddStream(stream);
new_streams->AddStream(stream);
}
const RtpSenderInfo* sender_info =
FindSenderInfo(*current_senders, stream_id, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
OnRemoteSenderAdded(current_senders->back(), media_type);
}
}
// Add default sender if necessary.
if (default_sender_needed) {
rtc::scoped_refptr<MediaStreamInterface> default_stream =
remote_streams_->find(kDefaultStreamId);
if (!default_stream) {
// Create the new default MediaStream.
default_stream = MediaStreamProxy::Create(
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamId));
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
? kDefaultAudioSenderId
: kDefaultVideoSenderId;
const RtpSenderInfo* default_sender_info =
FindSenderInfo(*current_senders, kDefaultStreamId, default_sender_id);
if (!default_sender_info) {
current_senders->push_back(
RtpSenderInfo(kDefaultStreamId, default_sender_id, /*ssrc=*/0));
OnRemoteSenderAdded(current_senders->back(), media_type);
}
}
}
void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
RTC_LOG(LS_INFO) << "Creating " << cricket::MediaTypeToString(media_type)
<< " receiver for track_id=" << sender_info.sender_id
<< " and stream_id=" << sender_info.stream_id;
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
CreateAudioReceiver(stream, sender_info);
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
CreateVideoReceiver(stream, sender_info);
} else {
RTC_NOTREACHED() << "Invalid media type";
}
}
void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
RTC_LOG(LS_INFO) << "Removing " << cricket::MediaTypeToString(media_type)
<< " receiver for track_id=" << sender_info.sender_id
<< " and stream_id=" << sender_info.stream_id;
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
rtc::scoped_refptr<RtpReceiverInterface> receiver;
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
// will be notified which will end the AudioRtpReceiver::track().
receiver = RemoveAndStopReceiver(sender_info);
rtc::scoped_refptr<AudioTrackInterface> audio_track =
stream->FindAudioTrack(sender_info.sender_id);
if (audio_track) {
stream->RemoveTrack(audio_track);
}
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
// Stopping or destroying a VideoRtpReceiver will end the
// VideoRtpReceiver::track().
receiver = RemoveAndStopReceiver(sender_info);
rtc::scoped_refptr<VideoTrackInterface> video_track =
stream->FindVideoTrack(sender_info.sender_id);
if (video_track) {
// There's no guarantee the track is still available, e.g. the track may
// have been removed from the stream by an application.
stream->RemoveTrack(video_track);
}
} else {
RTC_NOTREACHED() << "Invalid media type";
}
if (receiver) {
Observer()->OnRemoveTrack(receiver);
}
}
void PeerConnection::UpdateEndedRemoteMediaStreams() {
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
for (size_t i = 0; i < remote_streams_->count(); ++i) {
MediaStreamInterface* stream = remote_streams_->at(i);
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
streams_to_remove.push_back(stream);
}
}
for (auto& stream : streams_to_remove) {
remote_streams_->RemoveStream(stream);
Observer()->OnRemoveStream(std::move(stream));
}
}
void PeerConnection::UpdateLocalSenders(
const std::vector<cricket::StreamParams>& streams,
cricket::MediaType media_type) {
std::vector<RtpSenderInfo>* current_senders = GetLocalSenderInfos(media_type);
// Find removed tracks. I.e., tracks where the track id, stream id or ssrc
// don't match the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const cricket::StreamParams* params =
cricket::GetStreamBySsrc(streams, info.first_ssrc);
if (!params || params->id != info.sender_id ||
params->first_stream_id() != info.stream_id) {
OnLocalSenderRemoved(info, media_type);
sender_it = current_senders->erase(sender_it);
} else {
++sender_it;
}
}
// Find new and active senders.
for (const cricket::StreamParams& params : streams) {
// The sync_label is the MediaStream label and the |stream.id| is the
// sender id.
const std::string& stream_id = params.first_stream_id();
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
const RtpSenderInfo* sender_info =
FindSenderInfo(*current_senders, stream_id, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
OnLocalSenderAdded(current_senders->back(), media_type);
}
}
}
void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
RTC_DCHECK(!IsUnifiedPlan());
auto sender = FindSenderById(sender_info.sender_id);
if (!sender) {
RTC_LOG(LS_WARNING) << "An unknown RtpSender with id "
<< sender_info.sender_id
<< " has been configured in the local description.";
return;
}
if (sender->media_type() != media_type) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
" description with an unexpected media type.";
return;
}
sender->internal()->set_stream_ids({sender_info.stream_id});
sender->internal()->SetSsrc(sender_info.first_ssrc);
}
void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
cricket::MediaType media_type) {
auto sender = FindSenderById(sender_info.sender_id);
if (!sender) {
// This is the normal case. I.e., RemoveStream has been called and the
// SessionDescriptions has been renegotiated.
return;
}
// A sender has been removed from the SessionDescription but it's still
// associated with the PeerConnection. This only occurs if the SDP doesn't
// match with the calls to CreateSender, AddStream and RemoveStream.
if (sender->media_type() != media_type) {
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
" description with an unexpected media type.";
return;
}
sender->internal()->SetSsrc(0);
}
void PeerConnection::UpdateLocalRtpDataChannels(
const cricket::StreamParamsVec& streams) {
std::vector<std::string> existing_channels;
// Find new and active data channels.
for (const cricket::StreamParams& params : streams) {
// |it->sync_label| is actually the data channel label. The reason is that
// we use the same naming of data channels as we do for
// MediaStreams and Tracks.
// For MediaStreams, the sync_label is the MediaStream label and the
// track label is the same as |streamid|.
const std::string& channel_label = params.first_stream_id();
auto data_channel_it = rtp_data_channels_.find(channel_label);
if (data_channel_it == rtp_data_channels_.end()) {
RTC_LOG(LS_ERROR) << "channel label not found";
continue;
}
// Set the SSRC the data channel should use for sending.
data_channel_it->second->SetSendSsrc(params.first_ssrc());
existing_channels.push_back(data_channel_it->first);
}
UpdateClosingRtpDataChannels(existing_channels, true);
}
void PeerConnection::UpdateRemoteRtpDataChannels(
const cricket::StreamParamsVec& streams) {
std::vector<std::string> existing_channels;
// Find new and active data channels.
for (const cricket::StreamParams& params : streams) {
// The data channel label is either the mslabel or the SSRC if the mslabel
// does not exist. Ex a=ssrc:444330170 mslabel:test1.
std::string label = params.first_stream_id().empty()
? rtc::ToString(params.first_ssrc())
: params.first_stream_id();
auto data_channel_it = rtp_data_channels_.find(label);
if (data_channel_it == rtp_data_channels_.end()) {
// This is a new data channel.
CreateRemoteRtpDataChannel(label, params.first_ssrc());
} else {
data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
}
existing_channels.push_back(label);
}
UpdateClosingRtpDataChannels(existing_channels, false);
}
void PeerConnection::UpdateClosingRtpDataChannels(
const std::vector<std::string>& active_channels,
bool is_local_update) {
auto it = rtp_data_channels_.begin();
while (it != rtp_data_channels_.end()) {
DataChannel* data_channel = it->second;
if (absl::c_linear_search(active_channels, data_channel->label())) {
++it;
continue;
}
if (is_local_update) {
data_channel->SetSendSsrc(0);
} else {
data_channel->RemotePeerRequestClose();
}
if (data_channel->state() == DataChannel::kClosed) {
rtp_data_channels_.erase(it);
it = rtp_data_channels_.begin();
} else {
++it;
}
}
}
void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
uint32_t remote_ssrc) {
rtc::scoped_refptr<DataChannel> channel(
InternalCreateDataChannel(label, nullptr));
if (!channel.get()) {
RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
"CreateDataChannel failed.";
return;
}
channel->SetReceiveSsrc(remote_ssrc);
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
DataChannelProxy::Create(signaling_thread(), channel);
Observer()->OnDataChannel(std::move(proxy_channel));
}
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
const std::string& label,
const InternalDataChannelInit* config) {
if (IsClosed()) {
return nullptr;
}
if (data_channel_type() == cricket::DCT_NONE) {
RTC_LOG(LS_ERROR)
<< "InternalCreateDataChannel: Data is not supported in this call.";
return nullptr;
}
InternalDataChannelInit new_config =
config ? (*config) : InternalDataChannelInit();
if (DataChannel::IsSctpLike(data_channel_type_)) {
if (new_config.id < 0) {
rtc::SSLRole role;
if ((GetSctpSslRole(&role)) &&
!sid_allocator_.AllocateSid(role, &new_config.id)) {
RTC_LOG(LS_ERROR)
<< "No id can be allocated for the SCTP data channel.";
return nullptr;
}
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
RTC_LOG(LS_ERROR) << "Failed to create a SCTP data channel "
"because the id is already in use or out of range.";
return nullptr;
}
}
rtc::scoped_refptr<DataChannel> channel(
DataChannel::Create(this, data_channel_type(), label, new_config));
if (!channel) {
sid_allocator_.ReleaseSid(new_config.id);
return nullptr;
}
if (channel->data_channel_type() == cricket::DCT_RTP) {
if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
RTC_LOG(LS_ERROR) << "DataChannel with label " << channel->label()
<< " already exists.";
return nullptr;
}
rtp_data_channels_[channel->label()] = channel;
} else {
RTC_DCHECK(DataChannel::IsSctpLike(data_channel_type_));
sctp_data_channels_.push_back(channel);
channel->SignalClosed.connect(this,
&PeerConnection::OnSctpDataChannelClosed);
}
SignalDataChannelCreated_(channel.get());
return channel;
}
bool PeerConnection::HasDataChannels() const {
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
}
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
std::vector<rtc::scoped_refptr<DataChannel>> channels_to_close;
for (const auto& channel : sctp_data_channels_) {
if (channel->id() < 0) {
int sid;
if (!sid_allocator_.AllocateSid(role, &sid)) {
RTC_LOG(LS_ERROR) << "Failed to allocate SCTP sid, closing channel.";
channels_to_close.push_back(channel);
continue;
}
channel->SetSctpSid(sid);
}
}
// Since closing modifies the list of channels, we have to do the actual
// closing outside the loop.
for (const auto& channel : channels_to_close) {
channel->CloseAbruptly();
}
}
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
RTC_DCHECK(signaling_thread()->IsCurrent());
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
++it) {
if (it->get() == channel) {
if (channel->id() >= 0) {
// After the closing procedure is done, it's safe to use this ID for
// another data channel.
sid_allocator_.ReleaseSid(channel->id());
}
// Since this method is triggered by a signal from the DataChannel,
// we can't free it directly here; we need to free it asynchronously.
sctp_data_channels_to_free_.push_back(*it);
sctp_data_channels_.erase(it);
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
nullptr);
return;
}
}
}
void PeerConnection::OnDataChannelDestroyed() {
// Use a temporary copy of the RTP/SCTP DataChannel list because the
// DataChannel may callback to us and try to modify the list.
std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
temp_rtp_dcs.swap(rtp_data_channels_);
for (const auto& kv : temp_rtp_dcs) {
kv.second->OnTransportChannelDestroyed();
}
std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
temp_sctp_dcs.swap(sctp_data_channels_);
for (const auto& channel : temp_sctp_dcs) {
channel->OnTransportChannelDestroyed();
}
}
void PeerConnection::OnDataChannelOpenMessage(
const std::string& label,
const InternalDataChannelInit& config) {
rtc::scoped_refptr<DataChannel> channel(
InternalCreateDataChannel(label, &config));
if (!channel.get()) {
RTC_LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
return;
}
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
DataChannelProxy::Create(signaling_thread(), channel);
Observer()->OnDataChannel(std::move(proxy_channel));
NoteUsageEvent(UsageEvent::DATA_ADDED);
}
bool PeerConnection::HandleOpenMessage_s(
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& buffer) {
if (params.type == cricket::DMT_CONTROL && IsOpenMessage(buffer)) {
// Received OPEN message; parse and signal that a new data channel should
// be created.
std::string label;
InternalDataChannelInit config;
config.id = params.ssrc;
if (!ParseDataChannelOpenMessage(buffer, &label, &config)) {
RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for ssrc "
<< params.ssrc;
return true;
}
config.open_handshake_role = InternalDataChannelInit::kAcker;
OnDataChannelOpenMessage(label, config);
return true;
}
return false;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetAudioTransceiver() const {
// This method only works with Plan B SDP, where there is a single
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
return transceiver;
}
}
RTC_NOTREACHED();
return nullptr;
}
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
PeerConnection::GetVideoTransceiver() const {
// This method only works with Plan B SDP, where there is a single
// audio/video transceiver.
RTC_DCHECK(!IsUnifiedPlan());
for (auto transceiver : transceivers_) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
return transceiver;
}
}
RTC_NOTREACHED();
return nullptr;
}
// TODO(bugs.webrtc.org/7600): Remove this when multiple transceivers with
// individual transceiver directions are supported.
bool PeerConnection::HasRtpSender(cricket::MediaType type) const {
switch (type) {
case cricket::MEDIA_TYPE_AUDIO:
return !GetAudioTransceiver()->internal()->senders().empty();
case cricket::MEDIA_TYPE_VIDEO:
return !GetVideoTransceiver()->internal()->senders().empty();
case cricket::MEDIA_TYPE_DATA:
return false;
}
RTC_NOTREACHED();
return false;
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
for (const auto& transceiver : transceivers_) {
for (auto sender : transceiver->internal()->senders()) {
if (sender->track() == track) {
return sender;
}
}
}
return nullptr;
}
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
PeerConnection::FindSenderById(const std::string& sender_id) const {
for (const auto& transceiver : transceivers_) {
for (auto sender : transceiver->internal()->senders()) {
if (sender->id() == sender_id) {
return sender;
}
}
}
return nullptr;
}
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
PeerConnection::FindReceiverById(const std::string& receiver_id) const {
for (const auto& transceiver : transceivers_) {
for (auto receiver : transceiver->internal()->receivers()) {
if (receiver->id() == receiver_id) {
return receiver;
}
}
}
return nullptr;
}
std::vector<PeerConnection::RtpSenderInfo>*
PeerConnection::GetRemoteSenderInfos(cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO)
? &remote_audio_sender_infos_
: &remote_video_sender_infos_;
}
std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos(
cricket::MediaType media_type) {
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
media_type == cricket::MEDIA_TYPE_VIDEO);
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_sender_infos_
: &local_video_sender_infos_;
}
const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo(
const std::vector<PeerConnection::RtpSenderInfo>& infos,
const std::string& stream_id,
const std::string sender_id) const {
for (const RtpSenderInfo& sender_info : infos) {
if (sender_info.stream_id == stream_id &&
sender_info.sender_id == sender_id) {
return &sender_info;
}
}
return nullptr;
}
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
for (const auto& channel : sctp_data_channels_) {
if (channel->id() == sid) {
return channel;
}
}
return nullptr;
}
PeerConnection::InitializePortAllocatorResult
PeerConnection::InitializePortAllocator_n(
const cricket::ServerAddresses& stun_servers,
const std::vector<cricket::RelayServerConfig>& turn_servers,
const RTCConfiguration& configuration) {
RTC_DCHECK_RUN_ON(network_thread());
port_allocator_->Initialize();
// To handle both internal and externally created port allocator, we will
// enable BUNDLE here.
int port_allocator_flags = port_allocator_->flags();
port_allocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
cricket::PORTALLOCATOR_ENABLE_IPV6 |
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
// If the disable-IPv6 flag was specified, we'll not override it
// by experiment.
if (configuration.disable_ipv6) {
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
} else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default")
.find("Disabled") == 0) {
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.prune_turn_ports,
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,
bool prune_turn_ports,
webrtc::TurnCustomizer* turn_customizer,
absl::optional<int> stun_candidate_keepalive_interval,
bool have_local_description) {
port_allocator_->SetCandidateFilter(
ConvertIceTransportTypeToCandidateFilter(type));
// According to JSEP, after setLocalDescription, changing the candidate pool
// size is not allowed, and changing the set of ICE servers will not result
// in new candidates being gathered.
if (have_local_description) {
port_allocator_->FreezeCandidatePool();
}
// Add the custom tls turn servers if they exist.
auto turn_servers_copy = turn_servers;
for (auto& turn_server : turn_servers_copy) {
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
}
// Call this last since it may create pooled allocator sessions using the
// candidate filter set above.
return port_allocator_->SetConfiguration(
stun_servers, std::move(turn_servers_copy), candidate_pool_size,
prune_turn_ports, turn_customizer, stun_candidate_keepalive_interval);
}
cricket::ChannelManager* PeerConnection::channel_manager() const {
return factory_->channel_manager();
}
bool PeerConnection::StartRtcEventLog_w(
std::unique_ptr<RtcEventLogOutput> output,
int64_t output_period_ms) {
RTC_DCHECK_RUN_ON(worker_thread());
if (!event_log_) {
return false;
}
return event_log_->StartLogging(std::move(output), output_period_ms);
}
void PeerConnection::StopRtcEventLog_w() {
RTC_DCHECK_RUN_ON(worker_thread());
if (event_log_) {
event_log_->StopLogging();
}
}
cricket::ChannelInterface* PeerConnection::GetChannel(
const std::string& content_name) {
for (const auto& transceiver : transceivers_) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && channel->content_name() == content_name) {
return channel;
}
}
if (rtp_data_channel() &&
rtp_data_channel()->content_name() == content_name) {
return rtp_data_channel();
}
return nullptr;
}
bool PeerConnection::GetSctpSslRole(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 SCTP transport.";
return false;
}
if (!sctp_transport_ && !media_transport_) {
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
"SSL Role of the SCTP transport.";
return false;
}
absl::optional<rtc::SSLRole> dtls_role;
if (sctp_mid_) {
dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_);
} else if (is_caller_) {
dtls_role = *is_caller_ ? rtc::SSL_SERVER : rtc::SSL_CLIENT;
}
if (dtls_role) {
*role = *dtls_role;
return true;
}
return false;
}
bool PeerConnection::GetSslRole(const std::string& content_name,
rtc::SSLRole* role) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_LOG(LS_INFO)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the session.";
return false;
}
auto dtls_role = transport_controller_->GetDtlsRole(content_name);
if (dtls_role) {
*role = *dtls_role;
return true;
}
return false;
}
void PeerConnection::SetSessionError(SessionError error,
const std::string& error_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (error != session_error_) {
session_error_ = error;
session_error_desc_ = error_desc;
}
}
RTCError PeerConnection::UpdateSessionState(
SdpType type,
cricket::ContentSource source,
const cricket::SessionDescription* description) {
RTC_DCHECK_RUN_ON(signaling_thread());
// If there's already a pending error then no state transition should happen.
// But all call-sites should be verifying this before calling us!
RTC_DCHECK(session_error() == SessionError::kNone);
// If this is answer-ish we're ready to let media flow.
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
EnableSending();
}
// Update the signaling state according to the specified state machine (see
// https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum).
if (type == SdpType::kOffer) {
ChangeSignalingState(source == cricket::CS_LOCAL
? PeerConnectionInterface::kHaveLocalOffer
: PeerConnectionInterface::kHaveRemoteOffer);
} else if (type == SdpType::kPrAnswer) {
ChangeSignalingState(source == cricket::CS_LOCAL
? PeerConnectionInterface::kHaveLocalPrAnswer
: PeerConnectionInterface::kHaveRemotePrAnswer);
} else {
RTC_DCHECK(type == SdpType::kAnswer);
ChangeSignalingState(PeerConnectionInterface::kStable);
}
// Update internal objects according to the session description's media
// descriptions.
RTCError error = PushdownMediaDescription(type, source);
if (!error.ok()) {
return error;
}
return RTCError::OK();
}
RTCError PeerConnection::PushdownMediaDescription(
SdpType type,
cricket::ContentSource source) {
const SessionDescriptionInterface* sdesc =
(source == cricket::CS_LOCAL ? local_description()
: remote_description());
RTC_DCHECK(sdesc);
// Push down the new SDP media section for each audio/video transceiver.
for (const auto& transceiver : transceivers_) {
const ContentInfo* content_info =
FindMediaSectionForTransceiver(transceiver, sdesc);
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (!channel || !content_info || content_info->rejected) {
continue;
}
const MediaContentDescription* content_desc =
content_info->media_description();
if (!content_desc) {
continue;
}
std::string error;
bool success = (source == cricket::CS_LOCAL)
? channel->SetLocalContent(content_desc, type, &error)
: channel->SetRemoteContent(content_desc, type, &error);
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
}
}
// If using the RtpDataChannel, push down the new SDP section for it too.
if (rtp_data_channel_) {
const ContentInfo* data_content =
cricket::GetFirstDataContent(sdesc->description());
if (data_content && !data_content->rejected) {
const MediaContentDescription* data_desc =
data_content->media_description();
if (data_desc) {
std::string error;
bool success =
(source == cricket::CS_LOCAL)
? rtp_data_channel_->SetLocalContent(data_desc, type, &error)
: rtp_data_channel_->SetRemoteContent(data_desc, type, &error);
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
}
}
}
}
// Need complete offer/answer with an SCTP m= section before starting SCTP,
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
if (sctp_transport_ && local_description() && remote_description()) {
auto local_sctp_description = cricket::GetFirstSctpDataContentDescription(
local_description()->description());
auto remote_sctp_description = cricket::GetFirstSctpDataContentDescription(
remote_description()->description());
if (local_sctp_description && remote_sctp_description) {
int max_message_size;
// A remote max message size of zero means "any size supported".
// We configure the connection with our own max message size.
if (remote_sctp_description->max_message_size() == 0) {
max_message_size = local_sctp_description->max_message_size();
} else {
max_message_size =
std::min(local_sctp_description->max_message_size(),
remote_sctp_description->max_message_size());
}
sctp_transport_->Start(local_sctp_description->port(),
remote_sctp_description->port(), max_message_size);
}
}
return RTCError::OK();
}
RTCError PeerConnection::PushdownTransportDescription(
cricket::ContentSource source,
SdpType type) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (source == cricket::CS_LOCAL) {
const SessionDescriptionInterface* sdesc = local_description();
RTC_DCHECK(sdesc);
return transport_controller_->SetLocalDescription(type,
sdesc->description());
} else {
const SessionDescriptionInterface* sdesc = remote_description();
RTC_DCHECK(sdesc);
return transport_controller_->SetRemoteDescription(type,
sdesc->description());
}
}
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;
}
cricket::IceConfig PeerConnection::ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) const {
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_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.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.regather_all_networks_interval_range =
config.ice_regather_interval_range;
ice_config.network_preference = config.network_preference;
return ice_config;
}
bool PeerConnection::SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_, "
"sctp_transport_, and media_transport_ are NULL.";
return false;
}
if (media_transport_) {
SendDataParams send_params;
send_params.type = ToWebrtcDataMessageType(params.type);
send_params.ordered = params.ordered;
if (params.max_rtx_count >= 0) {
send_params.max_rtx_count = params.max_rtx_count;
} else if (params.max_rtx_ms >= 0) {
send_params.max_rtx_ms = params.max_rtx_ms;
}
return media_transport_->SendData(params.sid, send_params, payload).ok();
}
return rtp_data_channel_
? rtp_data_channel_->SendData(params, payload, result)
: network_thread()->Invoke<bool>(
RTC_FROM_HERE,
Bind(&cricket::SctpTransportInternal::SendData,
cricket_sctp_transport(), params, payload, result));
}
bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
// Don't log an error here, because DataChannels are expected to call
// ConnectDataChannel in this state. It's the only way to initially tell
// whether or not the underlying transport is ready.
return false;
}
if (media_transport_) {
SignalMediaTransportWritable_s.connect(webrtc_data_channel,
&DataChannel::OnChannelReady);
SignalMediaTransportReceivedData_s.connect(webrtc_data_channel,
&DataChannel::OnDataReceived);
SignalMediaTransportChannelClosing_s.connect(
webrtc_data_channel, &DataChannel::OnClosingProcedureStartedRemotely);
SignalMediaTransportChannelClosed_s.connect(
webrtc_data_channel, &DataChannel::OnClosingProcedureComplete);
} else if (rtp_data_channel_) {
rtp_data_channel_->SignalReadyToSendData.connect(
webrtc_data_channel, &DataChannel::OnChannelReady);
rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel,
&DataChannel::OnDataReceived);
} else {
SignalSctpReadyToSendData.connect(webrtc_data_channel,
&DataChannel::OnChannelReady);
SignalSctpDataReceived.connect(webrtc_data_channel,
&DataChannel::OnDataReceived);
SignalSctpClosingProcedureStartedRemotely.connect(
webrtc_data_channel, &DataChannel::OnClosingProcedureStartedRemotely);
SignalSctpClosingProcedureComplete.connect(
webrtc_data_channel, &DataChannel::OnClosingProcedureComplete);
}
return true;
}
void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
RTC_LOG(LS_ERROR)
<< "DisconnectDataChannel called when rtp_data_channel_ and "
"sctp_transport_ are NULL.";
return;
}
if (media_transport_) {
SignalMediaTransportWritable_s.disconnect(webrtc_data_channel);
SignalMediaTransportReceivedData_s.disconnect(webrtc_data_channel);
SignalMediaTransportChannelClosing_s.disconnect(webrtc_data_channel);
SignalMediaTransportChannelClosed_s.disconnect(webrtc_data_channel);
} else if (rtp_data_channel_) {
rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
} else {
SignalSctpReadyToSendData.disconnect(webrtc_data_channel);
SignalSctpDataReceived.disconnect(webrtc_data_channel);
SignalSctpClosingProcedureStartedRemotely.disconnect(webrtc_data_channel);
SignalSctpClosingProcedureComplete.disconnect(webrtc_data_channel);
}
}
void PeerConnection::AddSctpDataStream(int sid) {
if (media_transport_) {
media_transport_->OpenChannel(sid);
return;
}
if (!sctp_transport_) {
RTC_LOG(LS_ERROR)
<< "AddSctpDataStream called when sctp_transport_ is NULL.";
return;
}
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream,
cricket_sctp_transport(), sid));
}
void PeerConnection::RemoveSctpDataStream(int sid) {
if (media_transport_) {
media_transport_->CloseChannel(sid);
return;
}
if (!sctp_transport_) {
RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
"NULL.";
return;
}
network_thread()->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream,
cricket_sctp_transport(), sid));
}
bool PeerConnection::ReadyToSendData() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
(media_transport_ && media_transport_ready_to_send_data_) ||
sctp_ready_to_send_data_;
}
void PeerConnection::OnDataReceived(int channel_id,
DataMessageType type,
const rtc::CopyOnWriteBuffer& buffer) {
RTC_DCHECK_RUN_ON(network_thread());
cricket::ReceiveDataParams params;
params.sid = channel_id;
params.type = ToCricketDataMessageType(type);
media_transport_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(), [this, params, buffer] {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!HandleOpenMessage_s(params, buffer)) {
SignalMediaTransportReceivedData_s(params, buffer);
}
});
}
void PeerConnection::OnChannelClosing(int channel_id) {
RTC_DCHECK_RUN_ON(network_thread());
media_transport_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(), [this, channel_id] {
RTC_DCHECK_RUN_ON(signaling_thread());
SignalMediaTransportChannelClosing_s(channel_id);
});
}
void PeerConnection::OnChannelClosed(int channel_id) {
RTC_DCHECK_RUN_ON(network_thread());
media_transport_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(), [this, channel_id] {
RTC_DCHECK_RUN_ON(signaling_thread());
SignalMediaTransportChannelClosed_s(channel_id);
});
}
absl::optional<std::string> PeerConnection::sctp_transport_name() const {
RTC_DCHECK_RUN_ON(signaling_thread());
if (sctp_mid_ && transport_controller_) {
auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_);
if (dtls_transport) {
return dtls_transport->transport_name();
}
return absl::optional<std::string>();
}
return absl::optional<std::string>();
}
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
cricket::CandidateStatsList candidate_states_list;
network_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&cricket::PortAllocator::GetCandidateStatsFromPooledSessions,
port_allocator_.get(), &candidate_states_list));
return candidate_states_list;
}
std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
std::map<std::string, std::string> transport_names_by_mid;
for (const auto& transceiver : transceivers_) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel) {
transport_names_by_mid[channel->content_name()] =
channel->transport_name();
}
}
if (rtp_data_channel_) {
transport_names_by_mid[rtp_data_channel_->content_name()] =
rtp_data_channel_->transport_name();
}
if (sctp_transport_) {
absl::optional<std::string> transport_name = sctp_transport_name();
RTC_DCHECK(transport_name);
transport_names_by_mid[*sctp_mid_] = *transport_name;
}
return transport_names_by_mid;
}
std::map<std::string, cricket::TransportStats>
PeerConnection::GetTransportStatsByNames(
const std::set<std::string>& transport_names) {
if (!network_thread()->IsCurrent()) {
return network_thread()
->Invoke<std::map<std::string, cricket::TransportStats>>(
RTC_FROM_HERE,
[&] { return GetTransportStatsByNames(transport_names); });
}
RTC_DCHECK_RUN_ON(network_thread());
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) {
if (!certificate) {
return false;
}
*certificate = transport_controller_->GetLocalCertificate(transport_name);
return *certificate != nullptr;
}
std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
const std::string& transport_name) {
return transport_controller_->GetRemoteSSLCertChain(transport_name);
}
cricket::DataChannelType PeerConnection::data_channel_type() const {
return data_channel_type_;
}
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_ice_restarts_.find(content_name) !=
pending_ice_restarts_.end();
}
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
return transport_controller_->NeedsIceRestart(content_name);
}
void PeerConnection::OnCertificateReady(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
transport_controller_->SetLocalCertificate(certificate);
}
void PeerConnection::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) {
SetSessionError(SessionError::kTransport,
rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp);
}
void PeerConnection::OnTransportControllerConnectionState(
cricket::IceConnectionState state) {
switch (state) {
case cricket::kIceConnectionConnecting:
// If the current state is Connected or Completed, then there were
// writable channels but now there are not, so the next state must
// be Disconnected.
// kIceConnectionConnecting is currently used as the default,
// un-connected state by the TransportController, so its only use is
// detecting disconnections.
if (ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionConnected ||
ice_connection_state_ ==
PeerConnectionInterface::kIceConnectionCompleted) {
SetIceConnectionState(
PeerConnectionInterface::kIceConnectionDisconnected);
}
break;
case cricket::kIceConnectionFailed:
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
break;
case cricket::kIceConnectionConnected:
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
"all transports are writable.";
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
break;
case cricket::kIceConnectionCompleted:
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
"all transports are complete.";
if (ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionConnected) {
// If jumping directly from "checking" to "connected",
// signal "connected" first.
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
ReportTransportStats();
break;
default:
RTC_NOTREACHED();
}
}
void PeerConnection::OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const cricket::Candidates& candidates) {
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));
if (local_description()) {
mutable_local_description()->AddCandidate(candidate.get());
}
OnIceCandidate(std::move(candidate));
}
}
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;
}
}
if (local_description()) {
mutable_local_description()->RemoveCandidates(candidates);
}
OnIceCandidatesRemoved(candidates);
}
void PeerConnection::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
}
void PeerConnection::EnableSending() {
for (const auto& transceiver : transceivers_) {
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel && !channel->enabled()) {
channel->Enable(true);
}
}
if (rtp_data_channel_ && !rtp_data_channel_->enabled()) {
rtp_data_channel_->Enable(true);
}
}
// 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;
}
bool PeerConnection::UseCandidatesInSessionDescription(
const SessionDescriptionInterface* remote_desc) {
if (!remote_desc) {
return true;
}
bool ret = true;
for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
const IceCandidateCollection* candidates = remote_desc->candidates(m);
for (size_t n = 0; n < candidates->count(); ++n) {
const IceCandidateInterface* candidate = candidates->at(n);
bool valid = false;
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
if (valid) {
RTC_LOG(LS_INFO)
<< "UseCandidatesInSessionDescription: Not ready to use "
"candidate.";
}
continue;
}
ret = UseCandidate(candidate);
if (!ret) {
break;
}
}
}
return ret;
}
bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
RTCErrorOr<const cricket::ContentInfo*> result =
FindContentInfo(remote_description(), candidate);
if (!result.ok()) {
RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate. "
<< result.error().message();
return false;
}
std::vector<cricket::Candidate> candidates;
candidates.push_back(candidate->candidate());
// Invoking BaseSession method to handle remote candidates.
RTCError error = transport_controller_->AddRemoteCandidates(
result.value()->name, candidates);
if (error.ok()) {
// 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();
}
return true;
}
RTCErrorOr<const cricket::ContentInfo*> PeerConnection::FindContentInfo(
const SessionDescriptionInterface* description,
const IceCandidateInterface* candidate) {
if (candidate->sdp_mline_index() >= 0) {
size_t mediacontent_index =
static_cast<size_t>(candidate->sdp_mline_index());
size_t content_size = description->description()->contents().size();
if (mediacontent_index < content_size) {
return &description->description()->contents()[mediacontent_index];
} else {
return RTCError(RTCErrorType::INVALID_RANGE,
"Media line index (" +
rtc::ToString(candidate->sdp_mline_index()) +
") out of range (number of mlines: " +
rtc::ToString(content_size) + ").");
}
} else if (!candidate->sdp_mid().empty()) {
auto& contents = description->description()->contents();
auto it = absl::c_find_if(
contents, [candidate](const cricket::ContentInfo& content_info) {
return content_info.mid() == candidate->sdp_mid();
});
if (it == contents.end()) {
return RTCError(
RTCErrorType::INVALID_PARAMETER,
"Mid " + candidate->sdp_mid() +
" specified but no media section with that mid found.");
} else {
return &*it;
}
}
return RTCError(RTCErrorType::INVALID_PARAMETER,
"Neither sdp_mline_index nor sdp_mid specified.");
}
void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) {
// Destroy video channel first since it may have a pointer to the
// voice channel.
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
if (!video_info || video_info->rejected) {
DestroyTransceiverChannel(GetVideoTransceiver());
}
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
if (!audio_info || audio_info->rejected) {
DestroyTransceiverChannel(GetAudioTransceiver());
}
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
if (!data_info || data_info->rejected) {
DestroyDataChannel();
}
}
RTCErrorOr<const cricket::ContentGroup*> PeerConnection::GetEarlyBundleGroup(
const SessionDescription& desc) const {
const cricket::ContentGroup* bundle_group = nullptr;
if (configuration_.bundle_policy ==
PeerConnectionInterface::kBundlePolicyMaxBundle) {
bundle_group = desc.GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
if (!bundle_group) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"max-bundle configured but session description "
"has no BUNDLE group");
}
}
return bundle_group;
}
RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
// Creating the media channels. Transports should already have been created
// at this point.
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
if (voice && !voice->rejected &&
!GetAudioTransceiver()->internal()->channel()) {
cricket::VoiceChannel* voice_channel = CreateVoiceChannel(voice->name);
if (!voice_channel) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create voice channel.");
}
GetAudioTransceiver()->internal()->SetChannel(voice_channel);
}
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
if (video && !video->rejected &&
!GetVideoTransceiver()->internal()->channel()) {
cricket::VideoChannel* video_channel = CreateVideoChannel(video->name);
if (!video_channel) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create video channel.");
}
GetVideoTransceiver()->internal()->SetChannel(video_channel);
}
const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
!rtp_data_channel_ && !sctp_transport_ && !media_transport_) {
if (!CreateDataChannel(data->name)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create data channel.");
}
}
return RTCError::OK();
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
const std::string& mid) {
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
MediaTransportConfig media_transport_config =
transport_controller_->GetMediaTransportConfig(mid);
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
call_ptr_, configuration_.media_config, rtp_transport,
media_transport_config, signaling_thread(), mid, SrtpRequired(),
GetCryptoOptions(), &ssrc_generator_, audio_options_);
if (!voice_channel) {
return nullptr;
}
voice_channel->SignalDtlsSrtpSetupFailure.connect(
this, &PeerConnection::OnDtlsSrtpSetupFailure);
voice_channel->SignalSentPacket.connect(this,
&PeerConnection::OnSentPacket_w);
voice_channel->SetRtpTransport(rtp_transport);
return voice_channel;
}
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
const std::string& mid) {
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
MediaTransportConfig media_transport_config =
transport_controller_->GetMediaTransportConfig(mid);
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
call_ptr_, configuration_.media_config, rtp_transport,
media_transport_config, signaling_thread(), mid, SrtpRequired(),
GetCryptoOptions(), &ssrc_generator_, video_options_,
video_bitrate_allocator_factory_.get());
if (!video_channel) {
return nullptr;
}
video_channel->SignalDtlsSrtpSetupFailure.connect(
this, &PeerConnection::OnDtlsSrtpSetupFailure);
video_channel->SignalSentPacket.connect(this,
&PeerConnection::OnSentPacket_w);
video_channel->SetRtpTransport(rtp_transport);
return video_channel;
}
bool PeerConnection::CreateDataChannel(const std::string& mid) {
switch (data_channel_type_) {
case cricket::DCT_MEDIA_TRANSPORT:
if (network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetupMediaTransportForDataChannels_n,
this, mid))) {
for (const auto& channel : sctp_data_channels_) {
channel->OnTransportChannelCreated();
}
return true;
}
return false;
case cricket::DCT_SCTP:
if (!sctp_factory_) {
RTC_LOG(LS_ERROR)
<< "Trying to create SCTP transport, but didn't compile with "
"SCTP support (HAVE_SCTP)";
return false;
}
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::CreateSctpTransport_n, this, mid))) {
return false;
}
for (const auto& channel : sctp_data_channels_) {
channel->OnTransportChannelCreated();
}
return true;
case cricket::DCT_RTP:
default:
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
configuration_.media_config, rtp_transport, signaling_thread(), mid,
SrtpRequired(), GetCryptoOptions(), &ssrc_generator_);
if (!rtp_data_channel_) {
return false;
}
rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
this, &PeerConnection::OnDtlsSrtpSetupFailure);
rtp_data_channel_->SignalSentPacket.connect(
this, &PeerConnection::OnSentPacket_w);
rtp_data_channel_->SetRtpTransport(rtp_transport);
return true;
}
return true;
}
Call::Stats PeerConnection::GetCallStats() {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<Call::Stats>(
RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this));
}
RTC_DCHECK_RUN_ON(worker_thread());
if (call_) {
return call_->GetStats();
} else {
return Call::Stats();
}
}
bool PeerConnection::CreateSctpTransport_n(const std::string& mid) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(sctp_factory_);
rtc::scoped_refptr<DtlsTransport> webrtc_dtls_transport =
transport_controller_->LookupDtlsTransportByMid(mid);
cricket::DtlsTransportInternal* dtls_transport =
webrtc_dtls_transport->internal();
RTC_DCHECK(dtls_transport);
std::unique_ptr<cricket::SctpTransportInternal> cricket_sctp_transport =
sctp_factory_->CreateSctpTransport(dtls_transport);
RTC_DCHECK(cricket_sctp_transport);
sctp_invoker_.reset(new rtc::AsyncInvoker());
cricket_sctp_transport->SignalReadyToSendData.connect(
this, &PeerConnection::OnSctpTransportReadyToSendData_n);
cricket_sctp_transport->SignalDataReceived.connect(
this, &PeerConnection::OnSctpTransportDataReceived_n);
// TODO(deadbeef): All we do here is AsyncInvoke to fire the signal on
// another thread. Would be nice if there was a helper class similar to
// sigslot::repeater that did this for us, eliminating a bunch of boilerplate
// code.
cricket_sctp_transport->SignalClosingProcedureStartedRemotely.connect(
this, &PeerConnection::OnSctpClosingProcedureStartedRemotely_n);
cricket_sctp_transport->SignalClosingProcedureComplete.connect(
this, &PeerConnection::OnSctpClosingProcedureComplete_n);
sctp_mid_ = mid;
sctp_transport_ = new rtc::RefCountedObject<SctpTransport>(
std::move(cricket_sctp_transport));
sctp_transport_->SetDtlsTransport(std::move(webrtc_dtls_transport));
return true;
}
void PeerConnection::DestroySctpTransport_n() {
RTC_DCHECK_RUN_ON(network_thread());
sctp_transport_->Clear();
sctp_transport_ = nullptr;
sctp_mid_.reset();
sctp_invoker_.reset(nullptr);
}
void PeerConnection::OnSctpTransportReadyToSendData_n() {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
// Note: Cannot use rtc::Bind here because it will grab a reference to
// PeerConnection and potentially cause PeerConnection to live longer than
// expected. It is safe not to grab a reference since the sctp_invoker_ will
// be destroyed before PeerConnection is destroyed, and at that point all
// pending tasks will be cleared.
sctp_invoker_->AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(), [this] {
OnSctpTransportReadyToSendData_s(true);
});
}
void PeerConnection::OnSctpTransportReadyToSendData_s(bool ready) {
RTC_DCHECK_RUN_ON(signaling_thread());
sctp_ready_to_send_data_ = ready;
SignalSctpReadyToSendData(ready);
}
void PeerConnection::OnSctpTransportDataReceived_n(
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
// Note: Cannot use rtc::Bind here because it will grab a reference to
// PeerConnection and potentially cause PeerConnection to live longer than
// expected. It is safe not to grab a reference since the sctp_invoker_ will
// be destroyed before PeerConnection is destroyed, and at that point all
// pending tasks will be cleared.
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(), [this, params, payload] {
OnSctpTransportDataReceived_s(params, payload);
});
}
void PeerConnection::OnSctpTransportDataReceived_s(
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!HandleOpenMessage_s(params, payload)) {
SignalSctpDataReceived(params, payload);
}
}
void PeerConnection::OnSctpClosingProcedureStartedRemotely_n(int sid) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(),
rtc::Bind(&sigslot::signal1<int>::operator(),
&SignalSctpClosingProcedureStartedRemotely, sid));
}
void PeerConnection::OnSctpClosingProcedureComplete_n(int sid) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(),
rtc::Bind(&sigslot::signal1<int>::operator(),
&SignalSctpClosingProcedureComplete, sid));
}
bool PeerConnection::SetupMediaTransportForDataChannels_n(
const std::string& mid) {
media_transport_ =
transport_controller_->GetMediaTransportForDataChannel(mid);
if (!media_transport_) {
RTC_LOG(LS_ERROR)
<< "Media transport is not available for data channels, mid=" << mid;
return false;
}
media_transport_invoker_ = absl::make_unique<rtc::AsyncInvoker>();
media_transport_->SetDataSink(this);
media_transport_data_mid_ = mid;
transport_controller_->SignalMediaTransportStateChanged.connect(
this, &PeerConnection::OnMediaTransportStateChanged_n);
// Check the initial state right away, in case transport is already writable.
OnMediaTransportStateChanged_n();
return true;
}
void PeerConnection::TeardownMediaTransportForDataChannels_n() {
if (!media_transport_) {
return;
}
transport_controller_->SignalMediaTransportStateChanged.disconnect(this);
media_transport_data_mid_.reset();
media_transport_->SetDataSink(nullptr);
media_transport_invoker_ = nullptr;
media_transport_ = nullptr;
}
void PeerConnection::OnMediaTransportStateChanged_n() {
if (!media_transport_data_mid_ ||
transport_controller_->GetMediaTransportState(
*media_transport_data_mid_) != MediaTransportState::kWritable) {
return;
}
media_transport_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread(), [this] {
RTC_DCHECK_RUN_ON(signaling_thread());
media_transport_ready_to_send_data_ = true;
SignalMediaTransportWritable_s(media_transport_ready_to_send_data_);
});
}
// Returns false if bundle is enabled and rtcp_mux is disabled.
bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
if (!bundle_enabled)
return true;
const cricket::ContentGroup* bundle_group =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
RTC_DCHECK(bundle_group != NULL);
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);
if (bundle_group->HasContentName(content->name) && !content->rejected &&
content->type == MediaProtocolType::kRtp) {
if (!HasRtcpMuxEnabled(content))
return false;
}
}
// RTCP-MUX is enabled in all the contents.
return true;
}
bool PeerConnection::HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
return content->media_description()->rtcp_mux();
}
static RTCError ValidateMids(const cricket::SessionDescription& description) {
std::set<std::string> mids;
for (const cricket::ContentInfo& content : description.contents()) {
if (content.name.empty()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"A media section is missing a MID attribute.");
}
if (!mids.insert(content.name).second) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Duplicate a=mid value '" + content.name + "'.");
}
}
return RTCError::OK();
}
RTCError PeerConnection::ValidateSessionDescription(
const SessionDescriptionInterface* sdesc,
cricket::ContentSource source) {
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
if (!sdesc || !sdesc->description()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
}
SdpType type = sdesc->GetType();
if ((source == cricket::CS_LOCAL && !ExpectSetLocalDescription(type)) ||
(source == cricket::CS_REMOTE && !ExpectSetRemoteDescription(type))) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_STATE,
"Called in wrong state: " + GetSignalingStateString(signaling_state()));
}
RTCError error = ValidateMids(*sdesc->description());
if (!error.ok()) {
return error;
}
// Verify crypto settings.
std::string crypto_error;
if (webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
dtls_enabled_) {
RTCError crypto_error = VerifyCrypto(sdesc->description(), dtls_enabled_);
if (!crypto_error.ok()) {
return crypto_error;
}
}
// Verify ice-ufrag and ice-pwd.
if (!VerifyIceUfragPwdPresent(sdesc->description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kSdpWithoutIceUfragPwd);
}
if (!ValidateBundleSettings(sdesc->description())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kBundleWithoutRtcpMux);
}
// TODO(skvlad): When the local rtcp-mux policy is Require, reject any
// m-lines that do not rtcp-mux enabled.
// Verify m-lines in Answer when compared against Offer.
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
// With an answer we want to compare the new answer session description with
// the offer's session description from the current negotiation.
const cricket::SessionDescription* offer_desc =
(source == cricket::CS_LOCAL) ? remote_description()->description()
: local_description()->description();
if (!MediaSectionsHaveSameCount(*offer_desc, *sdesc->description()) ||
!MediaSectionsInSameOrder(*offer_desc, nullptr, *sdesc->description(),
type)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kMlineMismatchInAnswer);
}
} else {
// The re-offers should respect the order of m= sections in current
// description. See RFC3264 Section 8 paragraph 4 for more details.
// With a re-offer, either the current local or current remote descriptions
// could be the most up to date, so we would like to check against both of
// them if they exist. It could be the case that one of them has a 0 port
// for a media section, but the other does not. This is important to check
// against in the case that we are recycling an m= section.
const cricket::SessionDescription* current_desc = nullptr;
const cricket::SessionDescription* secondary_current_desc = nullptr;
if (local_description()) {
current_desc = local_description()->description();
if (remote_description()) {
secondary_current_desc = remote_description()->description();
}
} else if (remote_description()) {
current_desc = remote_description()->description();
}
if (current_desc &&
!MediaSectionsInSameOrder(*current_desc, secondary_current_desc,
*sdesc->description(), type)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kMlineMismatchInSubsequentOffer);
}
}
if (IsUnifiedPlan()) {
// Ensure that each audio and video media section has at most one
// "StreamParams". This will return an error if receiving a session
// description from a "Plan B" endpoint which adds multiple tracks of the
// same type. With Unified Plan, there can only be at most one track per
// media section.
for (const ContentInfo& content : sdesc->description()->contents()) {
const MediaContentDescription& desc = *content.description;
if ((desc.type() == cricket::MEDIA_TYPE_AUDIO ||
desc.type() == cricket::MEDIA_TYPE_VIDEO) &&
desc.streams().size() > 1u) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Media section has more than one track specified "
"with a=ssrc lines which is not supported with "
"Unified Plan.");
}
}
}
return RTCError::OK();
}
bool PeerConnection::ExpectSetLocalDescription(SdpType type) {
PeerConnectionInterface::SignalingState state = signaling_state();
if (type == SdpType::kOffer) {
return (state == PeerConnectionInterface::kStable) ||
(state == PeerConnectionInterface::kHaveLocalOffer);
} else {
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
return (state == PeerConnectionInterface::kHaveRemoteOffer) ||
(state == PeerConnectionInterface::kHaveLocalPrAnswer);
}
}
bool PeerConnection::ExpectSetRemoteDescription(SdpType type) {
PeerConnectionInterface::SignalingState state = signaling_state();
if (type == SdpType::kOffer) {
return (state == PeerConnectionInterface::kStable) ||
(state == PeerConnectionInterface::kHaveRemoteOffer);
} else {
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
return (state == PeerConnectionInterface::kHaveLocalOffer) ||
(state == PeerConnectionInterface::kHaveRemotePrAnswer);
}
}
const char* PeerConnection::SessionErrorToString(SessionError error) const {
switch (error) {
case SessionError::kNone:
return "ERROR_NONE";
case SessionError::kContent:
return "ERROR_CONTENT";
case SessionError::kTransport:
return "ERROR_TRANSPORT";
}
RTC_NOTREACHED();
return "";
}
std::string PeerConnection::GetSessionErrorMsg() {
rtc::StringBuilder desc;
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
desc << kSessionErrorDesc << session_error_desc() << ".";
return desc.Release();
}
void PeerConnection::ReportSdpFormatReceived(
const SessionDescriptionInterface& remote_offer) {
int num_audio_mlines = 0;
int num_video_mlines = 0;
int num_audio_tracks = 0;
int num_video_tracks = 0;
for (const ContentInfo& content : remote_offer.description()->contents()) {
cricket::MediaType media_type = content.media_description()->type();
int num_tracks = std::max(
1, static_cast<int>(content.media_description()->streams().size()));
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
num_audio_mlines += 1;
num_audio_tracks += num_tracks;
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
num_video_mlines += 1;
num_video_tracks += num_tracks;
}
}
SdpFormatReceived format = kSdpFormatReceivedNoTracks;
if (num_audio_mlines > 1 || num_video_mlines > 1) {
format = kSdpFormatReceivedComplexUnifiedPlan;
} else if (num_audio_tracks > 1 || num_video_tracks > 1) {
format = kSdpFormatReceivedComplexPlanB;
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
format = kSdpFormatReceivedSimple;
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
kSdpFormatReceivedMax);
}
void PeerConnection::NoteUsageEvent(UsageEvent event) {
RTC_DCHECK_RUN_ON(signaling_thread());
usage_event_accumulator_ |= static_cast<int>(event);
}
void PeerConnection::ReportUsagePattern() const {
RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
usage_event_accumulator_,
static_cast<int>(UsageEvent::MAX_VALUE));
const int bad_bits =
static_cast<int>(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED) |
static_cast<int>(UsageEvent::CANDIDATE_COLLECTED);
const int good_bits =
static_cast<int>(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED) |
static_cast<int>(UsageEvent::REMOTE_CANDIDATE_ADDED) |
static_cast<int>(UsageEvent::ICE_STATE_CONNECTED);
if ((usage_event_accumulator_ & bad_bits) == bad_bits &&
(usage_event_accumulator_ & good_bits) == 0) {
// If called after close(), we can't report, because observer may have
// been deallocated, and therefore pointer is null. Write to log instead.
if (observer_) {
Observer()->OnInterestingUsage(usage_event_accumulator_);
} else {
RTC_LOG(LS_INFO) << "Interesting usage signature "
<< usage_event_accumulator_
<< " observed after observer shutdown";
}
}
}
void PeerConnection::ReportNegotiatedSdpSemantics(
const SessionDescriptionInterface& answer) {
SdpSemanticNegotiated semantics_negotiated;
switch (answer.description()->msid_signaling()) {
case 0:
semantics_negotiated = kSdpSemanticNegotiatedNone;
break;
case cricket::kMsidSignalingMediaSection:
semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
break;
case cricket::kMsidSignalingSsrcAttribute:
semantics_negotiated = kSdpSemanticNegotiatedPlanB;
break;
case cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute:
semantics_negotiated = kSdpSemanticNegotiatedMixed;
break;
default:
RTC_NOTREACHED();
}
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpSemanticNegotiated",
semantics_negotiated, kSdpSemanticNegotiatedMax);
}
// We need to check the local/remote description for the Transport instead of
// the session, because a new Transport added during renegotiation may have
// them unset while the session has them set from the previous negotiation.
// Not doing so may trigger the auto generation of transport description and
// mess up DTLS identity information, ICE credential, etc.
bool PeerConnection::ReadyToUseRemoteCandidate(
const IceCandidateInterface* candidate,
const SessionDescriptionInterface* remote_desc,
bool* valid) {
*valid = true;
const SessionDescriptionInterface* current_remote_desc =
remote_desc ? remote_desc : remote_description();
if (!current_remote_desc) {
return false;
}
RTCErrorOr<const cricket::ContentInfo*> result =
FindContentInfo(current_remote_desc, candidate);
if (!result.ok()) {
RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate. "
<< result.error().message();
*valid = false;
return false;
}
std::string transport_name = GetTransportName(result.value()->name);
return !transport_name.empty();
}
bool PeerConnection::SrtpRequired() const {
return !configuration_.use_datagram_transport &&
(dtls_enabled_ ||
webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED);
}
void PeerConnection::OnTransportControllerGatheringState(
cricket::IceGatheringState state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (state == cricket::kIceGatheringGathering) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
} else if (state == cricket::kIceGatheringComplete) {
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
}
}
void PeerConnection::ReportTransportStats() {
std::map<std::string, std::set<cricket::MediaType>>
media_types_by_transport_name;
for (const auto& transceiver : transceivers_) {
if (transceiver->internal()->channel()) {
const std::string& transport_name =
transceiver->internal()->channel()->transport_name();
media_types_by_transport_name[transport_name].insert(
transceiver->media_type());
}
}
if (rtp_data_channel()) {
media_types_by_transport_name[rtp_data_channel()->transport_name()].insert(
cricket::MEDIA_TYPE_DATA);
}
absl::optional<std::string> transport_name = sctp_transport_name();
if (transport_name) {
media_types_by_transport_name[*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(stats, media_types);
}
}
}
// Walk through the ConnectionInfos to gather best connection usage
// for IPv4 and IPv6.
void PeerConnection::ReportBestConnectionState(
const cricket::TransportStats& stats) {
for (const cricket::TransportChannelStats& channel_stats :
stats.channel_stats) {
for (const cricket::ConnectionInfo& connection_info :
channel_stats.connection_infos) {
if (!connection_info.best_connection) {
continue;
}
const cricket::Candidate& local = connection_info.local_candidate;
const cricket::Candidate& remote = connection_info.remote_candidate;
// Increment the counter for IceCandidatePairType.
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
(local.type() == RELAY_PORT_TYPE &&
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
} else {
RTC_CHECK(0);
}
// 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;
}
}
}
void PeerConnection::ReportNegotiatedCiphers(
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::SRTP_INVALID_CRYPTO_SUITE &&
ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
return;
}
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
for (cricket::MediaType media_type : media_types) {
switch (media_type) {
case cricket::MEDIA_TYPE_AUDIO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
break;
default:
RTC_NOTREACHED();
continue;
}
}
}
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
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::SSL_CIPHER_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_VIDEO:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
break;
case cricket::MEDIA_TYPE_DATA:
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
break;
default:
RTC_NOTREACHED();
continue;
}
}
}
}
void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
RTC_DCHECK_RUN_ON(worker_thread());
RTC_DCHECK(call_);
call_->OnSentPacket(sent_packet);
}
const std::string PeerConnection::GetTransportName(
const std::string& content_name) {
cricket::ChannelInterface* channel = GetChannel(content_name);
if (channel) {
return channel->transport_name();
}
if (sctp_transport_) {
RTC_DCHECK(sctp_mid_);
if (content_name == *sctp_mid_) {
return *sctp_transport_name();
}
}
// Return an empty string if failed to retrieve the transport name.
return "";
}
void PeerConnection::DestroyTransceiverChannel(
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
transceiver) {
RTC_DCHECK(transceiver);
cricket::ChannelInterface* channel = transceiver->internal()->channel();
if (channel) {
transceiver->internal()->SetChannel(nullptr);
DestroyChannelInterface(channel);
}
}
void PeerConnection::DestroyDataChannel() {
if (rtp_data_channel_) {
OnDataChannelDestroyed();
DestroyChannelInterface(rtp_data_channel_);
rtp_data_channel_ = nullptr;
}
// Note: Cannot use rtc::Bind to create a functor to invoke because it will
// grab a reference to this PeerConnection. If this is called from the
// PeerConnection destructor, the RefCountedObject vtable will have already
// been destroyed (since it is a subclass of PeerConnection) and using
// rtc::Bind will cause "Pure virtual function called" error to appear.
if (sctp_transport_) {
OnDataChannelDestroyed();
network_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { DestroySctpTransport_n(); });
sctp_ready_to_send_data_ = false;
}
if (media_transport_) {
OnDataChannelDestroyed();
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
RTC_DCHECK_RUN_ON(network_thread());
TeardownMediaTransportForDataChannels_n();
});
}
}
void PeerConnection::DestroyChannelInterface(
cricket::ChannelInterface* channel) {
RTC_DCHECK(channel);
switch (channel->media_type()) {
case cricket::MEDIA_TYPE_AUDIO:
channel_manager()->DestroyVoiceChannel(
static_cast<cricket::VoiceChannel*>(channel));
break;
case cricket::MEDIA_TYPE_VIDEO:
channel_manager()->DestroyVideoChannel(
static_cast<cricket::VideoChannel*>(channel));
break;
case cricket::MEDIA_TYPE_DATA:
channel_manager()->DestroyRtpDataChannel(
static_cast<cricket::RtpDataChannel*>(channel));
break;
default:
RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
break;
}
}
bool PeerConnection::OnTransportChanged(
const std::string& mid,
RtpTransportInternal* rtp_transport,
rtc::scoped_refptr<DtlsTransport> dtls_transport,
MediaTransportInterface* media_transport) {
RTC_DCHECK_RUN_ON(network_thread());
RTC_DCHECK_RUNS_SERIALIZED(&use_media_transport_race_checker_);
bool ret = true;
auto base_channel = GetChannel(mid);
if (base_channel) {
ret = base_channel->SetRtpTransport(rtp_transport);
}
if (sctp_transport_ && mid == sctp_mid_) {
sctp_transport_->SetDtlsTransport(dtls_transport);
}
if (use_media_transport_) {
// Only pass media transport to call object if media transport is used
// for media (and not data channel).
call_ptr_->MediaTransportChange(media_transport);
}
return ret;
}
void PeerConnection::OnSetStreams() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (IsUnifiedPlan())
UpdateNegotiationNeeded();
}
PeerConnectionObserver* PeerConnection::Observer() const {
// In earlier production code, the pointer was not cleared on close,
// which might have led to undefined behavior if the observer was not
// deallocated, or strange crashes if it was.
// We use CHECK in order to catch such behavior if it exists.
// TODO(hta): Remove or replace with DCHECK if nothing is found.
RTC_CHECK(observer_);
return observer_;
}
CryptoOptions PeerConnection::GetCryptoOptions() {
// TODO(bugs.webrtc.org/9891) - Remove PeerConnectionFactory::CryptoOptions
// after it has been removed.
return configuration_.crypto_options.has_value()
? *configuration_.crypto_options
: factory_->options().crypto_options;
}
void PeerConnection::ClearStatsCache() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (stats_collector_) {
stats_collector_->ClearCachedStatsReport();
}
}
void PeerConnection::RequestUsagePatternReportForTesting() {
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN,
nullptr);
}
void PeerConnection::UpdateNegotiationNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!IsUnifiedPlan()) {
Observer()->OnRenegotiationNeeded();
return;
}
// If connection's [[IsClosed]] slot is true, abort these steps.
if (IsClosed())
return;
// If connection's signaling state is not "stable", abort these steps.
if (signaling_state() != kStable)
return;
// NOTE
// The negotiation-needed flag will be updated once the state transitions to
// "stable", as part of the steps for setting an RTCSessionDescription.
// If the result of checking if negotiation is needed is false, clear the
// negotiation-needed flag by setting connection's [[NegotiationNeeded]] slot
// to false, and abort these steps.
bool is_negotiation_needed = CheckIfNegotiationIsNeeded();
if (!is_negotiation_needed) {
is_negotiation_needed_ = false;
return;
}
// If connection's [[NegotiationNeeded]] slot is already true, abort these
// steps.
if (is_negotiation_needed_)
return;
// Set connection's [[NegotiationNeeded]] slot to true.
is_negotiation_needed_ = true;
// Queue a task that runs the following steps:
// If connection's [[IsClosed]] slot is true, abort these steps.
// If connection's [[NegotiationNeeded]] slot is false, abort these steps.
// Fire an event named negotiationneeded at connection.
Observer()->OnRenegotiationNeeded();
}
bool PeerConnection::CheckIfNegotiationIsNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
// 1. If any implementation-specific negotiation is required, as described at
// the start of this section, return true.
// 2. Let description be connection.[[CurrentLocalDescription]].
const SessionDescriptionInterface* description = current_local_description();
if (!description)
return true;
// 3. If connection has created any RTCDataChannels, and no m= section in
// description has been negotiated yet for data, return true.
if (!sctp_data_channels_.empty()) {
if (!cricket::GetFirstDataContent(description->description()->contents()))
return true;
}
// 4. For each transceiver in connection's set of transceivers, perform the
// following checks:
for (const auto& transceiver : transceivers_) {
const ContentInfo* current_local_msection =
FindTransceiverMSection(transceiver.get(), description);
const ContentInfo* current_remote_msection = FindTransceiverMSection(
transceiver.get(), current_remote_description());
// 4.3 If transceiver is stopped and is associated with an m= section,
// but the associated m= section is not yet rejected in
// connection.[[CurrentLocalDescription]] or
// connection.[[CurrentRemoteDescription]], return true.
if (transceiver->stopped()) {
if (current_local_msection && !current_local_msection->rejected &&
((current_remote_msection && !current_remote_msection->rejected) ||
!current_remote_msection)) {
return true;
}
continue;
}
// 4.1 If transceiver isn't stopped and isn't yet associated with an m=
// section in description, return true.
if (!current_local_msection)
return true;
const MediaContentDescription* current_local_media_description =
current_local_msection->media_description();
// 4.2 If transceiver isn't stopped and is associated with an m= section
// in description then perform the following checks:
// 4.2.1 If transceiver.[[Direction]] is "sendrecv" or "sendonly", and the
// associated m= section in description either doesn't contain a single
// "a=msid" line, or the number of MSIDs from the "a=msid" lines in this
// m= section, or the MSID values themselves, differ from what is in
// transceiver.sender.[[AssociatedMediaStreamIds]], return true.
if (RtpTransceiverDirectionHasSend(transceiver->direction())) {
if (current_local_media_description->streams().size() == 0)
return true;
std::vector<std::string> msection_msids;
for (const auto& stream : current_local_media_description->streams()) {
for (const std::string& msid : stream.stream_ids())
msection_msids.push_back(msid);
}
std::vector<std::string> transceiver_msids =
transceiver->sender()->stream_ids();
if (msection_msids.size() != transceiver_msids.size())
return true;
absl::c_sort(transceiver_msids);
absl::c_sort(msection_msids);
if (transceiver_msids != msection_msids)
return true;
}
// 4.2.2 If description is of type "offer", and the direction of the
// associated m= section in neither connection.[[CurrentLocalDescription]]
// nor connection.[[CurrentRemoteDescription]] matches
// transceiver.[[Direction]], return true.
if (description->GetType() == SdpType::kOffer) {
if (!current_remote_description())
return true;
if (!current_remote_msection)
return true;
RtpTransceiverDirection current_local_direction =
current_local_media_description->direction();
RtpTransceiverDirection current_remote_direction =
current_remote_msection->media_description()->direction();
if (transceiver->direction() != current_local_direction &&
transceiver->direction() !=
RtpTransceiverDirectionReversed(current_remote_direction)) {
return true;
}
}
// 4.2.3 If description is of type "answer", and the direction of the
// associated m= section in the description does not match
// transceiver.[[Direction]] intersected with the offered direction (as
// described in [JSEP] (section 5.3.1.)), return true.
if (description->GetType() == SdpType::kAnswer) {
if (!remote_description())
return true;
const ContentInfo* offered_remote_msection =
FindTransceiverMSection(transceiver.get(), remote_description());
RtpTransceiverDirection offered_direction =
offered_remote_msection
? offered_remote_msection->media_description()->direction()
: RtpTransceiverDirection::kInactive;
if (current_local_media_description->direction() !=
(RtpTransceiverDirectionIntersection(
transceiver->direction(),
RtpTransceiverDirectionReversed(offered_direction)))) {
return true;
}
}
}
// If all the preceding checks were performed and true was not returned,
// nothing remains to be negotiated; return false.
return false;
}
} // namespace webrtc