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peerconnection.cc
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/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/peerconnection.h"
#include <algorithm>
#include <limits>
#include <queue>
#include <set>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/jsepicecandidate.h"
#include "api/jsepsessiondescription.h"
#include "api/mediaconstraintsinterface.h"
#include "api/mediastreamproxy.h"
#include "api/mediastreamtrackproxy.h"
#include "call/call.h"
#include "logging/rtc_event_log/icelogger.h"
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/sctp/sctptransport.h"
#include "pc/audiotrack.h"
#include "pc/channel.h"
#include "pc/channelmanager.h"
#include "pc/dtmfsender.h"
#include "pc/mediastream.h"
#include "pc/mediastreamobserver.h"
#include "pc/remoteaudiosource.h"
#include "pc/rtpmediautils.h"
#include "pc/rtpreceiver.h"
#include "pc/rtpsender.h"
#include "pc/sctputils.h"
#include "pc/sdputils.h"
#include "pc/streamcollection.h"
#include "pc/videocapturertracksource.h"
#include "pc/videotrack.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/stringencode.h"
#include "rtc_base/stringutils.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::SessionDescription;
using cricket::MediaProtocolType;
using cricket::TransportInfo;
using cricket::LOCAL_PORT_TYPE;
using cricket::STUN_PORT_TYPE;
using cricket::RELAY_PORT_TYPE;
using cricket::PRFLX_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 {
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 AddRtpSenderOptions(
const std::vector<rtc::scoped_refptr<
RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
cricket::MediaDescriptionOptions* audio_media_description_options,
cricket::MediaDescriptionOptions* video_media_description_options) {
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(), 1);
}
}
}
}
// 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_private = IPIsPrivate(local.address().ipaddr());
bool remote_private = IPIsPrivate(remote.address().ipaddr());
if (local_private) {
if (remote_private) {
return kIceCandidatePairHostPrivateHostPrivate;
} else {
return kIceCandidatePairHostPrivateHostPublic;
}
} 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],
¤t_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) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
kEnumCounterKeyProtocolMax);
static const std::map<std::pair<KeyExchangeProtocolType, cricket::MediaType>,
KeyExchangeProtocolMedia>
proto_media_counter_map = {
{{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}};
auto it = proto_media_counter_map.find({protocol_type, media_type});
if (it != proto_media_counter_map.end()) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
it->second, 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;
}
bool GetTrackIdBySsrc(const SessionDescription* session_description,
uint32_t ssrc,
std::string* track_id) {
RTC_DCHECK(track_id != NULL);
const cricket::AudioContentDescription* audio_desc =
cricket::GetFirstAudioContentDescription(session_description);
if (audio_desc) {
const auto* found = cricket::GetStreamBySsrc(audio_desc->streams(), ssrc);
if (found) {
*track_id = found->id;
return true;
}
}
const cricket::VideoContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(session_description);
if (video_desc) {
const auto* found = cricket::GetStreamBySsrc(video_desc->streams(), ssrc);
if (found) {
*track_id = found->id;
return true;
}
}
return false;
}
// Get the SCTP port out of a SessionDescription.
// Return -1 if not found.
int GetSctpPort(const SessionDescription* session_description) {
const cricket::DataContentDescription* data_desc =
GetFirstDataContentDescription(session_description);
RTC_DCHECK(data_desc);
if (!data_desc) {
return -1;
}
std::string value;
cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
cricket::kGoogleSctpDataCodecName);
for (const cricket::DataCodec& codec : data_desc->codecs()) {
if (!codec.Matches(match_pattern)) {
continue;
}
if (codec.GetParam(cricket::kCodecParamPort, &value)) {
return rtc::FromString<int>(value);
}
}
return -1;
}
// 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) {
std::ostringstream oss;
oss << "Failed to set " << (source == cricket::CS_LOCAL ? "local" : "remote")
<< " " << SdpTypeToString(type) << " sdp: " << error.message();
return oss.str();
}
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;
}
} // 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 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> 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;
};
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 &&
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 &&
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;
}
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;
}
bool ConvertConstraintsToOfferAnswerOptions(
const MediaConstraintsInterface* constraints,
PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options) {
if (!constraints) {
return true;
}
bool value = false;
size_t mandatory_constraints_satisfied = 0;
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveAudio, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->offer_to_receive_audio =
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
kOfferToReceiveMediaTrue
: 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kOfferToReceiveVideo, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->offer_to_receive_video =
value ? PeerConnectionInterface::RTCOfferAnswerOptions::
kOfferToReceiveMediaTrue
: 0;
}
if (FindConstraint(constraints,
MediaConstraintsInterface::kVoiceActivityDetection, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->voice_activity_detection = value;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kUseRtpMux, &value,
&mandatory_constraints_satisfied)) {
offer_answer_options->use_rtp_mux = value;
}
if (FindConstraint(constraints, MediaConstraintsInterface::kIceRestart,
&value, &mandatory_constraints_satisfied)) {
offer_answer_options->ice_restart = value;
}
return mandatory_constraints_satisfied == constraints->GetMandatory().size();
}
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
std::unique_ptr<RtcEventLog> event_log,
std::unique_ptr<Call> call)
: factory_(factory),
event_log_(std::move(event_log)),
rtcp_cname_(GenerateRtcpCname()),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()),
call_(std::move(call)) {}
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 (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();
transport_controller_.reset();
// port_allocator_ lives on the network thread and should be destroyed there.
network_thread()->Invoke<void>(RTC_FROM_HERE,
[this] { port_allocator_.reset(); });
// call_ and event_log_ must be destroyed on the worker thread.
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
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 (auto transceiver : transceivers_) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
DestroyTransceiverChannel(transceiver);
}
}
for (auto transceiver : transceivers_) {
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
DestroyTransceiverChannel(transceiver);
}
}
DestroyDataChannel();
}
bool PeerConnection::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies) {
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.
if (!network_thread()->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::InitializePortAllocator_n, this,
stun_servers, turn_servers, configuration))) {
return false;
}
// 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 (port_allocator_flags_ & cricket::PORTALLOCATOR_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;
config.crypto_options = options.crypto_options;
config.transport_observer = this;
config.event_log = event_log_.get();
#if defined(ENABLE_EXTERNAL_AUTH)
config.enable_external_auth = true;
#endif
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
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_->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;
// 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];
}