blob: d03f7568327fcaabafcaaa8452da44a1bf7bbef6 [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 "webrtc/pc/webrtcsession.h"
#include <limits.h>
#include <algorithm>
#include <set>
#include <utility>
#include <vector>
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/call/call.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/sctp/sctptransportinternal.h"
#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/pc/channel.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/mediasession.h"
#include "webrtc/pc/sctputils.h"
#include "webrtc/pc/webrtcsessiondescriptionfactory.h"
#include "webrtc/rtc_base/basictypes.h"
#include "webrtc/rtc_base/bind.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/helpers.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/stringencode.h"
#include "webrtc/rtc_base/stringutils.h"
#ifdef HAVE_QUIC
#include "webrtc/p2p/quic/quictransportchannel.h"
#endif // HAVE_QUIC
using cricket::ContentInfo;
using cricket::ContentInfos;
using cricket::MediaContentDescription;
using cricket::SessionDescription;
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 kCreateChannelFailed[] = "Failed to create channels.";
const char kInvalidCandidates[] = "Description contains invalid candidates.";
const char kInvalidSdp[] = "Invalid session description.";
const char kMlineMismatch[] =
"Offer and answer descriptions m-lines are not matching. Rejecting answer.";
const char kPushDownTDFailed[] =
"Failed to push down transport description:";
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.";
const char kEnableBundleFailed[] = "Failed to enable BUNDLE.";
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;
}
// Compares |answer| against |offer|. Comparision is done
// for number of m-lines in answer against offer. If matches true will be
// returned otherwise false.
static bool VerifyMediaDescriptions(
const SessionDescription* answer, const SessionDescription* offer) {
if (offer->contents().size() != answer->contents().size())
return false;
for (size_t i = 0; i < offer->contents().size(); ++i) {
if ((offer->contents()[i].name) != answer->contents()[i].name) {
return false;
}
const MediaContentDescription* offer_mdesc =
static_cast<const MediaContentDescription*>(
offer->contents()[i].description);
const MediaContentDescription* answer_mdesc =
static_cast<const MediaContentDescription*>(
answer->contents()[i].description);
if (offer_mdesc->type() != answer_mdesc->type()) {
return false;
}
}
return true;
}
// 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.
static bool VerifyCrypto(const SessionDescription* desc,
bool dtls_enabled,
std::string* error) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
const ContentInfos& contents = desc->contents();
for (size_t index = 0; index < contents.size(); ++index) {
const ContentInfo* cinfo = &contents[index];
if (cinfo->rejected) {
continue;
}
if (bundle && bundle->HasContentName(cinfo->name) &&
cinfo->name != *(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 =
static_cast<const MediaContentDescription*>(cinfo->description);
const TransportInfo* tinfo = desc->GetTransportInfoByName(cinfo->name);
if (!media || !tinfo) {
// Something is not right.
LOG(LS_ERROR) << kInvalidSdp;
*error = kInvalidSdp;
return false;
}
if (dtls_enabled) {
if (!tinfo->description.identity_fingerprint) {
LOG(LS_WARNING) <<
"Session description must have DTLS fingerprint if DTLS enabled.";
*error = kSdpWithoutDtlsFingerprint;
return false;
}
} else {
if (media->cryptos().empty()) {
LOG(LS_WARNING) <<
"Session description must have SDES when DTLS disabled.";
*error = kSdpWithoutSdesCrypto;
return false;
}
}
}
return true;
}
// 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.
static bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
const ContentInfos& contents = desc->contents();
for (size_t index = 0; index < contents.size(); ++index) {
const ContentInfo* cinfo = &contents[index];
if (cinfo->rejected) {
continue;
}
if (bundle && bundle->HasContentName(cinfo->name) &&
cinfo->name != *(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(cinfo->name);
if (!tinfo) {
// Something is not right.
LOG(LS_ERROR) << kInvalidSdp;
return false;
}
if (tinfo->description.ice_ufrag.empty() ||
tinfo->description.ice_pwd.empty()) {
LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
return false;
}
}
return true;
}
static bool GetTrackIdBySsrc(const SessionDescription* session_description,
uint32_t ssrc,
std::string* track_id) {
RTC_DCHECK(track_id != NULL);
const cricket::ContentInfo* audio_info =
cricket::GetFirstAudioContent(session_description);
if (audio_info) {
const cricket::MediaContentDescription* audio_content =
static_cast<const cricket::MediaContentDescription*>(
audio_info->description);
const auto* found =
cricket::GetStreamBySsrc(audio_content->streams(), ssrc);
if (found) {
*track_id = found->id;
return true;
}
}
const cricket::ContentInfo* video_info =
cricket::GetFirstVideoContent(session_description);
if (video_info) {
const cricket::MediaContentDescription* video_content =
static_cast<const cricket::MediaContentDescription*>(
video_info->description);
const auto* found =
cricket::GetStreamBySsrc(video_content->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.
static int GetSctpPort(const SessionDescription* session_description) {
const ContentInfo* content_info = GetFirstDataContent(session_description);
RTC_DCHECK(content_info);
if (!content_info) {
return -1;
}
const cricket::DataContentDescription* data =
static_cast<const cricket::DataContentDescription*>(
(content_info->description));
std::string value;
cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
cricket::kGoogleSctpDataCodecName);
for (const cricket::DataCodec& codec : data->codecs()) {
if (!codec.Matches(match_pattern)) {
continue;
}
if (codec.GetParam(cricket::kCodecParamPort, &value)) {
return rtc::FromString<int>(value);
}
}
return -1;
}
static bool BadSdp(const std::string& source,
const std::string& type,
const std::string& reason,
std::string* err_desc) {
std::ostringstream desc;
desc << "Failed to set " << source;
if (!type.empty()) {
desc << " " << type;
}
desc << " sdp: " << reason;
if (err_desc) {
*err_desc = desc.str();
}
LOG(LS_ERROR) << desc.str();
return false;
}
static bool BadSdp(cricket::ContentSource source,
const std::string& type,
const std::string& reason,
std::string* err_desc) {
if (source == cricket::CS_LOCAL) {
return BadSdp("local", type, reason, err_desc);
} else {
return BadSdp("remote", type, reason, err_desc);
}
}
static bool BadLocalSdp(const std::string& type,
const std::string& reason,
std::string* err_desc) {
return BadSdp(cricket::CS_LOCAL, type, reason, err_desc);
}
static bool BadRemoteSdp(const std::string& type,
const std::string& reason,
std::string* err_desc) {
return BadSdp(cricket::CS_REMOTE, type, reason, err_desc);
}
static bool BadOfferSdp(cricket::ContentSource source,
const std::string& reason,
std::string* err_desc) {
return BadSdp(source, SessionDescriptionInterface::kOffer, reason, err_desc);
}
static bool BadPranswerSdp(cricket::ContentSource source,
const std::string& reason,
std::string* err_desc) {
return BadSdp(source, SessionDescriptionInterface::kPrAnswer,
reason, err_desc);
}
static bool BadAnswerSdp(cricket::ContentSource source,
const std::string& reason,
std::string* err_desc) {
return BadSdp(source, SessionDescriptionInterface::kAnswer, reason, err_desc);
}
#define GET_STRING_OF_STATE(state) \
case webrtc::WebRtcSession::state: \
result = #state; \
break;
static std::string GetStateString(webrtc::WebRtcSession::State state) {
std::string result;
switch (state) {
GET_STRING_OF_STATE(STATE_INIT)
GET_STRING_OF_STATE(STATE_SENTOFFER)
GET_STRING_OF_STATE(STATE_RECEIVEDOFFER)
GET_STRING_OF_STATE(STATE_SENTPRANSWER)
GET_STRING_OF_STATE(STATE_RECEIVEDPRANSWER)
GET_STRING_OF_STATE(STATE_INPROGRESS)
GET_STRING_OF_STATE(STATE_CLOSED)
default:
RTC_NOTREACHED();
break;
}
return result;
}
#define GET_STRING_OF_ERROR_CODE(err) \
case webrtc::WebRtcSession::err: \
result = #err; \
break;
static std::string GetErrorCodeString(webrtc::WebRtcSession::Error err) {
std::string result;
switch (err) {
GET_STRING_OF_ERROR_CODE(ERROR_NONE)
GET_STRING_OF_ERROR_CODE(ERROR_CONTENT)
GET_STRING_OF_ERROR_CODE(ERROR_TRANSPORT)
default:
RTC_NOTREACHED();
break;
}
return result;
}
static std::string MakeErrorString(const std::string& error,
const std::string& desc) {
std::ostringstream ret;
ret << error << " " << desc;
return ret.str();
}
static std::string MakeTdErrorString(const std::string& desc) {
return MakeErrorString(kPushDownTDFailed, desc);
}
// 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)) {
LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
<< ".";
return true;
}
return false;
}
WebRtcSession::WebRtcSession(
Call* call,
cricket::ChannelManager* channel_manager,
const cricket::MediaConfig& media_config,
RtcEventLog* event_log,
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
cricket::PortAllocator* port_allocator,
std::unique_ptr<cricket::TransportController> transport_controller,
std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory)
: network_thread_(network_thread),
worker_thread_(worker_thread),
signaling_thread_(signaling_thread),
// 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 |sid_| is max limited to LLONG_MAX.
sid_(rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX)),
transport_controller_(std::move(transport_controller)),
sctp_factory_(std::move(sctp_factory)),
media_config_(media_config),
event_log_(event_log),
call_(call),
channel_manager_(channel_manager),
ice_observer_(NULL),
ice_connection_state_(PeerConnectionInterface::kIceConnectionNew),
ice_connection_receiving_(true),
older_version_remote_peer_(false),
dtls_enabled_(false),
data_channel_type_(cricket::DCT_NONE),
metrics_observer_(NULL) {
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
transport_controller_->SignalConnectionState.connect(
this, &WebRtcSession::OnTransportControllerConnectionState);
transport_controller_->SignalReceiving.connect(
this, &WebRtcSession::OnTransportControllerReceiving);
transport_controller_->SignalGatheringState.connect(
this, &WebRtcSession::OnTransportControllerGatheringState);
transport_controller_->SignalCandidatesGathered.connect(
this, &WebRtcSession::OnTransportControllerCandidatesGathered);
transport_controller_->SignalCandidatesRemoved.connect(
this, &WebRtcSession::OnTransportControllerCandidatesRemoved);
transport_controller_->SignalDtlsHandshakeError.connect(
this, &WebRtcSession::OnTransportControllerDtlsHandshakeError);
}
WebRtcSession::~WebRtcSession() {
RTC_DCHECK(signaling_thread()->IsCurrent());
// Destroy video_channel_ first since it may have a pointer to the
// voice_channel_.
if (video_channel_) {
DestroyVideoChannel();
}
if (voice_channel_) {
DestroyVoiceChannel();
}
if (rtp_data_channel_) {
DestroyDataChannel();
}
if (sctp_transport_) {
SignalDataChannelDestroyed();
network_thread_->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
}
#ifdef HAVE_QUIC
if (quic_data_transport_) {
quic_data_transport_.reset();
}
#endif
LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
}
bool WebRtcSession::Initialize(
const PeerConnectionFactoryInterface::Options& options,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
bundle_policy_ = rtc_configuration.bundle_policy;
rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
// Obtain a certificate from RTCConfiguration if any were provided (optional).
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
if (!rtc_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 = rtc_configuration.certificates[0];
}
SetIceConfig(ParseIceConfig(rtc_configuration));
if (options.disable_encryption) {
dtls_enabled_ = false;
} else {
// Enable DTLS by default if we have an identity store or a certificate.
dtls_enabled_ = (cert_generator || certificate);
// |rtc_configuration| can override the default |dtls_enabled_| value.
if (rtc_configuration.enable_dtls_srtp) {
dtls_enabled_ = *(rtc_configuration.enable_dtls_srtp);
}
}
// Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
// It takes precendence over the disable_sctp_data_channels
// PeerConnectionFactoryInterface::Options.
if (rtc_configuration.enable_rtp_data_channel) {
data_channel_type_ = cricket::DCT_RTP;
}
#ifdef HAVE_QUIC
else if (rtc_configuration.enable_quic) {
// Use QUIC instead of DTLS when |enable_quic| is true.
data_channel_type_ = cricket::DCT_QUIC;
transport_controller_->use_quic();
if (dtls_enabled_) {
LOG(LS_INFO) << "Using QUIC instead of DTLS";
}
quic_data_transport_.reset(
new QuicDataTransport(signaling_thread(), worker_thread(),
network_thread(), transport_controller_.get()));
}
#endif // HAVE_QUIC
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 =
rtc_configuration.screencast_min_bitrate;
audio_options_.combined_audio_video_bwe =
rtc_configuration.combined_audio_video_bwe;
audio_options_.audio_jitter_buffer_max_packets =
rtc::Optional<int>(rtc_configuration.audio_jitter_buffer_max_packets);
audio_options_.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(
rtc_configuration.audio_jitter_buffer_fast_accelerate);
if (!dtls_enabled_) {
// Construct with DTLS disabled.
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager_, this, id(),
std::unique_ptr<rtc::RTCCertificateGeneratorInterface>()));
} else {
// Construct with DTLS enabled.
if (!certificate) {
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager_, this, id(),
std::move(cert_generator)));
} else {
// Use the already generated certificate.
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager_, this, id(), certificate));
}
}
webrtc_session_desc_factory_->SignalCertificateReady.connect(
this, &WebRtcSession::OnCertificateReady);
if (options.disable_encryption) {
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
options.crypto_options.enable_encrypted_rtp_header_extensions);
return true;
}
void WebRtcSession::Close() {
SetState(STATE_CLOSED);
RemoveUnusedChannels(nullptr);
call_ = nullptr;
RTC_DCHECK(!voice_channel_);
RTC_DCHECK(!video_channel_);
RTC_DCHECK(!rtp_data_channel_);
RTC_DCHECK(!sctp_transport_);
}
cricket::BaseChannel* WebRtcSession::GetChannel(
const std::string& content_name) {
if (voice_channel() && voice_channel()->content_name() == content_name) {
return voice_channel();
}
if (video_channel() && video_channel()->content_name() == content_name) {
return video_channel();
}
if (rtp_data_channel() &&
rtp_data_channel()->content_name() == content_name) {
return rtp_data_channel();
}
return nullptr;
}
cricket::SecurePolicy WebRtcSession::SdesPolicy() const {
return webrtc_session_desc_factory_->SdesPolicy();
}
bool WebRtcSession::GetSctpSslRole(rtc::SSLRole* role) {
if (!local_description() || !remote_description()) {
LOG(LS_INFO) << "Local and Remote descriptions must be applied to get the "
<< "SSL Role of the SCTP transport.";
return false;
}
if (!sctp_transport_) {
LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
<< "SSL Role of the SCTP transport.";
return false;
}
return transport_controller_->GetSslRole(*sctp_transport_name_, role);
}
bool WebRtcSession::GetSslRole(const std::string& content_name,
rtc::SSLRole* role) {
if (!local_description() || !remote_description()) {
LOG(LS_INFO) << "Local and Remote descriptions must be applied to get the "
<< "SSL Role of the session.";
return false;
}
return transport_controller_->GetSslRole(GetTransportName(content_name),
role);
}
void WebRtcSession::CreateOffer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
const cricket::MediaSessionOptions& session_options) {
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
}
void WebRtcSession::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const cricket::MediaSessionOptions& session_options) {
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
}
bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
std::string* err_desc) {
RTC_DCHECK(signaling_thread()->IsCurrent());
// Takes the ownership of |desc| regardless of the result.
std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
// Validate SDP.
if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {
return false;
}
// Update the initial_offerer flag if this session is the initial_offerer.
Action action = GetAction(desc->type());
if (state() == STATE_INIT && action == kOffer) {
initial_offerer_ = true;
transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
}
if (action == kAnswer) {
current_local_description_.reset(desc_temp.release());
pending_local_description_.reset(nullptr);
current_remote_description_.reset(pending_remote_description_.release());
} else {
pending_local_description_.reset(desc_temp.release());
}
// Transport and Media channels will be created only when offer is set.
if (action == kOffer && !CreateChannels(local_description()->description())) {
// TODO(mallinath) - Handle CreateChannel failure, as new local description
// is applied. Restore back to old description.
return BadLocalSdp(desc->type(), kCreateChannelFailed, err_desc);
}
// Remove unused channels if MediaContentDescription is rejected.
RemoveUnusedChannels(local_description()->description());
if (!UpdateSessionState(action, cricket::CS_LOCAL, err_desc)) {
return false;
}
if (remote_description()) {
// Now that we have a local description, we can push down remote candidates.
UseCandidatesInSessionDescription(remote_description());
}
pending_ice_restarts_.clear();
if (error() != ERROR_NONE) {
return BadLocalSdp(desc->type(), GetSessionErrorMsg(), err_desc);
}
return true;
}
bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,
std::string* err_desc) {
RTC_DCHECK(signaling_thread()->IsCurrent());
// Takes the ownership of |desc| regardless of the result.
std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
// Validate SDP.
if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {
return false;
}
const SessionDescriptionInterface* old_remote_description =
remote_description();
// Grab ownership of the description being replaced for the remainder of this
// method, since it's used below.
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
Action action = GetAction(desc->type());
if (action == kAnswer) {
replaced_remote_description.reset(
pending_remote_description_ ? pending_remote_description_.release()
: current_remote_description_.release());
current_remote_description_.reset(desc_temp.release());
pending_remote_description_.reset(nullptr);
current_local_description_.reset(pending_local_description_.release());
} else {
replaced_remote_description.reset(pending_remote_description_.release());
pending_remote_description_.reset(desc_temp.release());
}
// Transport and Media channels will be created only when offer is set.
if (action == kOffer && !CreateChannels(desc->description())) {
// TODO(mallinath) - Handle CreateChannel failure, as new local description
// is applied. Restore back to old description.
return BadRemoteSdp(desc->type(), kCreateChannelFailed, err_desc);
}
// Remove unused channels if MediaContentDescription is rejected.
RemoveUnusedChannels(desc->description());
// NOTE: Candidates allocation will be initiated only when SetLocalDescription
// is called.
if (!UpdateSessionState(action, cricket::CS_REMOTE, err_desc)) {
return false;
}
if (local_description() && !UseCandidatesInSessionDescription(desc)) {
return BadRemoteSdp(desc->type(), kInvalidCandidates, err_desc);
}
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, desc,
content.name)) {
if (action == 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, desc);
}
}
}
if (error() != ERROR_NONE) {
return BadRemoteSdp(desc->type(), GetSessionErrorMsg(), err_desc);
}
// 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 (desc->type() != SessionDescriptionInterface::kOffer &&
ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew) {
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
}
return true;
}
void WebRtcSession::LogState(State old_state, State new_state) {
LOG(LS_INFO) << "Session:" << id()
<< " Old state:" << GetStateString(old_state)
<< " New state:" << GetStateString(new_state);
}
void WebRtcSession::SetState(State state) {
RTC_DCHECK(signaling_thread_->IsCurrent());
if (state != state_) {
LogState(state_, state);
state_ = state;
SignalState(this, state_);
}
}
void WebRtcSession::SetError(Error error, const std::string& error_desc) {
RTC_DCHECK(signaling_thread_->IsCurrent());
if (error != error_) {
error_ = error;
error_desc_ = error_desc;
}
}
bool WebRtcSession::UpdateSessionState(
Action action, cricket::ContentSource source,
std::string* err_desc) {
RTC_DCHECK(signaling_thread()->IsCurrent());
// 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(error() == ERROR_NONE);
std::string td_err;
if (action == kOffer) {
if (!PushdownTransportDescription(source, cricket::CA_OFFER, &td_err)) {
return BadOfferSdp(source, MakeTdErrorString(td_err), err_desc);
}
SetState(source == cricket::CS_LOCAL ? STATE_SENTOFFER
: STATE_RECEIVEDOFFER);
if (!PushdownMediaDescription(cricket::CA_OFFER, source, err_desc)) {
SetError(ERROR_CONTENT, *err_desc);
}
if (error() != ERROR_NONE) {
return BadOfferSdp(source, GetSessionErrorMsg(), err_desc);
}
} else if (action == kPrAnswer) {
if (!PushdownTransportDescription(source, cricket::CA_PRANSWER, &td_err)) {
return BadPranswerSdp(source, MakeTdErrorString(td_err), err_desc);
}
EnableChannels();
SetState(source == cricket::CS_LOCAL ? STATE_SENTPRANSWER
: STATE_RECEIVEDPRANSWER);
if (!PushdownMediaDescription(cricket::CA_PRANSWER, source, err_desc)) {
SetError(ERROR_CONTENT, *err_desc);
}
if (error() != ERROR_NONE) {
return BadPranswerSdp(source, GetSessionErrorMsg(), err_desc);
}
} else if (action == kAnswer) {
const cricket::ContentGroup* local_bundle =
local_description()->description()->GetGroupByName(
cricket::GROUP_TYPE_BUNDLE);
const cricket::ContentGroup* remote_bundle =
remote_description()->description()->GetGroupByName(
cricket::GROUP_TYPE_BUNDLE);
if (local_bundle && remote_bundle) {
// The answerer decides the transport to bundle on.
const cricket::ContentGroup* answer_bundle =
(source == cricket::CS_LOCAL ? local_bundle : remote_bundle);
if (!EnableBundle(*answer_bundle)) {
LOG(LS_WARNING) << "Failed to enable BUNDLE.";
return BadAnswerSdp(source, kEnableBundleFailed, err_desc);
}
}
// Only push down the transport description after enabling BUNDLE; we don't
// want to push down a description on a transport about to be destroyed.
if (!PushdownTransportDescription(source, cricket::CA_ANSWER, &td_err)) {
return BadAnswerSdp(source, MakeTdErrorString(td_err), err_desc);
}
EnableChannels();
SetState(STATE_INPROGRESS);
if (!PushdownMediaDescription(cricket::CA_ANSWER, source, err_desc)) {
SetError(ERROR_CONTENT, *err_desc);
}
if (error() != ERROR_NONE) {
return BadAnswerSdp(source, GetSessionErrorMsg(), err_desc);
}
}
return true;
}
WebRtcSession::Action WebRtcSession::GetAction(const std::string& type) {
if (type == SessionDescriptionInterface::kOffer) {
return WebRtcSession::kOffer;
} else if (type == SessionDescriptionInterface::kPrAnswer) {
return WebRtcSession::kPrAnswer;
} else if (type == SessionDescriptionInterface::kAnswer) {
return WebRtcSession::kAnswer;
}
RTC_NOTREACHED() << "unknown action type";
return WebRtcSession::kOffer;
}
bool WebRtcSession::PushdownMediaDescription(
cricket::ContentAction action,
cricket::ContentSource source,
std::string* err) {
auto set_content = [this, action, source, err](cricket::BaseChannel* ch) {
if (!ch) {
return true;
} else if (source == cricket::CS_LOCAL) {
return ch->PushdownLocalDescription(local_description()->description(),
action, err);
} else {
return ch->PushdownRemoteDescription(remote_description()->description(),
action, err);
}
};
bool ret = (set_content(voice_channel()) && set_content(video_channel()) &&
set_content(rtp_data_channel()));
// 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() &&
cricket::GetFirstDataContent(local_description()->description()) &&
cricket::GetFirstDataContent(remote_description()->description())) {
ret &= network_thread_->Invoke<bool>(
RTC_FROM_HERE,
rtc::Bind(&WebRtcSession::PushdownSctpParameters_n, this, source));
}
return ret;
}
bool WebRtcSession::PushdownSctpParameters_n(cricket::ContentSource source) {
RTC_DCHECK(network_thread_->IsCurrent());
RTC_DCHECK(local_description());
RTC_DCHECK(remote_description());
// Apply the SCTP port (which is hidden inside a DataCodec structure...)
// When we support "max-message-size", that would also be pushed down here.
return sctp_transport_->Start(
GetSctpPort(local_description()->description()),
GetSctpPort(remote_description()->description()));
}
bool WebRtcSession::PushdownTransportDescription(cricket::ContentSource source,
cricket::ContentAction action,
std::string* error_desc) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (source == cricket::CS_LOCAL) {
return PushdownLocalTransportDescription(local_description()->description(),
action, error_desc);
}
return PushdownRemoteTransportDescription(remote_description()->description(),
action, error_desc);
}
bool WebRtcSession::PushdownLocalTransportDescription(
const SessionDescription* sdesc,
cricket::ContentAction action,
std::string* err) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (!sdesc) {
return false;
}
for (const TransportInfo& tinfo : sdesc->transport_infos()) {
if (!transport_controller_->SetLocalTransportDescription(
tinfo.content_name, tinfo.description, action, err)) {
return false;
}
}
return true;
}
bool WebRtcSession::PushdownRemoteTransportDescription(
const SessionDescription* sdesc,
cricket::ContentAction action,
std::string* err) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (!sdesc) {
return false;
}
for (const TransportInfo& tinfo : sdesc->transport_infos()) {
if (!transport_controller_->SetRemoteTransportDescription(
tinfo.content_name, tinfo.description, action, err)) {
return false;
}
}
return true;
}
bool WebRtcSession::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;
}
bool WebRtcSession::EnableBundle(const cricket::ContentGroup& bundle) {
const std::string* first_content_name = bundle.FirstContentName();
if (!first_content_name) {
LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
return false;
}
const std::string& transport_name = *first_content_name;
#ifdef HAVE_QUIC
if (quic_data_transport_ &&
bundle.HasContentName(quic_data_transport_->content_name()) &&
quic_data_transport_->transport_name() != transport_name) {
LOG(LS_ERROR) << "Unable to BUNDLE " << quic_data_transport_->content_name()
<< " on " << transport_name << "with QUIC.";
}
#endif
auto maybe_set_transport = [this, bundle,
transport_name](cricket::BaseChannel* ch) {
if (!ch || !bundle.HasContentName(ch->content_name())) {
return true;
}
std::string old_transport_name = ch->transport_name();
if (old_transport_name == transport_name) {
LOG(LS_INFO) << "BUNDLE already enabled for " << ch->content_name()
<< " on " << transport_name << ".";
return true;
}
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
bool need_rtcp = (ch->rtcp_dtls_transport() != nullptr);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (need_rtcp) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
ch->SetTransports(rtp_dtls_transport, rtcp_dtls_transport);
LOG(LS_INFO) << "Enabled BUNDLE for " << ch->content_name() << " on "
<< transport_name << ".";
transport_controller_->DestroyDtlsTransport(
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
// If the channel needs rtcp, it means that the channel used to have a
// rtcp transport which needs to be deleted now.
if (need_rtcp) {
transport_controller_->DestroyDtlsTransport(
old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
return true;
};
if (!maybe_set_transport(voice_channel()) ||
!maybe_set_transport(video_channel()) ||
!maybe_set_transport(rtp_data_channel())) {
return false;
}
// For SCTP, transport creation/deletion happens here instead of in the
// object itself.
if (sctp_transport_) {
RTC_DCHECK(sctp_transport_name_);
RTC_DCHECK(sctp_content_name_);
if (transport_name != *sctp_transport_name_ &&
bundle.HasContentName(*sctp_content_name_)) {
network_thread_->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::ChangeSctpTransport_n, this,
transport_name));
}
}
return true;
}
bool WebRtcSession::ProcessIceMessage(const IceCandidateInterface* candidate) {
if (!remote_description()) {
LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "
<< "without any remote session description.";
return false;
}
if (!candidate) {
LOG(LS_ERROR) << "ProcessIceMessage: Candidate is NULL.";
return false;
}
bool valid = false;
bool ready = ReadyToUseRemoteCandidate(candidate, NULL, &valid);
if (!valid) {
return false;
}
// Add this candidate to the remote session description.
if (!mutable_remote_description()->AddCandidate(candidate)) {
LOG(LS_ERROR) << "ProcessIceMessage: Candidate cannot be used.";
return false;
}
if (ready) {
return UseCandidate(candidate);
} else {
LOG(LS_INFO) << "ProcessIceMessage: Not ready to use candidate.";
return true;
}
}
bool WebRtcSession::RemoveRemoteIceCandidates(
const std::vector<cricket::Candidate>& candidates) {
if (!remote_description()) {
LOG(LS_ERROR) << "RemoveRemoteIceCandidates: ICE candidates can't be "
<< "removed without any remote session description.";
return false;
}
if (candidates.empty()) {
LOG(LS_ERROR) << "RemoveRemoteIceCandidates: candidates are empty.";
return false;
}
size_t number_removed =
mutable_remote_description()->RemoveCandidates(candidates);
if (number_removed != candidates.size()) {
LOG(LS_ERROR) << "RemoveRemoteIceCandidates: Failed to remove candidates. "
<< "Requested " << candidates.size() << " but only "
<< number_removed << " are removed.";
}
// Remove the candidates from the transport controller.
std::string error;
bool res = transport_controller_->RemoveRemoteCandidates(candidates, &error);
if (!res && !error.empty()) {
LOG(LS_ERROR) << "Error when removing remote candidates: " << error;
}
return true;
}
cricket::IceConfig WebRtcSession::ParseIceConfig(
const PeerConnectionInterface::RTCConfiguration& config) const {
cricket::ContinualGatheringPolicy gathering_policy;
// TODO(honghaiz): Add the third continual gathering policy in
// PeerConnectionInterface and map it to GATHER_CONTINUALLY_AND_RECOVER.
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 = 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 =
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_min_interval = config.ice_check_min_interval;
ice_config.regather_all_networks_interval_range =
config.ice_regather_interval_range;
return ice_config;
}
void WebRtcSession::SetIceConfig(const cricket::IceConfig& config) {
transport_controller_->SetIceConfig(config);
}
void WebRtcSession::MaybeStartGathering() {
transport_controller_->MaybeStartGathering();
}
bool WebRtcSession::GetLocalTrackIdBySsrc(uint32_t ssrc,
std::string* track_id) {
if (!local_description()) {
return false;
}
return webrtc::GetTrackIdBySsrc(local_description()->description(), ssrc,
track_id);
}
bool WebRtcSession::GetRemoteTrackIdBySsrc(uint32_t ssrc,
std::string* track_id) {
if (!remote_description()) {
return false;
}
return webrtc::GetTrackIdBySsrc(remote_description()->description(), ssrc,
track_id);
}
std::string WebRtcSession::BadStateErrMsg(State state) {
std::ostringstream desc;
desc << "Called in wrong state: " << GetStateString(state);
return desc.str();
}
bool WebRtcSession::SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) {
if (!rtp_data_channel_ && !sctp_transport_) {
LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
<< "and sctp_transport_ are NULL.";
return false;
}
return rtp_data_channel_
? rtp_data_channel_->SendData(params, payload, result)
: network_thread_->Invoke<bool>(
RTC_FROM_HERE,
Bind(&cricket::SctpTransportInternal::SendData,
sctp_transport_.get(), params, payload, result));
}
bool WebRtcSession::ConnectDataChannel(DataChannel* webrtc_data_channel) {
if (!rtp_data_channel_ && !sctp_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 (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);
SignalSctpStreamClosedRemotely.connect(
webrtc_data_channel, &DataChannel::OnStreamClosedRemotely);
}
return true;
}
void WebRtcSession::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
if (!rtp_data_channel_ && !sctp_transport_) {
LOG(LS_ERROR) << "DisconnectDataChannel called when rtp_data_channel_ and "
"sctp_transport_ are NULL.";
return;
}
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);
SignalSctpStreamClosedRemotely.disconnect(webrtc_data_channel);
}
}
void WebRtcSession::AddSctpDataStream(int sid) {
if (!sctp_transport_) {
LOG(LS_ERROR) << "AddSctpDataStream called when sctp_transport_ is NULL.";
return;
}
network_thread_->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream,
sctp_transport_.get(), sid));
}
void WebRtcSession::RemoveSctpDataStream(int sid) {
if (!sctp_transport_) {
LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
<< "NULL.";
return;
}
network_thread_->Invoke<void>(
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream,
sctp_transport_.get(), sid));
}
bool WebRtcSession::ReadyToSendData() const {
return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
sctp_ready_to_send_data_;
}
std::unique_ptr<SessionStats> WebRtcSession::GetStats_s() {
RTC_DCHECK(signaling_thread()->IsCurrent());
ChannelNamePairs channel_name_pairs;
if (voice_channel()) {
channel_name_pairs.voice = rtc::Optional<ChannelNamePair>(ChannelNamePair(
voice_channel()->content_name(), voice_channel()->transport_name()));
}
if (video_channel()) {
channel_name_pairs.video = rtc::Optional<ChannelNamePair>(ChannelNamePair(
video_channel()->content_name(), video_channel()->transport_name()));
}
if (rtp_data_channel()) {
channel_name_pairs.data = rtc::Optional<ChannelNamePair>(
ChannelNamePair(rtp_data_channel()->content_name(),
rtp_data_channel()->transport_name()));
}
if (sctp_transport_) {
RTC_DCHECK(sctp_content_name_);
RTC_DCHECK(sctp_transport_name_);
channel_name_pairs.data = rtc::Optional<ChannelNamePair>(
ChannelNamePair(*sctp_content_name_, *sctp_transport_name_));
}
return GetStats(channel_name_pairs);
}
std::unique_ptr<SessionStats> WebRtcSession::GetStats(
const ChannelNamePairs& channel_name_pairs) {
if (network_thread()->IsCurrent()) {
return GetStats_n(channel_name_pairs);
}
return network_thread()->Invoke<std::unique_ptr<SessionStats>>(
RTC_FROM_HERE,
rtc::Bind(&WebRtcSession::GetStats_n, this, channel_name_pairs));
}
bool WebRtcSession::GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
return transport_controller_->GetLocalCertificate(transport_name,
certificate);
}
std::unique_ptr<rtc::SSLCertificate> WebRtcSession::GetRemoteSSLCertificate(
const std::string& transport_name) {
return transport_controller_->GetRemoteSSLCertificate(transport_name);
}
cricket::DataChannelType WebRtcSession::data_channel_type() const {
return data_channel_type_;
}
bool WebRtcSession::IceRestartPending(const std::string& content_name) const {
return pending_ice_restarts_.find(content_name) !=
pending_ice_restarts_.end();
}
void WebRtcSession::SetNeedsIceRestartFlag() {
transport_controller_->SetNeedsIceRestartFlag();
}
bool WebRtcSession::NeedsIceRestart(const std::string& content_name) const {
return transport_controller_->NeedsIceRestart(content_name);
}
void WebRtcSession::OnCertificateReady(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
transport_controller_->SetLocalCertificate(certificate);
}
void WebRtcSession::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) {
SetError(ERROR_TRANSPORT,
rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp);
}
bool WebRtcSession::waiting_for_certificate_for_testing() const {
return webrtc_session_desc_factory_->waiting_for_certificate_for_testing();
}
const rtc::scoped_refptr<rtc::RTCCertificate>&
WebRtcSession::certificate_for_testing() {
return transport_controller_->certificate_for_testing();
}
void WebRtcSession::SetIceConnectionState(
PeerConnectionInterface::IceConnectionState state) {
if (ice_connection_state_ == state) {
return;
}
LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
<< " => " << state;
RTC_DCHECK(ice_connection_state_ !=
PeerConnectionInterface::kIceConnectionClosed);
ice_connection_state_ = state;
if (ice_observer_) {
ice_observer_->OnIceConnectionStateChange(ice_connection_state_);
}
}
void WebRtcSession::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:
LOG(LS_INFO) << "Changing to ICE connected state because "
<< "all transports are writable.";
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
break;
case cricket::kIceConnectionCompleted:
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);
if (metrics_observer_) {
ReportTransportStats();
}
break;
default:
RTC_NOTREACHED();
}
}
void WebRtcSession::OnTransportControllerReceiving(bool receiving) {
SetIceConnectionReceiving(receiving);
}
void WebRtcSession::SetIceConnectionReceiving(bool receiving) {
if (ice_connection_receiving_ == receiving) {
return;
}
ice_connection_receiving_ = receiving;
if (ice_observer_) {
ice_observer_->OnIceConnectionReceivingChange(receiving);
}
}
void WebRtcSession::OnTransportControllerCandidatesGathered(
const std::string& transport_name,
const cricket::Candidates& candidates) {
RTC_DCHECK(signaling_thread()->IsCurrent());
int sdp_mline_index;
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
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());
}
if (ice_observer_) {
ice_observer_->OnIceCandidate(std::move(candidate));
}
}
}
void WebRtcSession::OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
RTC_DCHECK(signaling_thread()->IsCurrent());
// Sanity check.
for (const cricket::Candidate& candidate : candidates) {
if (candidate.transport_name().empty()) {
LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
<< "empty content name in candidate "
<< candidate.ToString();
return;
}
}
if (local_description()) {
mutable_local_description()->RemoveCandidates(candidates);
}
if (ice_observer_) {
ice_observer_->OnIceCandidatesRemoved(candidates);
}
}
void WebRtcSession::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
if (metrics_observer_) {
metrics_observer_->IncrementEnumCounter(
webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
}
}
// Enabling voice and video (and RTP data) channel.
void WebRtcSession::EnableChannels() {
if (voice_channel_ && !voice_channel_->enabled())
voice_channel_->Enable(true);
if (video_channel_ && !video_channel_->enabled())
video_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 WebRtcSession::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 WebRtcSession::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) {
LOG(LS_INFO) << "UseCandidatesInSessionDescription: Not ready to use "
<< "candidate.";
}
continue;
}
ret = UseCandidate(candidate);
if (!ret) {
break;
}
}
}
return ret;
}
bool WebRtcSession::UseCandidate(const IceCandidateInterface* candidate) {
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
size_t remote_content_size =
remote_description()->description()->contents().size();
if (mediacontent_index >= remote_content_size) {
LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index.";
return false;
}
cricket::ContentInfo content =
remote_description()->description()->contents()[mediacontent_index];
std::vector<cricket::Candidate> candidates;
candidates.push_back(candidate->candidate());
// Invoking BaseSession method to handle remote candidates.
std::string error;
if (transport_controller_->AddRemoteCandidates(content.name, candidates,
&error)) {
// 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 {
if (!error.empty()) {
LOG(LS_WARNING) << error;
}
}
return true;
}
void WebRtcSession::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) && video_channel_) {
DestroyVideoChannel();
}
const cricket::ContentInfo* voice_info =
cricket::GetFirstAudioContent(desc);
if ((!voice_info || voice_info->rejected) && voice_channel_) {
DestroyVoiceChannel();
}
const cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(desc);
if (!data_info || data_info->rejected) {
if (rtp_data_channel_) {
DestroyDataChannel();
}
if (sctp_transport_) {
SignalDataChannelDestroyed();
network_thread_->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
}
#ifdef HAVE_QUIC
// Clean up the existing QuicDataTransport and its QuicTransportChannels.
if (quic_data_transport_) {
quic_data_transport_.reset();
}
#endif
}
}
// Returns the name of the transport channel when BUNDLE is enabled, or nullptr
// if the channel is not part of any bundle.
const std::string* WebRtcSession::GetBundleTransportName(
const cricket::ContentInfo* content,
const cricket::ContentGroup* bundle) {
if (!bundle) {
return nullptr;
}
const std::string* first_content_name = bundle->FirstContentName();
if (!first_content_name) {
LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
return nullptr;
}
if (!bundle->HasContentName(content->name)) {
LOG(LS_WARNING) << content->name << " is not part of any bundle group";
return nullptr;
}
LOG(LS_INFO) << "Bundling " << content->name << " on " << *first_content_name;
return first_content_name;
}
bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
const cricket::ContentGroup* bundle_group = nullptr;
if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {
bundle_group = desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
if (!bundle_group) {
LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";
return false;
}
}
// Creating the media channels and transport proxies.
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
if (voice && !voice->rejected && !voice_channel_) {
if (!CreateVoiceChannel(voice,
GetBundleTransportName(voice, bundle_group))) {
LOG(LS_ERROR) << "Failed to create voice channel.";
return false;
}
}
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
if (video && !video->rejected && !video_channel_) {
if (!CreateVideoChannel(video,
GetBundleTransportName(video, bundle_group))) {
LOG(LS_ERROR) << "Failed to create video channel.";
return false;
}
}
const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);
if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
!rtp_data_channel_ && !sctp_transport_) {
if (!CreateDataChannel(data, GetBundleTransportName(data, bundle_group))) {
LOG(LS_ERROR) << "Failed to create data channel.";
return false;
}
}
return true;
}
bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content,
const std::string* bundle_transport) {
bool require_rtcp_mux =
rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
std::string transport_name =
bundle_transport ? *bundle_transport : content->name;
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (!require_rtcp_mux) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
voice_channel_.reset(channel_manager_->CreateVoiceChannel(
call_, media_config_, rtp_dtls_transport, rtcp_dtls_transport,
transport_controller_->signaling_thread(), content->name, SrtpRequired(),
audio_options_));
if (!voice_channel_) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (rtcp_dtls_transport) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
}
return false;
}
voice_channel_->SignalRtcpMuxFullyActive.connect(
this, &WebRtcSession::DestroyRtcpTransport_n);
voice_channel_->SignalDtlsSrtpSetupFailure.connect(
this, &WebRtcSession::OnDtlsSrtpSetupFailure);
SignalVoiceChannelCreated();
voice_channel_->SignalSentPacket.connect(this,
&WebRtcSession::OnSentPacket_w);
return true;
}
bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
const std::string* bundle_transport) {
bool require_rtcp_mux =
rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
std::string transport_name =
bundle_transport ? *bundle_transport : content->name;
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (!require_rtcp_mux) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
video_channel_.reset(channel_manager_->CreateVideoChannel(
call_, media_config_, rtp_dtls_transport, rtcp_dtls_transport,
transport_controller_->signaling_thread(), content->name, SrtpRequired(),
video_options_));
if (!video_channel_) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (rtcp_dtls_transport) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
}
return false;
}
video_channel_->SignalRtcpMuxFullyActive.connect(
this, &WebRtcSession::DestroyRtcpTransport_n);
video_channel_->SignalDtlsSrtpSetupFailure.connect(
this, &WebRtcSession::OnDtlsSrtpSetupFailure);
SignalVideoChannelCreated();
video_channel_->SignalSentPacket.connect(this,
&WebRtcSession::OnSentPacket_w);
return true;
}
bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content,
const std::string* bundle_transport) {
const std::string transport_name =
bundle_transport ? *bundle_transport : content->name;
#ifdef HAVE_QUIC
if (data_channel_type_ == cricket::DCT_QUIC) {
RTC_DCHECK(transport_controller_->quic());
quic_data_transport_->SetTransports(transport_name);
return true;
}
#endif // HAVE_QUIC
bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
if (sctp) {
if (!sctp_factory_) {
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(&WebRtcSession::CreateSctpTransport_n,
this, content->name, transport_name))) {
return false;
};
} else {
bool require_rtcp_mux =
rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;
std::string transport_name =
bundle_transport ? *bundle_transport : content->name;
cricket::DtlsTransportInternal* rtp_dtls_transport =
transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
if (!require_rtcp_mux) {
rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
rtp_data_channel_.reset(channel_manager_->CreateRtpDataChannel(
media_config_, rtp_dtls_transport, rtcp_dtls_transport,
transport_controller_->signaling_thread(), content->name,
SrtpRequired()));
if (!rtp_data_channel_) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (rtcp_dtls_transport) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
}
return false;
}
rtp_data_channel_->SignalRtcpMuxFullyActive.connect(
this, &WebRtcSession::DestroyRtcpTransport_n);
rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
this, &WebRtcSession::OnDtlsSrtpSetupFailure);
rtp_data_channel_->SignalSentPacket.connect(this,
&WebRtcSession::OnSentPacket_w);
}
SignalDataChannelCreated();
return true;
}
Call::Stats WebRtcSession::GetCallStats() {
if (!worker_thread()->IsCurrent()) {
return worker_thread()->Invoke<Call::Stats>(
RTC_FROM_HERE, rtc::Bind(&WebRtcSession::GetCallStats, this));
}
if (!call_)
return Call::Stats();
return call_->GetStats();
}
std::unique_ptr<SessionStats> WebRtcSession::GetStats_n(
const ChannelNamePairs& channel_name_pairs) {
RTC_DCHECK(network_thread()->IsCurrent());
std::unique_ptr<SessionStats> session_stats(new SessionStats());
for (const auto channel_name_pair : { &channel_name_pairs.voice,
&channel_name_pairs.video,
&channel_name_pairs.data }) {
if (*channel_name_pair) {
cricket::TransportStats transport_stats;
if (!transport_controller_->GetStats((*channel_name_pair)->transport_name,
&transport_stats)) {
return nullptr;
}
session_stats->proxy_to_transport[(*channel_name_pair)->content_name] =
(*channel_name_pair)->transport_name;
session_stats->transport_stats[(*channel_name_pair)->transport_name] =
std::move(transport_stats);
}
}
return session_stats;
}
bool WebRtcSession::CreateSctpTransport_n(const std::string& content_name,
const std::string& transport_name) {
RTC_DCHECK(network_thread_->IsCurrent());
RTC_DCHECK(sctp_factory_);
cricket::DtlsTransportInternal* tc =
transport_controller_->CreateDtlsTransport_n(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
sctp_transport_ = sctp_factory_->CreateSctpTransport(tc);
RTC_DCHECK(sctp_transport_);
sctp_invoker_.reset(new rtc::AsyncInvoker());
sctp_transport_->SignalReadyToSendData.connect(
this, &WebRtcSession::OnSctpTransportReadyToSendData_n);
sctp_transport_->SignalDataReceived.connect(
this, &WebRtcSession::OnSctpTransportDataReceived_n);
sctp_transport_->SignalStreamClosedRemotely.connect(
this, &WebRtcSession::OnSctpStreamClosedRemotely_n);
sctp_transport_name_ = rtc::Optional<std::string>(transport_name);
sctp_content_name_ = rtc::Optional<std::string>(content_name);
return true;
}
void WebRtcSession::ChangeSctpTransport_n(const std::string& transport_name) {
RTC_DCHECK(network_thread_->IsCurrent());
RTC_DCHECK(sctp_transport_);
RTC_DCHECK(sctp_transport_name_);
std::string old_sctp_transport_name = *sctp_transport_name_;
sctp_transport_name_ = rtc::Optional<std::string>(transport_name);
cricket::DtlsTransportInternal* tc =
transport_controller_->CreateDtlsTransport_n(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
sctp_transport_->SetTransportChannel(tc);
transport_controller_->DestroyDtlsTransport_n(
old_sctp_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
}
void WebRtcSession::DestroySctpTransport_n() {
RTC_DCHECK(network_thread_->IsCurrent());
sctp_transport_.reset(nullptr);
sctp_content_name_.reset();
sctp_transport_name_.reset();
sctp_invoker_.reset(nullptr);
sctp_ready_to_send_data_ = false;
}
void WebRtcSession::OnSctpTransportReadyToSendData_n() {
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
RTC_DCHECK(network_thread_->IsCurrent());
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread_,
rtc::Bind(&WebRtcSession::OnSctpTransportReadyToSendData_s, this, true));
}
void WebRtcSession::OnSctpTransportReadyToSendData_s(bool ready) {
RTC_DCHECK(signaling_thread_->IsCurrent());
sctp_ready_to_send_data_ = ready;
SignalSctpReadyToSendData(ready);
}
void WebRtcSession::OnSctpTransportDataReceived_n(
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
RTC_DCHECK(network_thread_->IsCurrent());
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread_,
rtc::Bind(&WebRtcSession::OnSctpTransportDataReceived_s, this, params,
payload));
}
void WebRtcSession::OnSctpTransportDataReceived_s(
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK(signaling_thread_->IsCurrent());
if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
// 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(payload, &label, &config)) {
LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
<< params.ssrc;
return;
}
config.open_handshake_role = InternalDataChannelInit::kAcker;
SignalDataChannelOpenMessage(label, config);
} else {
// Otherwise just forward the signal.
SignalSctpDataReceived(params, payload);
}
}
void WebRtcSession::OnSctpStreamClosedRemotely_n(int sid) {
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
RTC_DCHECK(network_thread_->IsCurrent());
sctp_invoker_->AsyncInvoke<void>(
RTC_FROM_HERE, signaling_thread_,
rtc::Bind(&sigslot::signal1<int>::operator(),
&SignalSctpStreamClosedRemotely, sid));
}
// Returns false if bundle is enabled and rtcp_mux is disabled.
bool WebRtcSession::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 == cricket::NS_JINGLE_RTP) {
if (!HasRtcpMuxEnabled(content))
return false;
}
}
// RTCP-MUX is enabled in all the contents.
return true;
}
bool WebRtcSession::HasRtcpMuxEnabled(
const cricket::ContentInfo* content) {
const cricket::MediaContentDescription* description =
static_cast<cricket::MediaContentDescription*>(content->description);
return description->rtcp_mux();
}
bool WebRtcSession::ValidateSessionDescription(
const SessionDescriptionInterface* sdesc,
cricket::ContentSource source, std::string* err_desc) {
std::string type;
if (error() != ERROR_NONE) {
return BadSdp(source, type, GetSessionErrorMsg(), err_desc);
}
if (!sdesc || !sdesc->description()) {
return BadSdp(source, type, kInvalidSdp, err_desc);
}
type = sdesc->type();
Action action = GetAction(sdesc->type());
if (source == cricket::CS_LOCAL) {
if (!ExpectSetLocalDescription(action))
return BadLocalSdp(type, BadStateErrMsg(state()), err_desc);
} else {
if (!ExpectSetRemoteDescription(action))
return BadRemoteSdp(type, BadStateErrMsg(state()), err_desc);
}
// Verify crypto settings.
std::string crypto_error;
if ((webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
dtls_enabled_) &&
!VerifyCrypto(sdesc->description(), dtls_enabled_, &crypto_error)) {
return BadSdp(source, type, crypto_error, err_desc);
}
// Verify ice-ufrag and ice-pwd.
if (!VerifyIceUfragPwdPresent(sdesc->description())) {
return BadSdp(source, type, kSdpWithoutIceUfragPwd, err_desc);
}
if (!ValidateBundleSettings(sdesc->description())) {
return BadSdp(source, type, kBundleWithoutRtcpMux, err_desc);
}
// 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 (action == kAnswer) {
const cricket::SessionDescription* offer_desc =
(source == cricket::CS_LOCAL) ? remote_description()->description()
: local_description()->description();
if (!VerifyMediaDescriptions(sdesc->description(), offer_desc)) {
return BadAnswerSdp(source, kMlineMismatch, err_desc);
}
}
return true;
}
bool WebRtcSession::ExpectSetLocalDescription(Action action) {
return ((action == kOffer && state() == STATE_INIT) ||
// update local offer
(action == kOffer && state() == STATE_SENTOFFER) ||
// update the current ongoing session.
(action == kOffer && state() == STATE_INPROGRESS) ||
// accept remote offer
(action == kAnswer && state() == STATE_RECEIVEDOFFER) ||
(action == kAnswer && state() == STATE_SENTPRANSWER) ||
(action == kPrAnswer && state() == STATE_RECEIVEDOFFER) ||
(action == kPrAnswer && state() == STATE_SENTPRANSWER));
}
bool WebRtcSession::ExpectSetRemoteDescription(Action action) {
return ((action == kOffer && state() == STATE_INIT) ||
// update remote offer
(action == kOffer && state() == STATE_RECEIVEDOFFER) ||
// update the current ongoing session
(action == kOffer && state() == STATE_INPROGRESS) ||
// accept local offer
(action == kAnswer && state() == STATE_SENTOFFER) ||
(action == kAnswer && state() == STATE_RECEIVEDPRANSWER) ||
(action == kPrAnswer && state() == STATE_SENTOFFER) ||
(action == kPrAnswer && state() == STATE_RECEIVEDPRANSWER));
}
std::string WebRtcSession::GetSessionErrorMsg() {
std::ostringstream desc;
desc << kSessionError << GetErrorCodeString(error()) << ". ";
desc << kSessionErrorDesc << error_desc() << ".";
return desc.str();
}
// 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 WebRtcSession::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;
}
size_t mediacontent_index =
static_cast<size_t>(candidate->sdp_mline_index());
size_t remote_content_size =
current_remote_desc->description()->contents().size();
if (mediacontent_index >= remote_content_size) {
LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate media index "
<< mediacontent_index;
*valid = false;
return false;
}
cricket::ContentInfo content =
current_remote_desc->description()->contents()[mediacontent_index];
const std::string transport_name = GetTransportName(content.name);
if (transport_name.empty()) {
return false;
}
return transport_controller_->ReadyForRemoteCandidates(transport_name);
}
bool WebRtcSession::SrtpRequired() const {
return dtls_enabled_ ||
webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED;
}
void WebRtcSession::OnTransportControllerGatheringState(
cricket::IceGatheringState state) {
RTC_DCHECK(signaling_thread()->IsCurrent());
if (state == cricket::kIceGatheringGathering) {
if (ice_observer_) {
ice_observer_->OnIceGatheringChange(
PeerConnectionInterface::kIceGatheringGathering);
}
} else if (state == cricket::kIceGatheringComplete) {
if (ice_observer_) {
ice_observer_->OnIceGatheringChange(
PeerConnectionInterface::kIceGatheringComplete);
}
}
}
void WebRtcSession::ReportTransportStats() {
// Use a set so we don't report the same stats twice if two channels share
// a transport.
std::set<std::string> transport_names;
if (voice_channel()) {
transport_names.insert(voice_channel()->transport_name());
}
if (video_channel()) {
transport_names.insert(video_channel()->transport_name());
}
if (rtp_data_channel()) {
transport_names.insert(rtp_data_channel()->transport_name());
}
if (sctp_transport_name_) {
transport_names.insert(*sctp_transport_name_);
}
for (const auto& name : transport_names) {
cricket::TransportStats stats;
if (transport_controller_->GetStats(name, &stats)) {
ReportBestConnectionState(stats);
ReportNegotiatedCiphers(stats);
}
}
}
// Walk through the ConnectionInfos to gather best connection usage
// for IPv4 and IPv6.
void WebRtcSession::ReportBestConnectionState(
const cricket::TransportStats& stats) {
RTC_DCHECK(metrics_observer_ != NULL);
for (cricket::TransportChannelStatsList::const_iterator it =
stats.channel_stats.begin();
it != stats.channel_stats.end(); ++it) {
for (cricket::ConnectionInfos::const_iterator it_info =
it->connection_infos.begin();
it_info != it->connection_infos.end(); ++it_info) {
if (!it_info->best_connection) {
continue;
}
PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
const cricket::Candidate& local = it_info->local_candidate;
const cricket::Candidate& remote = it_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)) {
type = kEnumCounterIceCandidatePairTypeTcp;
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
type = kEnumCounterIceCandidatePairTypeUdp;
} else {
RTC_CHECK(0);
}
metrics_observer_->IncrementEnumCounter(
type, GetIceCandidatePairCounter(local, remote),
kIceCandidatePairMax);
// Increment the counter for IP type.
if (local.address().family() == AF_INET) {
metrics_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kBestConnections_IPv4,
kPeerConnectionAddressFamilyCounter_Max);
} else if (local.address().family() == AF_INET6) {
metrics_observer_->IncrementEnumCounter(
kEnumCounterAddressFamily, kBestConnections_IPv6,
kPeerConnectionAddressFamilyCounter_Max);
} else {
RTC_CHECK(0);
}
return;
}
}
}
void WebRtcSession::ReportNegotiatedCiphers(
const cricket::TransportStats& stats) {
RTC_DCHECK(metrics_observer_ != NULL);
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;
}
PeerConnectionEnumCounterType srtp_counter_type;
PeerConnectionEnumCounterType ssl_counter_type;
if (stats.transport_name == cricket::CN_AUDIO) {
srtp_counter_type = kEnumCounterAudioSrtpCipher;
ssl_counter_type = kEnumCounterAudioSslCipher;
} else if (stats.transport_name == cricket::CN_VIDEO) {
srtp_counter_type = kEnumCounterVideoSrtpCipher;
ssl_counter_type = kEnumCounterVideoSslCipher;
} else if (stats.transport_name == cricket::CN_DATA) {
srtp_counter_type = kEnumCounterDataSrtpCipher;
ssl_counter_type = kEnumCounterDataSslCipher;
} else {
RTC_NOTREACHED();
return;
}
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
metrics_observer_->IncrementSparseEnumCounter(srtp_counter_type,
srtp_crypto_suite);
}
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
metrics_observer_->IncrementSparseEnumCounter(ssl_counter_type,
ssl_cipher_suite);
}
}
void WebRtcSession::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
RTC_DCHECK(worker_thread()->IsCurrent());
RTC_DCHECK(call_);
call_->OnSentPacket(sent_packet);
}
const std::string WebRtcSession::GetTransportName(
const std::string& content_name) {
cricket::BaseChannel* channel = GetChannel(content_name);
if (!channel) {
#ifdef HAVE_QUIC
if (data_channel_type_ == cricket::DCT_QUIC && quic_data_transport_ &&
content_name == quic_data_transport_->transport_name()) {
return quic_data_transport_->transport_name();
}
#endif
if (sctp_transport_) {
RTC_DCHECK(sctp_content_name_);
RTC_DCHECK(sctp_transport_name_);
if (content_name == *sctp_content_name_) {
return *sctp_transport_name_;
}
}
// Return an empty string if failed to retrieve the transport name.
return "";
}
return channel->transport_name();
}
void WebRtcSession::DestroyRtcpTransport_n(const std::string& transport_name) {
RTC_DCHECK(network_thread()->IsCurrent());
transport_controller_->DestroyDtlsTransport_n(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
void WebRtcSession::DestroyVideoChannel() {
SignalVideoChannelDestroyed();
RTC_DCHECK(video_channel_->rtp_dtls_transport());
std::string transport_name;
transport_name = video_channel_->rtp_dtls_transport()->transport_name();
bool need_to_delete_rtcp = (video_channel_->rtcp_dtls_transport() != nullptr);
channel_manager_->DestroyVideoChannel(video_channel_.release());
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (need_to_delete_rtcp) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
}
void WebRtcSession::DestroyVoiceChannel() {
SignalVoiceChannelDestroyed();
RTC_DCHECK(voice_channel_->rtp_dtls_transport());
std::string transport_name;
transport_name = voice_channel_->rtp_dtls_transport()->transport_name();
bool need_to_delete_rtcp = (voice_channel_->rtcp_dtls_transport() != nullptr);
channel_manager_->DestroyVoiceChannel(voice_channel_.release());
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (need_to_delete_rtcp) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
}
void WebRtcSession::DestroyDataChannel() {
SignalDataChannelDestroyed();
RTC_DCHECK(rtp_data_channel_->rtp_dtls_transport());
std::string transport_name;
transport_name = rtp_data_channel_->rtp_dtls_transport()->transport_name();
bool need_to_delete_rtcp =
(rtp_data_channel_->rtcp_dtls_transport() != nullptr);
channel_manager_->DestroyRtpDataChannel(rtp_data_channel_.release());
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
if (need_to_delete_rtcp) {
transport_controller_->DestroyDtlsTransport(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
}
}
} // namespace webrtc