pc/webrtcsession.cc

/*
 *  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/webrtcsession.h"

#include <limits.h>

#include <algorithm>
#include <set>
#include <utility>
#include <vector>

#include "api/call/audio_sink.h"
#include "api/jsepicecandidate.h"
#include "api/jsepsessiondescription.h"
#include "api/peerconnectioninterface.h"
#include "call/call.h"
#include "media/base/mediaconstants.h"
#include "media/sctp/sctptransportinternal.h"
#include "p2p/base/portallocator.h"
#include "pc/channel.h"
#include "pc/channelmanager.h"
#include "pc/mediasession.h"
#include "pc/sctputils.h"
#include "pc/webrtcsessiondescriptionfactory.h"
#include "rtc_base/basictypes.h"
#include "rtc_base/bind.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/stringencode.h"
#include "rtc_base/stringutils.h"

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 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 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;
}

// Verify that the order of media sections in |desc1| matches |desc2|. The
// number of m= sections could be different.
static bool MediaSectionsInSameOrder(const SessionDescription* desc1,
                                     const SessionDescription* desc2) {
  if (!desc1 || !desc2) {
    return false;
  }
  for (size_t i = 0;
       i < desc1->contents().size() && i < desc2->contents().size(); ++i) {
    if ((desc2->contents()[i].name) != desc1->contents()[i].name) {
      return false;
    }
    const MediaContentDescription* desc2_mdesc =
        static_cast<const MediaContentDescription*>(
            desc2->contents()[i].description);
    const MediaContentDescription* desc1_mdesc =
        static_cast<const MediaContentDescription*>(
            desc1->contents()[i].description);
    if (desc2_mdesc->type() != desc1_mdesc->type()) {
      return false;
    }
  }
  return true;
}

static bool MediaSectionsHaveSameCount(const SessionDescription* desc1,
                                       const SessionDescription* desc2) {
  if (!desc1 || !desc2) {
    return false;
  }
  return desc1->contents().size() == desc2->contents().size();
}

// 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 channels first since they may have a pointer to a voice
  // channel.
  for (auto* channel : video_channels_) {
    DestroyVideoChannel(channel);
  }
  for (auto* channel : voice_channels_) {
    DestroyVoiceChannel(channel);
  }
  if (rtp_data_channel_) {
    DestroyDataChannel();
  }
  if (sctp_transport_) {
    SignalDataChannelDestroyed();
    network_thread_->Invoke<void>(
        RTC_FROM_HERE, rtc::Bind(&WebRtcSession::DestroySctpTransport_n, this));
  }

  LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
}

void WebRtcSession::Initialize(
    const PeerConnectionFactoryInterface::Options& options,
    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
    const PeerConnectionInterface::RTCConfiguration& rtc_configuration,
    PeerConnection* pc) {
  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;
  } 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);

  // Whether the certificate generator/certificate is null or not determines
  // what WebRtcSessionDescriptionFactory will do, so make sure that we give it
  // the right instructions by clearing the variables if needed.
  if (!dtls_enabled_) {
    cert_generator.reset();
    certificate = nullptr;
  } else if (certificate) {
    // Favor generated certificate over the certificate generator.
    cert_generator.reset();
  }

  webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
      signaling_thread(), channel_manager_, pc, id(), std::move(cert_generator),
      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);
}

void WebRtcSession::Close() {
  SetState(STATE_CLOSED);
  RemoveUnusedChannels(nullptr);
  call_ = nullptr;
  RTC_DCHECK(voice_channels_.empty());
  RTC_DCHECK(video_channels_.empty());
  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;
}

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(
    std::unique_ptr<SessionDescriptionInterface> desc,
    std::string* err_desc) {
  RTC_DCHECK(signaling_thread()->IsCurrent());

  // Validate SDP.
  if (!ValidateSessionDescription(desc.get(), 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_ = std::move(desc);
    pending_local_description_ = nullptr;
    current_remote_description_ = std::move(pending_remote_description_);
  } else {
    pending_local_description_ = std::move(desc);
  }

  // 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(local_description()->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(local_description()->type(), GetSessionErrorMsg(),
                       err_desc);
  }
  return true;
}

bool WebRtcSession::SetRemoteDescription(
    std::unique_ptr<SessionDescriptionInterface> desc,
    std::string* err_desc) {
  RTC_DCHECK(signaling_thread()->IsCurrent());

  // Validate SDP.
  if (!ValidateSessionDescription(desc.get(), cricket::CS_REMOTE, err_desc)) {
    return false;
  }

  // Hold this pointer so candidates can be copied to it later in the method.
  SessionDescriptionInterface* desc_ptr = desc.get();

  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 as |old_remote_description|.
  std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
  Action action = GetAction(desc->type());
  if (action == kAnswer) {
    replaced_remote_description = pending_remote_description_
                                      ? std::move(pending_remote_description_)
                                      : std::move(current_remote_description_);
    current_remote_description_ = std::move(desc);
    pending_remote_description_ = nullptr;
    current_local_description_ = std::move(pending_local_description_);
  } else {
    replaced_remote_description = std::move(pending_remote_description_);
    pending_remote_description_ = std::move(desc);
  }

  // Transport and Media channels will be created only when offer is set.
  if (action == kOffer &&
      !CreateChannels(remote_description()->description())) {
    // TODO(mallinath) - Handle CreateChannel failure, as new local description
    // is applied. Restore back to old description.
    return BadRemoteSdp(remote_description()->type(), kCreateChannelFailed,
                        err_desc);
  }

  // Remove unused channels if MediaContentDescription is rejected.
  RemoveUnusedChannels(remote_description()->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(remote_description())) {
    return BadRemoteSdp(remote_description()->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, remote_description(),
                                   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_ptr);
      }
    }
  }

  if (error() != ERROR_NONE) {
    return BadRemoteSdp(remote_description()->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 (remote_description()->type() != SessionDescriptionInterface::kOffer &&
      ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew) {
    SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
  }
  return true;
}

// TODO(steveanton): Eventually it'd be nice to store the channels as a single
// vector of BaseChannel pointers instead of separate voice and video channel
// vectors. At that point, this will become a simple getter.
std::vector<cricket::BaseChannel*> WebRtcSession::Channels() const {
  std::vector<cricket::BaseChannel*> channels;
  channels.insert(channels.end(), voice_channels_.begin(),
                  voice_channels_.end());
  channels.insert(channels.end(), video_channels_.begin(),
                  video_channels_.end());
  if (rtp_data_channel_) {
    channels.push_back(rtp_data_channel_.get());
  }
  return channels;
}

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) {
  const SessionDescription* sdesc =
      (source == cricket::CS_LOCAL ? local_description() : remote_description())
          ->description();
  RTC_DCHECK(sdesc);
  bool all_success = true;
  for (auto* channel : Channels()) {
    // TODO(steveanton): Add support for multiple channels of the same type.
    const ContentInfo* content_info =
        cricket::GetFirstMediaContent(sdesc->contents(), channel->media_type());
    if (!content_info) {
      continue;
    }
    const MediaContentDescription* content_desc =
        static_cast<const MediaContentDescription*>(content_info->description);
    if (content_desc && !content_info->rejected) {
      bool success = (source == cricket::CS_LOCAL)
                         ? channel->SetLocalContent(content_desc, action, err)
                         : channel->SetRemoteContent(content_desc, action, err);
      if (!success) {
        all_success = false;
        break;
      }
    }
  }
  // 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())) {
    all_success &= network_thread_->Invoke<bool>(
        RTC_FROM_HERE,
        rtc::Bind(&WebRtcSession::PushdownSctpParameters_n, this, source));
  }
  return all_success;
}

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;

  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) channels.
void WebRtcSession::EnableChannels() {
  for (cricket::VoiceChannel* voice_channel : voice_channels_) {
    if (!voice_channel->enabled()) {
      voice_channel->Enable(true);
    }
  }

  for (cricket::VideoChannel* video_channel : video_channels_) {
    if (!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) {
  // TODO(steveanton): Add support for multiple audio/video channels.
  // 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()) {
    RemoveAndDestroyVideoChannel(video_channel());
  }

  const cricket::ContentInfo* voice_info = cricket::GetFirstAudioContent(desc);
  if ((!voice_info || voice_info->rejected) && voice_channel()) {
    RemoveAndDestroyVoiceChannel(voice_channel());
  }

  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));
    }
  }
}

// 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) {
  // TODO(steveanton): Add support for multiple audio/video channels.
  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) {
  // TODO(steveanton): Check to see if it's safe to create multiple voice
  // channels.
  RTC_DCHECK(voice_channels_.empty());

  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);
  }

  cricket::VoiceChannel* voice_channel = 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_RTCP);
    }
    return false;
  }

  voice_channels_.push_back(voice_channel);

  voice_channel->SignalRtcpMuxFullyActive.connect(
      this, &WebRtcSession::DestroyRtcpTransport_n);
  voice_channel->SignalDtlsSrtpSetupFailure.connect(
      this, &WebRtcSession::OnDtlsSrtpSetupFailure);

  // TODO(steveanton): This should signal which voice channel was created since
  // we can have multiple.
  SignalVoiceChannelCreated();
  voice_channel->SignalSentPacket.connect(this, &WebRtcSession::OnSentPacket_w);
  return true;
}

bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,
                                       const std::string* bundle_transport) {
  // TODO(steveanton): Check to see if it's safe to create multiple video
  // channels.
  RTC_DCHECK(video_channels_.empty());

  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);
  }

  cricket::VideoChannel* video_channel = 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_RTCP);
    }
    return false;
  }

  video_channels_.push_back(video_channel);

  video_channel->SignalRtcpMuxFullyActive.connect(
      this, &WebRtcSession::DestroyRtcpTransport_n);
  video_channel->SignalDtlsSrtpSetupFailure.connect(
      this, &WebRtcSession::OnDtlsSrtpSetupFailure);

  // TODO(steveanton): This should signal which video channel was created since
  // we can have multiple.
  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;
  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_RTCP);
      }
      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 || action == kPrAnswer) {
    const cricket::SessionDescription* offer_desc =
        (source == cricket::CS_LOCAL) ? remote_description()->description()
                                      : local_description()->description();
    if (!MediaSectionsHaveSameCount(sdesc->description(), offer_desc) ||
        !MediaSectionsInSameOrder(sdesc->description(), offer_desc)) {
      return BadAnswerSdp(source, kMlineMismatchInAnswer, err_desc);
    }
  } else {
    // The re-offers should respect the order of m= sections in current local
    // description. See RFC3264 Section 8 paragraph 4 for more details.
    if (local_description() &&
        !MediaSectionsInSameOrder(sdesc->description(),
                                  local_description()->description())) {
      return BadOfferSdp(source, kMlineMismatchInSubsequentOffer, 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) {
    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::RemoveAndDestroyVideoChannel(
    cricket::VideoChannel* video_channel) {
  auto it =
      std::find(video_channels_.begin(), video_channels_.end(), video_channel);
  RTC_DCHECK(it != video_channels_.end());
  if (it == video_channels_.end()) {
    return;
  }
  video_channels_.erase(it);
  DestroyVideoChannel(video_channel);
}

void WebRtcSession::DestroyVideoChannel(cricket::VideoChannel* video_channel) {
  // TODO(steveanton): This should take an identifier for the video channel
  // since we now support more than one.
  SignalVideoChannelDestroyed();
  RTC_DCHECK(video_channel->rtp_dtls_transport());
  const std::string transport_name =
      video_channel->rtp_dtls_transport()->transport_name();
  const bool need_to_delete_rtcp =
      (video_channel->rtcp_dtls_transport() != nullptr);
  // The above need to be cached before destroying the video channel so that we
  // do not access uninitialized memory.
  channel_manager_->DestroyVideoChannel(video_channel);
  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::RemoveAndDestroyVoiceChannel(
    cricket::VoiceChannel* voice_channel) {
  auto it =
      std::find(voice_channels_.begin(), voice_channels_.end(), voice_channel);
  RTC_DCHECK(it != voice_channels_.end());
  if (it == voice_channels_.end()) {
    return;
  }
  voice_channels_.erase(it);
  DestroyVoiceChannel(voice_channel);
}

void WebRtcSession::DestroyVoiceChannel(cricket::VoiceChannel* voice_channel) {
  // TODO(steveanton): This should take an identifier for the voice channel
  // since we now support more than one.
  SignalVoiceChannelDestroyed();
  RTC_DCHECK(voice_channel->rtp_dtls_transport());
  const std::string transport_name =
      voice_channel->rtp_dtls_transport()->transport_name();
  const bool need_to_delete_rtcp =
      (voice_channel->rtcp_dtls_transport() != nullptr);
  // The above need to be cached before destroying the video channel so that we
  // do not access uninitialized memory.
  channel_manager_->DestroyVoiceChannel(voice_channel);
  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