zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
db0cd9e774dbe91f41fea349fef995c2b5887c25
platform-external-webrtc/webrtc/api/webrtcsession.cc
Taylor Brandstetter db0cd9e774 Adding getParameters/setParameters APIs to RtpReceiver. 
This is similar to how a "receive" method is used to apply
RtpParameters to an RtpReceiver in ORTC. Currently, SetParameters
doesn't allow changing the parameters, so the main use of the API is
to retrieve the set of configured codecs. But other uses will likely
be made possible in the future.

R=glaznev@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1917193008 .

Cr-Commit-Position: refs/heads/master@{#12761}
2016-05-16 18:40:38 +00:00

2174 lines
75 KiB
C++
Raw Blame History

/*
* 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/api/webrtcsession.h"
#include <limits.h>
#include <algorithm>
#include <set>
#include <utility>
#include <vector>
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/sctputils.h"
#include "webrtc/api/webrtcsessiondescriptionfactory.h"
#include "webrtc/audio_sink.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/videocapturer.h"
#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/pc/channel.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/mediasession.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 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 kDtlsSetupFailureRtp[] =
"Couldn't set up DTLS-SRTP on RTP channel.";
const char kDtlsSetupFailureRtcp[] =
"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. Mismatches, such as replying with a DTLS fingerprint to SDES
// keys, will be caught in Transport negotiation, and backstopped by Channel's
// |secure_required| check.
static bool VerifyCrypto(const SessionDescription* desc,
bool dtls_enabled,
std::string* error) {
const ContentInfos& contents = desc->contents();
for (size_t index = 0; index < contents.size(); ++index) {
const ContentInfo* cinfo = &contents[index];
if (cinfo->rejected) {
continue;
}
// If the content isn't rejected, 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.
static bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
const ContentInfos& contents = desc->contents();
for (size_t index = 0; index < contents.size(); ++index) {
const ContentInfo* cinfo = &contents[index];
if (cinfo->rejected) {
continue;
}
// If the content isn't rejected, 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;
}
// Forces |sdesc->crypto_required| to the appropriate state based on the
// current security policy, to ensure a failure occurs if there is an error
// in crypto negotiation.
// Called when processing the local session description.
static void UpdateSessionDescriptionSecurePolicy(cricket::CryptoType type,
SessionDescription* sdesc) {
if (!sdesc) {
return;
}
// Updating the |crypto_required_| in MediaContentDescription to the
// appropriate state based on the current security policy.
for (cricket::ContentInfos::iterator iter = sdesc->contents().begin();
iter != sdesc->contents().end(); ++iter) {
if (cricket::IsMediaContent(&*iter)) {
MediaContentDescription* mdesc =
static_cast<MediaContentDescription*> (iter->description);
if (mdesc) {
mdesc->set_crypto_required(type);
}
}
}
}
static bool GetAudioSsrcByTrackId(const SessionDescription* session_description,
const std::string& track_id,
uint32_t* ssrc) {
const cricket::ContentInfo* audio_info =
cricket::GetFirstAudioContent(session_description);
if (!audio_info) {
LOG(LS_ERROR) << "Audio not used in this call";
return false;
}
const cricket::MediaContentDescription* audio_content =
static_cast<const cricket::MediaContentDescription*>(
audio_info->description);
const cricket::StreamParams* stream =
cricket::GetStreamByIds(audio_content->streams(), "", track_id);
if (!stream) {
return false;
}
*ssrc = stream->first_ssrc();
return true;
}
static bool GetTrackIdBySsrc(const SessionDescription* session_description,
uint32_t ssrc,
std::string* track_id) {
ASSERT(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;
}
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:
ASSERT(false);
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_DCHECK(false);
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(webrtc::MediaControllerInterface* media_controller,
rtc::Thread* signaling_thread,
rtc::Thread* worker_thread,
cricket::PortAllocator* port_allocator)
: signaling_thread_(signaling_thread),
worker_thread_(worker_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_(new cricket::TransportController(signaling_thread,
worker_thread,
port_allocator)),
media_controller_(media_controller),
channel_manager_(media_controller_->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);
}
WebRtcSession::~WebRtcSession() {
ASSERT(signaling_thread()->IsCurrent());
// Destroy video_channel_ first since it may have a pointer to the
// voice_channel_.
if (video_channel_) {
SignalVideoChannelDestroyed();
channel_manager_->DestroyVideoChannel(video_channel_.release());
}
if (voice_channel_) {
SignalVoiceChannelDestroyed();
channel_manager_->DestroyVoiceChannel(voice_channel_.release());
}
if (data_channel_) {
SignalDataChannelDestroyed();
channel_manager_->DestroyDataChannel(data_channel_.release());
}
SignalDestroyed();
LOG(LS_INFO) << "Session: " << id() << " is destroyed.";
}
bool WebRtcSession::Initialize(
const PeerConnectionFactoryInterface::Options& options,
std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
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_ = (dtls_identity_store || 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);
if (!dtls_enabled_) {
// Construct with DTLS disabled.
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager_, this, id()));
} else {
// Construct with DTLS enabled.
if (!certificate) {
// Use the |dtls_identity_store| to generate a certificate.
RTC_DCHECK(dtls_identity_store);
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager_, std::move(dtls_identity_store),
this, id()));
} else {
// Use the already generated certificate.
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
signaling_thread(), channel_manager_, certificate, this, id()));
}
}
webrtc_session_desc_factory_->SignalCertificateReady.connect(
this, &WebRtcSession::OnCertificateReady);
if (options.disable_encryption) {
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
return true;
}
void WebRtcSession::Close() {
SetState(STATE_CLOSED);
RemoveUnusedChannels(nullptr);
ASSERT(!voice_channel_);
ASSERT(!video_channel_);
ASSERT(!data_channel_);
media_controller_->Close();
}
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 (data_channel() && data_channel()->content_name() == content_name) {
return data_channel();
}
return nullptr;
}
void WebRtcSession::SetSdesPolicy(cricket::SecurePolicy secure_policy) {
webrtc_session_desc_factory_->SetSdesPolicy(secure_policy);
}
cricket::SecurePolicy WebRtcSession::SdesPolicy() const {
return webrtc_session_desc_factory_->SdesPolicy();
}
bool WebRtcSession::GetSslRole(const std::string& transport_name,
rtc::SSLRole* role) {
if (!local_desc_ || !remote_desc_) {
LOG(LS_INFO) << "Local and Remote descriptions must be applied to get "
<< "SSL Role of the session.";
return false;
}
return transport_controller_->GetSslRole(transport_name, role);
}
bool WebRtcSession::GetSslRole(const cricket::BaseChannel* channel,
rtc::SSLRole* role) {
return channel && GetSslRole(channel->transport_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) {
ASSERT(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);
}
cricket::SecurePolicy sdes_policy =
webrtc_session_desc_factory_->SdesPolicy();
cricket::CryptoType crypto_required = dtls_enabled_ ?
cricket::CT_DTLS : (sdes_policy == cricket::SEC_REQUIRED ?
cricket::CT_SDES : cricket::CT_NONE);
// Update the MediaContentDescription crypto settings as per the policy set.
UpdateSessionDescriptionSecurePolicy(crypto_required, desc->description());
local_desc_.reset(desc_temp.release());
// Transport and Media channels will be created only when offer is set.
if (action == kOffer && !CreateChannels(local_desc_->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_desc_->description());
if (!UpdateSessionState(action, cricket::CS_LOCAL, err_desc)) {
return false;
}
if (remote_desc_) {
// Now that we have a local description, we can push down remote candidates.
UseCandidatesInSessionDescription(remote_desc_.get());
}
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) {
ASSERT(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;
}
std::unique_ptr<SessionDescriptionInterface> old_remote_desc(
remote_desc_.release());
remote_desc_.reset(desc_temp.release());
// Transport and Media channels will be created only when offer is set.
Action action = GetAction(desc->type());
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_desc_ && !UseCandidatesInSessionDescription(desc)) {
return BadRemoteSdp(desc->type(), kInvalidCandidates, err_desc);
}
if (old_remote_desc) {
for (const cricket::ContentInfo& content :
old_remote_desc->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_desc.get(), 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_desc.get(), 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) {
ASSERT(signaling_thread_->IsCurrent());
if (state != state_) {
LogState(state_, state);
state_ = state;
SignalState(this, state_);
}
}
void WebRtcSession::SetError(Error error, const std::string& error_desc) {
ASSERT(signaling_thread_->IsCurrent());
if (error != error_) {
error_ = error;
error_desc_ = error_desc;
}
}
bool WebRtcSession::UpdateSessionState(
Action action, cricket::ContentSource source,
std::string* err_desc) {
ASSERT(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!
ASSERT(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_desc_->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
const cricket::ContentGroup* remote_bundle =
remote_desc_->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;
}
ASSERT(false && "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_desc_->description(), action,
err);
} else {
return ch->PushdownRemoteDescription(remote_desc_->description(), action,
err);
}
};
return (set_content(voice_channel()) &&
set_content(video_channel()) &&
set_content(data_channel()));
}
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_desc_->description(), action,
error_desc);
}
return PushdownRemoteTransportDescription(remote_desc_->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::GetTransportStats(SessionStats* stats) {
ASSERT(signaling_thread()->IsCurrent());
return (GetChannelTransportStats(voice_channel(), stats) &&
GetChannelTransportStats(video_channel(), stats) &&
GetChannelTransportStats(data_channel(), stats));
}
bool WebRtcSession::GetChannelTransportStats(cricket::BaseChannel* ch,
SessionStats* stats) {
ASSERT(signaling_thread()->IsCurrent());
if (!ch) {
// Not using this channel.
return true;
}
const std::string& content_name = ch->content_name();
const std::string& transport_name = ch->transport_name();
stats->proxy_to_transport[content_name] = transport_name;
if (stats->transport_stats.find(transport_name) !=
stats->transport_stats.end()) {
// Transport stats already done for this transport.
return true;
}
cricket::TransportStats tstats;
if (!transport_controller_->GetStats(transport_name, &tstats)) {
return false;
}
stats->transport_stats[transport_name] = tstats;
return true;
}
bool WebRtcSession::GetLocalCertificate(
const std::string& transport_name,
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
ASSERT(signaling_thread()->IsCurrent());
return transport_controller_->GetLocalCertificate(transport_name,
certificate);
}
std::unique_ptr<rtc::SSLCertificate> WebRtcSession::GetRemoteSSLCertificate(
const std::string& transport_name) {
ASSERT(signaling_thread()->IsCurrent());
return transport_controller_->GetRemoteSSLCertificate(transport_name);
}
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;
cricket::BaseChannel* first_channel = GetChannel(transport_name);
auto maybe_set_transport = [this, bundle, transport_name,
first_channel](cricket::BaseChannel* ch) {
if (!ch || !bundle.HasContentName(ch->content_name())) {
return true;
}
if (ch->transport_name() == transport_name) {
LOG(LS_INFO) << "BUNDLE already enabled for " << ch->content_name()
<< " on " << transport_name << ".";
return true;
}
if (!ch->SetTransport(transport_name)) {
LOG(LS_WARNING) << "Failed to enable BUNDLE for " << ch->content_name();
return false;
}
LOG(LS_INFO) << "Enabled BUNDLE for " << ch->content_name() << " on "
<< transport_name << ".";
return true;
};
if (!maybe_set_transport(voice_channel()) ||
!maybe_set_transport(video_channel()) ||
!maybe_set_transport(data_channel())) {
return false;
}
return true;
}
bool WebRtcSession::ProcessIceMessage(const IceCandidateInterface* candidate) {
if (!remote_desc_) {
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 (!remote_desc_->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_desc_) {
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 = remote_desc_->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::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.gather_continually = (config.continual_gathering_policy ==
PeerConnectionInterface::GATHER_CONTINUALLY);
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_desc_) {
return false;
}
return webrtc::GetTrackIdBySsrc(local_desc_->description(), ssrc, track_id);
}
bool WebRtcSession::GetRemoteTrackIdBySsrc(uint32_t ssrc,
std::string* track_id) {
if (!remote_desc_) {
return false;
}
return webrtc::GetTrackIdBySsrc(remote_desc_->description(), ssrc, track_id);
}
std::string WebRtcSession::BadStateErrMsg(State state) {
std::ostringstream desc;
desc << "Called in wrong state: " << GetStateString(state);
return desc.str();
}
void WebRtcSession::SetAudioPlayout(uint32_t ssrc, bool enable) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_) {
LOG(LS_ERROR) << "SetAudioPlayout: No audio channel exists.";
return;
}
if (!voice_channel_->SetOutputVolume(ssrc, enable ? 1 : 0)) {
// Allow that SetOutputVolume fail if |enable| is false but assert
// otherwise. This in the normal case when the underlying media channel has
// already been deleted.
ASSERT(enable == false);
}
}
void WebRtcSession::SetAudioSend(uint32_t ssrc,
bool enable,
const cricket::AudioOptions& options,
cricket::AudioSource* source) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_) {
LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
return;
}
if (!voice_channel_->SetAudioSend(ssrc, enable, &options, source)) {
LOG(LS_ERROR) << "SetAudioSend: ssrc is incorrect: " << ssrc;
}
}
void WebRtcSession::SetAudioPlayoutVolume(uint32_t ssrc, double volume) {
ASSERT(signaling_thread()->IsCurrent());
ASSERT(volume >= 0 && volume <= 10);
if (!voice_channel_) {
LOG(LS_ERROR) << "SetAudioPlayoutVolume: No audio channel exists.";
return;
}
if (!voice_channel_->SetOutputVolume(ssrc, volume)) {
ASSERT(false);
}
}
void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
std::unique_ptr<AudioSinkInterface> sink) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_)
return;
voice_channel_->SetRawAudioSink(ssrc, std::move(sink));
}
RtpParameters WebRtcSession::GetAudioRtpSendParameters(uint32_t ssrc) const {
ASSERT(signaling_thread()->IsCurrent());
if (voice_channel_) {
return voice_channel_->GetRtpSendParameters(ssrc);
}
return RtpParameters();
}
bool WebRtcSession::SetAudioRtpSendParameters(uint32_t ssrc,
const RtpParameters& parameters) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_) {
return false;
}
return voice_channel_->SetRtpSendParameters(ssrc, parameters);
}
RtpParameters WebRtcSession::GetAudioRtpReceiveParameters(uint32_t ssrc) const {
ASSERT(signaling_thread()->IsCurrent());
if (voice_channel_) {
return voice_channel_->GetRtpReceiveParameters(ssrc);
}
return RtpParameters();
}
bool WebRtcSession::SetAudioRtpReceiveParameters(
uint32_t ssrc,
const RtpParameters& parameters) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_) {
return false;
}
return voice_channel_->SetRtpReceiveParameters(ssrc, parameters);
}
bool WebRtcSession::SetSource(
uint32_t ssrc,
rtc::VideoSourceInterface<cricket::VideoFrame>* source) {
ASSERT(signaling_thread()->IsCurrent());
if (!video_channel_) {
// |video_channel_| doesnt't exist. Probably because the remote end doesnt't
// support video.
LOG(LS_WARNING) << "Video not used in this call.";
return false;
}
video_channel_->SetSource(ssrc, source);
return true;
}
void WebRtcSession::SetVideoPlayout(
uint32_t ssrc,
bool enable,
rtc::VideoSinkInterface<cricket::VideoFrame>* sink) {
ASSERT(signaling_thread()->IsCurrent());
if (!video_channel_) {
LOG(LS_WARNING) << "SetVideoPlayout: No video channel exists.";
return;
}
if (!video_channel_->SetSink(ssrc, enable ? sink : NULL)) {
// Allow that SetSink fail if |sink| is NULL but assert otherwise.
// This in the normal case when the underlying media channel has already
// been deleted.
ASSERT(sink == NULL);
}
}
void WebRtcSession::SetVideoSend(uint32_t ssrc,
bool enable,
const cricket::VideoOptions* options) {
ASSERT(signaling_thread()->IsCurrent());
if (!video_channel_) {
LOG(LS_WARNING) << "SetVideoSend: No video channel exists.";
return;
}
if (!video_channel_->SetVideoSend(ssrc, enable, options)) {
// Allow that MuteStream fail if |enable| is false but assert otherwise.
// This in the normal case when the underlying media channel has already
// been deleted.
ASSERT(enable == false);
}
}
RtpParameters WebRtcSession::GetVideoRtpSendParameters(uint32_t ssrc) const {
ASSERT(signaling_thread()->IsCurrent());
if (video_channel_) {
return video_channel_->GetRtpSendParameters(ssrc);
}
return RtpParameters();
}
bool WebRtcSession::SetVideoRtpSendParameters(uint32_t ssrc,
const RtpParameters& parameters) {
ASSERT(signaling_thread()->IsCurrent());
if (!video_channel_) {
return false;
}
return video_channel_->SetRtpSendParameters(ssrc, parameters);
}
RtpParameters WebRtcSession::GetVideoRtpReceiveParameters(uint32_t ssrc) const {
ASSERT(signaling_thread()->IsCurrent());
if (video_channel_) {
return video_channel_->GetRtpReceiveParameters(ssrc);
}
return RtpParameters();
}
bool WebRtcSession::SetVideoRtpReceiveParameters(
uint32_t ssrc,
const RtpParameters& parameters) {
ASSERT(signaling_thread()->IsCurrent());
if (!video_channel_) {
return false;
}
return video_channel_->SetRtpReceiveParameters(ssrc, parameters);
}
bool WebRtcSession::CanInsertDtmf(const std::string& track_id) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_) {
LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
return false;
}
uint32_t send_ssrc = 0;
// The Dtmf is negotiated per channel not ssrc, so we only check if the ssrc
// exists.
if (!local_desc_ ||
!GetAudioSsrcByTrackId(local_desc_->description(), track_id,
&send_ssrc)) {
LOG(LS_ERROR) << "CanInsertDtmf: Track does not exist: " << track_id;
return false;
}
return voice_channel_->CanInsertDtmf();
}
bool WebRtcSession::InsertDtmf(const std::string& track_id,
int code, int duration) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_) {
LOG(LS_ERROR) << "InsertDtmf: No audio channel exists.";
return false;
}
uint32_t send_ssrc = 0;
if (!VERIFY(local_desc_ && GetAudioSsrcByTrackId(local_desc_->description(),
track_id, &send_ssrc))) {
LOG(LS_ERROR) << "InsertDtmf: Track does not exist: " << track_id;
return false;
}
if (!voice_channel_->InsertDtmf(send_ssrc, code, duration)) {
LOG(LS_ERROR) << "Failed to insert DTMF to channel.";
return false;
}
return true;
}
sigslot::signal0<>* WebRtcSession::GetOnDestroyedSignal() {
return &SignalDestroyed;
}
bool WebRtcSession::SendData(const cricket::SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
cricket::SendDataResult* result) {
if (!data_channel_) {
LOG(LS_ERROR) << "SendData called when data_channel_ is NULL.";
return false;
}
return data_channel_->SendData(params, payload, result);
}
bool WebRtcSession::ConnectDataChannel(DataChannel* webrtc_data_channel) {
if (!data_channel_) {
LOG(LS_ERROR) << "ConnectDataChannel called when data_channel_ is NULL.";
return false;
}
data_channel_->SignalReadyToSendData.connect(webrtc_data_channel,
&DataChannel::OnChannelReady);
data_channel_->SignalDataReceived.connect(webrtc_data_channel,
&DataChannel::OnDataReceived);
data_channel_->SignalStreamClosedRemotely.connect(
webrtc_data_channel, &DataChannel::OnStreamClosedRemotely);
return true;
}
void WebRtcSession::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
if (!data_channel_) {
LOG(LS_ERROR) << "DisconnectDataChannel called when data_channel_ is NULL.";
return;
}
data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
data_channel_->SignalStreamClosedRemotely.disconnect(webrtc_data_channel);
}
void WebRtcSession::AddSctpDataStream(int sid) {
if (!data_channel_) {
LOG(LS_ERROR) << "AddDataChannelStreams called when data_channel_ is NULL.";
return;
}
data_channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(sid));
data_channel_->AddSendStream(cricket::StreamParams::CreateLegacy(sid));
}
void WebRtcSession::RemoveSctpDataStream(int sid) {
if (!data_channel_) {
LOG(LS_ERROR) << "RemoveDataChannelStreams called when data_channel_ is "
<< "NULL.";
return;
}
data_channel_->RemoveRecvStream(sid);
data_channel_->RemoveSendStream(sid);
}
bool WebRtcSession::ReadyToSendData() const {
return data_channel_ && data_channel_->ready_to_send_data();
}
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::OnCertificateReady(
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
transport_controller_->SetLocalCertificate(certificate);
}
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;
}
// ASSERT that the requested transition is allowed. Note that
// WebRtcSession does not implement "kIceConnectionClosed" (that is handled
// within PeerConnection). This switch statement should compile away when
// ASSERTs are disabled.
LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
<< " => " << state;
switch (ice_connection_state_) {
case PeerConnectionInterface::kIceConnectionNew:
ASSERT(state == PeerConnectionInterface::kIceConnectionChecking);
break;
case PeerConnectionInterface::kIceConnectionChecking:
ASSERT(state == PeerConnectionInterface::kIceConnectionFailed ||
state == PeerConnectionInterface::kIceConnectionConnected);
break;
case PeerConnectionInterface::kIceConnectionConnected:
ASSERT(state == PeerConnectionInterface::kIceConnectionDisconnected ||
state == PeerConnectionInterface::kIceConnectionChecking ||
state == PeerConnectionInterface::kIceConnectionCompleted);
break;
case PeerConnectionInterface::kIceConnectionCompleted:
ASSERT(state == PeerConnectionInterface::kIceConnectionConnected ||
state == PeerConnectionInterface::kIceConnectionDisconnected);
break;
case PeerConnectionInterface::kIceConnectionFailed:
ASSERT(state == PeerConnectionInterface::kIceConnectionNew);
break;
case PeerConnectionInterface::kIceConnectionDisconnected:
ASSERT(state == PeerConnectionInterface::kIceConnectionChecking ||
state == PeerConnectionInterface::kIceConnectionConnected ||
state == PeerConnectionInterface::kIceConnectionCompleted ||
state == PeerConnectionInterface::kIceConnectionFailed);
break;
case PeerConnectionInterface::kIceConnectionClosed:
ASSERT(false);
break;
default:
ASSERT(false);
break;
}
ice_connection_state_ = state;
if (ice_observer_) {
ice_observer_->OnIceConnectionChange(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:
ASSERT(false);
}
}
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) {
ASSERT(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.
JsepIceCandidate candidate(transport_name, sdp_mline_index, *citer);
if (ice_observer_) {
ice_observer_->OnIceCandidate(&candidate);
}
if (local_desc_) {
local_desc_->AddCandidate(&candidate);
}
}
}
void WebRtcSession::OnTransportControllerCandidatesRemoved(
const std::vector<cricket::Candidate>& candidates) {
ASSERT(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_desc_) {
local_desc_->RemoveCandidates(candidates);
}
if (ice_observer_) {
ice_observer_->OnIceCandidatesRemoved(candidates);
}
}
// Enabling voice and video 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 (data_channel_ && !data_channel_->enabled())
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_desc_ || !sdp_mline_index) {
return false;
}
bool content_found = false;
const ContentInfos& contents = local_desc_->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_desc_->description()->contents().size();
if (mediacontent_index >= remote_content_size) {
LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index.";
return false;
}
cricket::ContentInfo content =
remote_desc_->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_) {
SignalVideoChannelDestroyed();
channel_manager_->DestroyVideoChannel(video_channel_.release());
}
const cricket::ContentInfo* voice_info =
cricket::GetFirstAudioContent(desc);
if ((!voice_info || voice_info->rejected) && voice_channel_) {
SignalVoiceChannelDestroyed();
channel_manager_->DestroyVoiceChannel(voice_channel_.release());
}
const cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(desc);
if ((!data_info || data_info->rejected) && data_channel_) {
SignalDataChannelDestroyed();
channel_manager_->DestroyDataChannel(data_channel_.release());
}
}
// TODO(mallinath) - Add a correct error code if the channels are not created
// due to BUNDLE is enabled but rtcp-mux is disabled.
bool WebRtcSession::CreateChannels(const SessionDescription* desc) {
// Creating the media channels and transport proxies.
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);
if (voice && !voice->rejected && !voice_channel_) {
if (!CreateVoiceChannel(voice)) {
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)) {
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 && !data_channel_) {
if (!CreateDataChannel(data)) {
LOG(LS_ERROR) << "Failed to create data channel.";
return false;
}
}
if (rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire) {
if (voice_channel()) {
voice_channel()->ActivateRtcpMux();
}
if (video_channel()) {
video_channel()->ActivateRtcpMux();
}
if (data_channel()) {
data_channel()->ActivateRtcpMux();
}
}
// Enable BUNDLE immediately when kBundlePolicyMaxBundle is in effect.
if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {
const cricket::ContentGroup* bundle_group = desc->GetGroupByName(
cricket::GROUP_TYPE_BUNDLE);
if (!bundle_group) {
LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";
return false;
}
if (!EnableBundle(*bundle_group)) {
LOG(LS_WARNING) << "max-bundle failed to enable bundling.";
return false;
}
}
return true;
}
bool WebRtcSession::CreateVoiceChannel(const cricket::ContentInfo* content) {
voice_channel_.reset(channel_manager_->CreateVoiceChannel(
media_controller_, transport_controller_.get(), content->name, true,
audio_options_));
if (!voice_channel_) {
return false;
}
voice_channel_->SignalDtlsSetupFailure.connect(
this, &WebRtcSession::OnDtlsSetupFailure);
SignalVoiceChannelCreated();
voice_channel_->SignalSentPacket.connect(this,
&WebRtcSession::OnSentPacket_w);
return true;
}
bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content) {
video_channel_.reset(channel_manager_->CreateVideoChannel(
media_controller_, transport_controller_.get(), content->name, true,
video_options_));
if (!video_channel_) {
return false;
}
video_channel_->SignalDtlsSetupFailure.connect(
this, &WebRtcSession::OnDtlsSetupFailure);
SignalVideoChannelCreated();
video_channel_->SignalSentPacket.connect(this,
&WebRtcSession::OnSentPacket_w);
return true;
}
bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content) {
bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
data_channel_.reset(channel_manager_->CreateDataChannel(
transport_controller_.get(), content->name, !sctp, data_channel_type_));
if (!data_channel_) {
return false;
}
if (sctp) {
data_channel_->SignalDataReceived.connect(
this, &WebRtcSession::OnDataChannelMessageReceived);
}
data_channel_->SignalDtlsSetupFailure.connect(
this, &WebRtcSession::OnDtlsSetupFailure);
SignalDataChannelCreated();
data_channel_->SignalSentPacket.connect(this, &WebRtcSession::OnSentPacket_w);
return true;
}
void WebRtcSession::OnDtlsSetupFailure(cricket::BaseChannel*, bool rtcp) {
SetError(ERROR_TRANSPORT,
rtcp ? kDtlsSetupFailureRtcp : kDtlsSetupFailureRtp);
}
void WebRtcSession::OnDataChannelMessageReceived(
cricket::DataChannel* channel,
const cricket::ReceiveDataParams& params,
const rtc::CopyOnWriteBuffer& payload) {
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
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);
}
// Otherwise ignore the message.
}
// 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);
ASSERT(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);
ASSERT(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);
}
// Verify m-lines in Answer when compared against Offer.
if (action == kAnswer) {
const cricket::SessionDescription* offer_desc =
(source == cricket::CS_LOCAL) ? remote_desc_->description()
: local_desc_->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_desc_.get();
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];
cricket::BaseChannel* channel = GetChannel(content.name);
if (!channel) {
return false;
}
return transport_controller_->ReadyForRemoteCandidates(
channel->transport_name());
}
void WebRtcSession::OnTransportControllerGatheringState(
cricket::IceGatheringState state) {
ASSERT(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 (data_channel()) {
transport_names.insert(data_channel()->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());
media_controller_->call_w()->OnSentPacket(sent_packet);
}
} // namespace webrtc
Reference in New Issue View Git Blame Copy Permalink