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webrtc / src / 33c31fbeef3157f888049edb49d6c9e3ffa851d9 / . / pc / sdp_offer_answer.cc
blob: 9bf6557ea95614b9338b3f1d0bc16c26dd2eee9f [file] [log] [blame]
/*
* Copyright 2020 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/sdp_offer_answer.h"
#include <algorithm>
#include <cstddef>
#include <cstdint>
#include <functional>
#include <iterator>
#include <map>
#include <memory>
#include <optional>
#include <queue>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/functional/any_invocable.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "absl/strings/str_cat.h"
#include "absl/strings/str_split.h"
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/jsep.h"
#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/set_local_description_observer_interface.h"
#include "api/set_remote_description_observer_interface.h"
#include "api/uma_metrics.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_codec_constants.h"
#include "call/payload_type.h"
#include "media/base/codec.h"
#include "media/base/codec_comparators.h"
#include "media/base/media_constants.h"
#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
#include "media/base/stream_params.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/p2p_transport_channel.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_description_factory.h"
#include "p2p/base/transport_info.h"
#include "pc/channel_interface.h"
#include "pc/codec_vendor.h"
#include "pc/connection_context.h"
#include "pc/dtls_transport.h"
#include "pc/jsep_transport_controller.h"
#include "pc/legacy_stats_collector.h"
#include "pc/media_options.h"
#include "pc/media_session.h"
#include "pc/media_stream.h"
#include "pc/media_stream_observer.h"
#include "pc/media_stream_proxy.h"
#include "pc/peer_connection_internal.h"
#include "pc/peer_connection_message_handler.h"
#include "pc/rtp_media_utils.h"
#include "pc/rtp_receiver.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_sender_proxy.h"
#include "pc/rtp_transceiver.h"
#include "pc/rtp_transmission_manager.h"
#include "pc/sdp_munging_detector.h"
#include "pc/session_description.h"
#include "pc/simulcast_description.h"
#include "pc/stream_collection.h"
#include "pc/transceiver_list.h"
#include "pc/usage_pattern.h"
#include "pc/webrtc_session_description_factory.h"
#include "rtc_base/checks.h"
#include "rtc_base/crypto_random.h"
#include "rtc_base/logging.h"
#include "rtc_base/operations_chain.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/weak_ptr.h"
#include "system_wrappers/include/metrics.h"
using ::webrtc::ContentInfo;
using ::webrtc::ContentInfos;
using webrtc::MediaContentDescription;
using ::webrtc::MediaProtocolType;
using ::webrtc::RidDescription;
using ::webrtc::RidDirection;
using ::webrtc::SessionDescription;
using ::webrtc::SimulcastDescription;
using ::webrtc::SimulcastLayer;
using ::webrtc::SimulcastLayerList;
using ::webrtc::StreamParams;
using ::webrtc::TransportInfo;
namespace webrtc {
namespace {
typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
// Error messages
const char kInvalidSdp[] = "Invalid session description.";
const char kInvalidCandidates[] = "Description contains invalid candidates.";
const char kBundleWithoutRtcpMux[] =
"rtcp-mux must be enabled when BUNDLE "
"is enabled.";
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 kSdpWithoutIceUfragPwd[] =
"Called with SDP without ice-ufrag and ice-pwd.";
const char kSdpWithoutDtlsFingerprint[] =
"Called with SDP without DTLS fingerprint.";
const char kSdpWithoutCrypto[] = "Called with SDP without crypto setup.";
const char kSessionError[] = "Session error code: ";
const char kSessionErrorDesc[] = "Session error description: ";
// The length of RTCP CNAMEs.
const int kRtcpCnameLength = 16;
// The maximum length of the MID attribute.
constexpr size_t kMidMaxSize = 16;
const char kDefaultStreamId[] = "default";
// NOTE: Duplicated in peer_connection.cc:
const char kDefaultAudioSenderId[] = "defaulta0";
const char kDefaultVideoSenderId[] = "defaultv0";
void NoteAddIceCandidateResult(int result) {
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result,
kAddIceCandidateMax);
}
std::map<std::string, const ContentGroup*> GetBundleGroupsByMid(
const SessionDescription* desc) {
std::vector<const ContentGroup*> bundle_groups =
desc->GetGroupsByName(GROUP_TYPE_BUNDLE);
std::map<std::string, const ContentGroup*> bundle_groups_by_mid;
for (const ContentGroup* bundle_group : bundle_groups) {
for (const std::string& content_name : bundle_group->content_names()) {
bundle_groups_by_mid[content_name] = bundle_group;
}
}
return bundle_groups_by_mid;
}
// 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 TransportDescription* new_transport_desc =
new_sd->GetTransportDescriptionByName(content_name);
const 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 (IceCredentialsChanged(
old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
<< ".";
return true;
}
return false;
}
// Generates a string error message for SetLocalDescription/SetRemoteDescription
// from an RTCError.
std::string GetSetDescriptionErrorMessage(ContentSource source,
SdpType type,
const RTCError& error) {
StringBuilder oss;
oss << "Failed to set " << (source == CS_LOCAL ? "local" : "remote") << " "
<< SdpTypeToString(type) << " sdp: ";
RTC_DCHECK(!absl::StartsWith(error.message(), oss.str())) << error.message();
oss << error.message();
return oss.Release();
}
std::string GetStreamIdsString(ArrayView<const std::string> stream_ids) {
std::string output = "streams=[";
const char* separator = "";
for (const auto& stream_id : stream_ids) {
output.append(separator).append(stream_id);
separator = ", ";
}
output.append("]");
return output;
}
const ContentInfo* FindTransceiverMSection(
RtpTransceiver* transceiver,
const SessionDescriptionInterface* session_description) {
return transceiver->mid()
? session_description->description()->GetContentByName(
*transceiver->mid())
: nullptr;
}
// If the direction is "recvonly" or "inactive", treat the description
// as containing no streams.
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
std::vector<StreamParams> GetActiveStreams(
const MediaContentDescription* desc) {
return RtpTransceiverDirectionHasSend(desc->direction())
? desc->streams()
: std::vector<StreamParams>();
}
// Logic to decide if an m= section can be recycled. This means that the new
// m= section is not rejected, but the old local or remote m= section is
// rejected. `old_content_one` and `old_content_two` refer to the m= section
// of the old remote and old local descriptions in no particular order.
// We need to check both the old local and remote because either
// could be the most current from the latest negotation.
bool IsMediaSectionBeingRecycled(SdpType type,
const ContentInfo& content,
const ContentInfo* old_content_one,
const ContentInfo* old_content_two) {
return type == SdpType::kOffer && !content.rejected &&
((old_content_one && old_content_one->rejected) ||
(old_content_two && old_content_two->rejected));
}
// Verify that the order of media sections in `new_desc` matches
// `current_desc`. The number of m= sections in `new_desc` should be no
// less than `current_desc`. In the case of checking an answer's
// `new_desc`, the `current_desc` is the last offer that was set as the
// local or remote. In the case of checking an offer's `new_desc` we
// check against the local and remote descriptions stored from the last
// negotiation, because either of these could be the most up to date for
// possible rejected m sections. These are the `current_desc` and
// `secondary_current_desc`.
bool MediaSectionsInSameOrder(const SessionDescription& current_desc,
const SessionDescription* secondary_current_desc,
const SessionDescription& new_desc,
const SdpType type) {
if (current_desc.contents().size() > new_desc.contents().size()) {
return false;
}
for (size_t i = 0; i < current_desc.contents().size(); ++i) {
const ContentInfo* secondary_content_info = nullptr;
if (secondary_current_desc &&
i < secondary_current_desc->contents().size()) {
secondary_content_info = &secondary_current_desc->contents()[i];
}
if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i],
&current_desc.contents()[i],
secondary_content_info)) {
// For new offer descriptions, if the media section can be recycled, it's
// valid for the MID and media type to change.
continue;
}
if (new_desc.contents()[i].mid() != current_desc.contents()[i].mid()) {
return false;
}
const MediaContentDescription* new_desc_mdesc =
new_desc.contents()[i].media_description();
const MediaContentDescription* current_desc_mdesc =
current_desc.contents()[i].media_description();
if (new_desc_mdesc->type() != current_desc_mdesc->type()) {
return false;
}
}
return true;
}
bool MediaSectionsHaveSameCount(const SessionDescription& desc1,
const SessionDescription& desc2) {
return desc1.contents().size() == desc2.contents().size();
}
// Checks that each non-rejected content has a DTLS
// fingerprint, unless it's in a BUNDLE group, in which case only the
// BUNDLE-tag section (first media section/description in the BUNDLE group)
// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
// by Channel's `srtp_required` check.
RTCError VerifyCrypto(
const SessionDescription* desc,
bool dtls_enabled,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
for (const ContentInfo& content_info : desc->contents()) {
if (content_info.rejected) {
continue;
}
const auto mid = content_info.mid();
auto it = bundle_groups_by_mid.find(mid);
const ContentGroup* bundle =
it != bundle_groups_by_mid.end() ? it->second : nullptr;
if (bundle && mid != *(bundle->FirstContentName())) {
// This isn't the first media section in the BUNDLE group, so it's not
// required to have crypto attributes, since only the crypto attributes
// from the first section actually get used.
continue;
}
// If the content isn't rejected or bundled into another m= section, crypto
// must be present.
const MediaContentDescription* media = content_info.media_description();
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
if (!media || !tinfo) {
// Something is not right.
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
}
if (dtls_enabled) {
if (!tinfo->description.identity_fingerprint) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kSdpWithoutDtlsFingerprint);
}
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kSdpWithoutCrypto);
}
}
return RTCError::OK();
}
// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
// media section/description in the BUNDLE group) needs a ufrag and pwd.
bool VerifyIceUfragPwdPresent(
const SessionDescription* desc,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
for (const ContentInfo& content_info : desc->contents()) {
if (content_info.rejected) {
continue;
}
const auto mid = content_info.mid();
auto it = bundle_groups_by_mid.find(mid);
const ContentGroup* bundle =
it != bundle_groups_by_mid.end() ? it->second : nullptr;
if (bundle && mid != *(bundle->FirstContentName())) {
// This isn't the first media section in the BUNDLE group, so it's not
// required to have ufrag/password, since only the ufrag/password from
// the first section actually get used.
continue;
}
// If the content isn't rejected or bundled into another m= section,
// ice-ufrag and ice-pwd must be present.
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
if (!tinfo) {
// Something is not right.
RTC_LOG(LS_ERROR) << kInvalidSdp;
return false;
}
if (tinfo->description.ice_ufrag.empty() ||
tinfo->description.ice_pwd.empty()) {
RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
return false;
}
}
return true;
}
RTCError ValidateMids(const SessionDescription& description) {
std::set<std::string> mids;
for (const ContentInfo& content : description.contents()) {
if (content.mid().empty()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"A media section is missing a MID attribute.");
}
if (content.mid().size() > kMidMaxSize) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"The MID attribute exceeds the maximum supported "
"length of 16 characters.");
}
if (!mids.insert(content.mid()).second) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Duplicate a=mid value '" + content.mid() + "'.");
}
}
return RTCError::OK();
}
RTCError FindDuplicateCodecParameters(
const RtpCodecParameters codec_parameters,
std::map<int, RtpCodecParameters>& payload_to_codec_parameters) {
auto existing_codec_parameters =
payload_to_codec_parameters.find(codec_parameters.payload_type);
if (existing_codec_parameters != payload_to_codec_parameters.end() &&
codec_parameters != existing_codec_parameters->second) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"A BUNDLE group contains a codec collision for "
"payload_type='" +
absl::StrCat(codec_parameters.payload_type) +
". All codecs must share the same type, "
"encoding name, clock rate and parameters.");
}
payload_to_codec_parameters.insert(
std::make_pair(codec_parameters.payload_type, codec_parameters));
return RTCError::OK();
}
RTCError ValidateBundledPayloadTypes(const SessionDescription& description) {
// https://www.rfc-editor.org/rfc/rfc8843#name-payload-type-pt-value-reuse
// ... all codecs associated with the payload type number MUST share an
// identical codec configuration. This means that the codecs MUST share
// the same media type, encoding name, clock rate, and any parameter
// that can affect the codec configuration and packetization.
std::vector<const ContentGroup*> bundle_groups =
description.GetGroupsByName(GROUP_TYPE_BUNDLE);
for (const ContentGroup* bundle_group : bundle_groups) {
std::map<int, RtpCodecParameters> payload_to_codec_parameters;
for (const std::string& content_name : bundle_group->content_names()) {
const ContentInfo* content_description =
description.GetContentByName(content_name);
if (!content_description) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"A BUNDLE group contains a MID='" + content_name +
"' matching no m= section.");
}
const MediaContentDescription* media_description =
content_description->media_description();
RTC_DCHECK(media_description);
if (content_description->rejected || !media_description ||
!media_description->has_codecs()) {
continue;
}
const auto type = media_description->type();
if (type == MediaType::AUDIO || type == MediaType::VIDEO) {
for (const auto& c : media_description->codecs()) {
auto error = FindDuplicateCodecParameters(
c.ToCodecParameters(), payload_to_codec_parameters);
if (!error.ok()) {
return error;
}
}
}
}
}
return RTCError::OK();
}
RTCError FindDuplicateHeaderExtensionIds(
const RtpExtension extension,
std::map<int, RtpExtension>& id_to_extension) {
auto existing_extension = id_to_extension.find(extension.id);
if (existing_extension != id_to_extension.end() &&
!(extension.uri == existing_extension->second.uri &&
extension.encrypt == existing_extension->second.encrypt)) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"A BUNDLE group contains a codec collision for "
"header extension id=" +
absl::StrCat(extension.id) +
". The id must be the same across all bundled media descriptions");
}
id_to_extension.insert(std::make_pair(extension.id, extension));
return RTCError::OK();
}
RTCError ValidateBundledRtpHeaderExtensions(
const SessionDescription& description) {
// https://www.rfc-editor.org/rfc/rfc8843#name-rtp-header-extensions-consi
// ... the identifier used for a given extension MUST identify the same
// extension across all the bundled media descriptions.
std::vector<const ContentGroup*> bundle_groups =
description.GetGroupsByName(GROUP_TYPE_BUNDLE);
for (const ContentGroup* bundle_group : bundle_groups) {
std::map<int, RtpExtension> id_to_extension;
for (const std::string& content_name : bundle_group->content_names()) {
const ContentInfo* content_description =
description.GetContentByName(content_name);
if (!content_description) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"A BUNDLE group contains a MID='" + content_name +
"' matching no m= section.");
}
const MediaContentDescription* media_description =
content_description->media_description();
RTC_DCHECK(media_description);
if (content_description->rejected || !media_description ||
!media_description->has_codecs()) {
continue;
}
for (const auto& extension : media_description->rtp_header_extensions()) {
auto error =
FindDuplicateHeaderExtensionIds(extension, id_to_extension);
if (!error.ok()) {
return error;
}
}
}
}
return RTCError::OK();
}
RTCError ValidateRtpHeaderExtensionsForSpecSimulcast(
const SessionDescription& description) {
for (const ContentInfo& content : description.contents()) {
if (content.type != MediaProtocolType::kRtp || content.rejected) {
continue;
}
const auto media_description = content.media_description();
if (!media_description->HasSimulcast()) {
continue;
}
auto extensions = media_description->rtp_header_extensions();
auto it = absl::c_find_if(extensions, [](const RtpExtension& ext) {
return ext.uri == RtpExtension::kRidUri;
});
if (it == extensions.end()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"The media section with MID='" + content.mid() +
"' negotiates simulcast but does not negotiate "
"the RID RTP header extension.");
}
}
return RTCError::OK();
}
RTCError ValidateSsrcGroups(const SessionDescription& description) {
for (const ContentInfo& content : description.contents()) {
if (content.type != MediaProtocolType::kRtp) {
continue;
}
for (const StreamParams& stream : content.media_description()->streams()) {
for (const SsrcGroup& group : stream.ssrc_groups) {
// Validate the number of SSRCs for standard SSRC group semantics such
// as FID and FEC-FR and the non-standard SIM group.
if ((group.semantics == kFidSsrcGroupSemantics &&
group.ssrcs.size() != 2) ||
(group.semantics == kFecFrSsrcGroupSemantics &&
group.ssrcs.size() != 2) ||
(group.semantics == kSimSsrcGroupSemantics &&
group.ssrcs.size() > kMaxSimulcastStreams)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"The media section with MID='" + content.mid() +
"' has a ssrc-group with semantics " +
group.semantics +
" and an unexpected number of SSRCs.");
}
}
}
}
return RTCError::OK();
}
RTCError ValidatePayloadTypes(const SessionDescription& description) {
for (const ContentInfo& content : description.contents()) {
if (content.type != MediaProtocolType::kRtp) {
continue;
}
const auto media_description = content.media_description();
RTC_DCHECK(media_description);
if (content.rejected || !media_description ||
!media_description->has_codecs()) {
continue;
}
const auto type = media_description->type();
if (type == MediaType::AUDIO || type == MediaType::VIDEO) {
for (const auto& codec : media_description->codecs()) {
if (!PayloadType::IsValid(codec.id, media_description->rtcp_mux())) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"The media section with MID='" + content.mid() +
"' used an invalid payload type " + absl::StrCat(codec.id) +
" for codec '" + codec.name + ", rtcp-mux:" +
(media_description->rtcp_mux() ? "enabled" : "disabled"));
}
}
}
}
return RTCError::OK();
}
bool IsValidOfferToReceiveMedia(int value) {
typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
return (value >= Options::kUndefined) &&
(value <= Options::kMaxOfferToReceiveMedia);
}
bool ValidateOfferAnswerOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
}
// This method will extract any send encodings that were sent by the remote
// connection. This is currently only relevant for Simulcast scenario (where
// the number of layers may be communicated by the server).
std::vector<RtpEncodingParameters> GetSendEncodingsFromRemoteDescription(
const MediaContentDescription& desc) {
if (!desc.HasSimulcast()) {
return {};
}
std::vector<RtpEncodingParameters> result;
const SimulcastDescription& simulcast = desc.simulcast_description();
// This is a remote description, the parameters we are after should appear
// as receive streams.
for (const auto& alternatives : simulcast.receive_layers()) {
RTC_DCHECK(!alternatives.empty());
// There is currently no way to specify or choose from alternatives.
// We will always use the first alternative, which is the most preferred.
const SimulcastLayer& layer = alternatives[0];
RtpEncodingParameters parameters;
parameters.rid = layer.rid;
parameters.active = !layer.is_paused;
// If a payload type has been specified for this rid, set the codec
// corresponding to that payload type.
auto rid_desc = std::find_if(
desc.receive_rids().begin(), desc.receive_rids().end(),
[&layer](const RidDescription& rid) { return rid.rid == layer.rid; });
if (rid_desc != desc.receive_rids().end() && !rid_desc->codecs.empty()) {
parameters.codec = rid_desc->codecs[0].ToCodecParameters();
}
result.push_back(parameters);
}
return result;
}
RTCError UpdateSimulcastLayerStatusInSender(
const std::vector<SimulcastLayer>& layers,
scoped_refptr<RtpSenderInternal> sender) {
RTC_DCHECK(sender);
RtpParameters parameters = sender->GetParametersInternalWithAllLayers();
std::vector<std::string> disabled_layers;
// The simulcast envelope cannot be changed, only the status of the streams.
// So we will iterate over the send encodings rather than the layers.
for (RtpEncodingParameters& encoding : parameters.encodings) {
auto iter = std::find_if(layers.begin(), layers.end(),
[&encoding](const SimulcastLayer& layer) {
return layer.rid == encoding.rid;
});
// A layer that cannot be found may have been removed by the remote party.
if (iter == layers.end()) {
disabled_layers.push_back(encoding.rid);
continue;
}
encoding.active = !iter->is_paused;
}
RTCError result = sender->SetParametersInternalWithAllLayers(parameters);
if (result.ok()) {
result = sender->DisableEncodingLayers(disabled_layers);
}
return result;
}
bool SimulcastIsRejected(const ContentInfo* local_content,
const MediaContentDescription& answer_media_desc,
bool enable_encrypted_rtp_header_extensions) {
bool simulcast_offered = local_content &&
local_content->media_description() &&
local_content->media_description()->HasSimulcast();
bool simulcast_answered = answer_media_desc.HasSimulcast();
bool rids_supported = RtpExtension::FindHeaderExtensionByUri(
answer_media_desc.rtp_header_extensions(), RtpExtension::kRidUri,
enable_encrypted_rtp_header_extensions
? RtpExtension::Filter::kPreferEncryptedExtension
: RtpExtension::Filter::kDiscardEncryptedExtension);
return simulcast_offered && (!simulcast_answered || !rids_supported);
}
RTCError DisableSimulcastInSender(scoped_refptr<RtpSenderInternal> sender) {
RTC_DCHECK(sender);
RtpParameters parameters = sender->GetParametersInternalWithAllLayers();
if (parameters.encodings.size() <= 1) {
return RTCError::OK();
}
std::vector<std::string> disabled_layers;
std::transform(
parameters.encodings.begin() + 1, parameters.encodings.end(),
std::back_inserter(disabled_layers),
[](const RtpEncodingParameters& encoding) { return encoding.rid; });
return sender->DisableEncodingLayers(disabled_layers);
}
// The SDP parser used to populate these values by default for the 'content
// name' if an a=mid line was absent.
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wdeprecated-declarations"
std::string GetDefaultMidForPlanB(MediaType media_type) {
switch (media_type) {
case MediaType::AUDIO:
return CN_AUDIO;
case MediaType::VIDEO:
return CN_VIDEO;
case MediaType::DATA:
return CN_DATA;
case MediaType::UNSUPPORTED:
return "not supported";
default:
// Fall through to RTC_CHECK_NOTREACHED
break;
}
RTC_DCHECK_NOTREACHED();
return "";
}
#pragma clang diagnostic pop
// Add options to |[audio/video]_media_description_options| from `senders`.
void AddPlanBRtpSenderOptions(
const std::vector<
scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
MediaDescriptionOptions* audio_media_description_options,
MediaDescriptionOptions* video_media_description_options,
int num_sim_layers) {
for (const auto& sender : senders) {
if (sender->media_type() == MediaType::AUDIO) {
if (audio_media_description_options) {
audio_media_description_options->AddAudioSender(
sender->id(), sender->internal()->stream_ids());
}
} else {
RTC_DCHECK(sender->media_type() == MediaType::VIDEO);
if (video_media_description_options) {
video_media_description_options->AddVideoSender(
sender->id(), sender->internal()->stream_ids(), {},
SimulcastLayerList(), num_sim_layers);
}
}
}
}
MediaDescriptionOptions GetMediaDescriptionOptionsForTransceiver(
RtpTransceiver* transceiver,
const std::string& mid,
bool is_create_offer) {
// NOTE: a stopping transceiver should be treated as a stopped one in
// createOffer as specified in
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer.
bool stopped =
is_create_offer ? transceiver->stopping() : transceiver->stopped();
MediaDescriptionOptions media_description_options(
transceiver->media_type(), mid, transceiver->direction(), stopped);
media_description_options.codec_preferences =
transceiver->filtered_codec_preferences();
media_description_options.header_extensions =
transceiver->GetHeaderExtensionsToNegotiate();
// This behavior is specified in JSEP. The gist is that:
// 1. The MSID is included if the RtpTransceiver's direction is sendonly or
// sendrecv.
// 2. If the MSID is included, then it must be included in any subsequent
// offer/answer exactly the same until the RtpTransceiver is stopped.
if (stopped || (!RtpTransceiverDirectionHasSend(transceiver->direction()) &&
!transceiver->has_ever_been_used_to_send())) {
return media_description_options;
}
SenderOptions sender_options;
sender_options.track_id = transceiver->sender()->id();
sender_options.stream_ids = transceiver->sender()->stream_ids();
// The following sets up RIDs and Simulcast.
// RIDs are included if Simulcast is requested or if any RID was specified.
RtpParameters send_parameters =
transceiver->sender_internal()->GetParametersInternalWithAllLayers();
bool has_rids = std::any_of(send_parameters.encodings.begin(),
send_parameters.encodings.end(),
[](const RtpEncodingParameters& encoding) {
return !encoding.rid.empty();
});
std::vector<RidDescription> send_rids;
SimulcastLayerList send_layers;
for (const RtpEncodingParameters& encoding : send_parameters.encodings) {
if (encoding.rid.empty()) {
continue;
}
auto send_rid = RidDescription(encoding.rid, RidDirection::kSend);
if (encoding.codec) {
auto send_codecs = transceiver->sender_internal()->GetSendCodecs();
for (const Codec& codec : send_codecs) {
if (IsSameRtpCodecIgnoringLevel(codec, *encoding.codec)) {
send_rid.codecs.push_back(codec);
break;
}
}
}
send_rids.push_back(send_rid);
send_layers.AddLayer(SimulcastLayer(encoding.rid, !encoding.active));
}
if (has_rids) {
sender_options.rids = send_rids;
}
sender_options.simulcast_layers = send_layers;
// When RIDs are configured, we must set num_sim_layers to 0 to.
// Otherwise, num_sim_layers must be 1 because either there is no
// simulcast, or simulcast is acheived by munging the SDP.
sender_options.num_sim_layers = has_rids ? 0 : 1;
media_description_options.sender_options.push_back(sender_options);
return media_description_options;
}
// Returns the ContentInfo at mline index `i`, or null if none exists.
const ContentInfo* GetContentByIndex(const SessionDescriptionInterface* sdesc,
size_t i) {
if (!sdesc) {
return nullptr;
}
const ContentInfos& contents = sdesc->description()->contents();
return (i < contents.size() ? &contents[i] : nullptr);
}
// From `rtc_options`, fill parts of `session_options` shared by all generated
// m= sectionss (in other words, nothing that involves a map/array).
void ExtractSharedMediaSessionOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
MediaSessionOptions* session_options) {
session_options->vad_enabled = rtc_options.voice_activity_detection;
session_options->bundle_enabled = rtc_options.use_rtp_mux;
session_options->raw_packetization_for_video =
rtc_options.raw_packetization_for_video;
}
// Generate a RTCP CNAME when a PeerConnection is created.
std::string GenerateRtcpCname() {
std::string cname;
if (!CreateRandomString(kRtcpCnameLength, &cname)) {
RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
RTC_DCHECK_NOTREACHED();
}
return cname;
}
// Check if we can send `new_stream` on a PeerConnection.
bool CanAddLocalMediaStream(StreamCollectionInterface* current_streams,
MediaStreamInterface* new_stream) {
if (!new_stream || !current_streams) {
return false;
}
if (current_streams->find(new_stream->id()) != nullptr) {
RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
<< " is already added.";
return false;
}
return true;
}
scoped_refptr<DtlsTransport> LookupDtlsTransportByMid(
Thread* network_thread,
JsepTransportController* controller,
const std::string& mid) {
// TODO(tommi): Can we post this (and associated operations where this
// function is called) to the network thread and avoid this BlockingCall?
// We might be able to simplify a few things if we set the transport on
// the network thread and then update the implementation to check that
// the set_ and relevant get methods are always called on the network
// thread (we'll need to update proxy maps).
return network_thread->BlockingCall(
[controller, &mid] { return controller->LookupDtlsTransportByMid(mid); });
}
bool ContentHasHeaderExtension(const ContentInfo& content_info,
absl::string_view header_extension_uri) {
for (const RtpExtension& rtp_header_extension :
content_info.media_description()->rtp_header_extensions()) {
if (rtp_header_extension.uri == header_extension_uri) {
return true;
}
}
return false;
}
// Matches the given host with the wildcard host. A wildcard `*` matches the
// current character and anything that comes after it. For instance, 127.0.*
// matches any string that starts with `127.0.`. For IPv6 addresses, the
// wildcard host should also start with `[` and end with `]` or a wildcard.
bool WildcardHostPrefixMatch(absl::string_view host,
absl::string_view wildcard_host) {
for (size_t i = 0; i < std::min(host.size(), wildcard_host.size()); ++i) {
if (host[i] == wildcard_host[i]) {
continue;
}
return wildcard_host[i] == '*';
}
return host.size() == wildcard_host.size();
}
} // namespace
void UpdateRtpHeaderExtensionPreferencesFromSdpMunging(
const SessionDescription* description,
TransceiverList* transceivers) {
// This integrates the RTP Header Extension Control API and local SDP munging
// for backward compability reasons. If something was enabled in the local
// description via SDP munging, consider it non-stopped in the API as well
// so that is shows up in subsequent offers/answers.
RTC_DCHECK(description);
RTC_DCHECK(transceivers);
for (const auto& content : description->contents()) {
auto transceiver = transceivers->FindByMid(content.mid());
if (!transceiver) {
continue;
}
auto extension_capabilities = transceiver->GetHeaderExtensionsToNegotiate();
// Set the capability of every extension we see here to "sendrecv".
for (auto& ext : content.media_description()->rtp_header_extensions()) {
auto it = absl::c_find_if(extension_capabilities,
[&ext](const RtpHeaderExtensionCapability c) {
return ext.uri == c.uri;
});
if (it != extension_capabilities.end()) {
it->direction = RtpTransceiverDirection::kSendRecv;
}
}
transceiver->SetHeaderExtensionsToNegotiate(extension_capabilities);
}
}
// This class stores state related to a SetRemoteDescription operation, captures
// and reports potential errors that might occur and makes sure to notify the
// observer of the operation and the operations chain of completion.
class SdpOfferAnswerHandler::RemoteDescriptionOperation {
public:
RemoteDescriptionOperation(
SdpOfferAnswerHandler* handler,
std::unique_ptr<SessionDescriptionInterface> desc,
scoped_refptr<SetRemoteDescriptionObserverInterface> observer,
std::function<void()> operations_chain_callback)
: handler_(handler),
desc_(std::move(desc)),
observer_(std::move(observer)),
operations_chain_callback_(std::move(operations_chain_callback)),
type_(desc_->GetType()),
unified_plan_(handler_->IsUnifiedPlan()) {}
~RemoteDescriptionOperation() {
RTC_DCHECK_RUN_ON(handler_->signaling_thread());
SignalCompletion();
operations_chain_callback_();
}
bool ok() const { return error_.ok(); }
// Notifies the observer that the operation is complete and releases the
// reference to the observer.
void SignalCompletion() {
if (!observer_)
return;
if (!error_.ok() && type_ != static_cast<SdpType>(-1)) {
std::string error_message =
GetSetDescriptionErrorMessage(CS_REMOTE, type_, error_);
RTC_LOG(LS_ERROR) << error_message;
error_.set_message(error_message);
}
observer_->OnSetRemoteDescriptionComplete(error_);
observer_ = nullptr; // Only fire the notification once.
}
// If a session error has occurred the PeerConnection is in a possibly
// inconsistent state so fail right away.
bool HaveSessionError() {
RTC_DCHECK(ok());
if (handler_->session_error() != SessionError::kNone)
InternalError(handler_->GetSessionErrorMsg());
return !ok();
}
// Returns true if the operation was a rollback operation. If this function
// returns true, the caller should consider the operation complete. Otherwise
// proceed to the next step.
bool MaybeRollback() {
RTC_DCHECK_RUN_ON(handler_->signaling_thread());
RTC_DCHECK(ok());
if (type_ != SdpType::kRollback) {
// Check if we can do an implicit rollback.
if (type_ == SdpType::kOffer && unified_plan_ &&
handler_->pc_->configuration()->enable_implicit_rollback &&
handler_->signaling_state() ==
PeerConnectionInterface::kHaveLocalOffer) {
handler_->Rollback(type_);
}
return false;
}
if (unified_plan_) {
error_ = handler_->Rollback(type_);
} else if (type_ == SdpType::kRollback) {
Unsupported("Rollback not supported in Plan B");
}
return true;
}
// Report to UMA the format of the received offer or answer.
void ReportOfferAnswerUma() {
RTC_DCHECK(ok());
if (type_ == SdpType::kOffer || type_ == SdpType::kAnswer) {
handler_->pc_->ReportSdpBundleUsage(*desc_);
}
}
// Checks if the session description for the operation is valid. If not, the
// function captures error information and returns false. Note that if the
// return value is false, the operation should be considered done.
bool IsDescriptionValid() {
RTC_DCHECK_RUN_ON(handler_->signaling_thread());
RTC_DCHECK(ok());
RTC_DCHECK(bundle_groups_by_mid_.empty()) << "Already called?";
bundle_groups_by_mid_ = GetBundleGroupsByMid(description());
error_ = handler_->ValidateSessionDescription(desc_.get(), CS_REMOTE,
bundle_groups_by_mid_);
return ok();
}
// Transfers ownership of the session description object over to `handler_`.
bool ReplaceRemoteDescriptionAndCheckError() {
RTC_DCHECK_RUN_ON(handler_->signaling_thread());
RTC_DCHECK(ok());
RTC_DCHECK(desc_);
RTC_DCHECK(!replaced_remote_description_);
#if RTC_DCHECK_IS_ON
const auto* existing_remote_description = handler_->remote_description();
#endif
error_ = handler_->ReplaceRemoteDescription(std::move(desc_), type_,
&replaced_remote_description_);
if (ok()) {
#if RTC_DCHECK_IS_ON
// Sanity check that our `old_remote_description()` method always returns
// the same value as `remote_description()` did before the call to
// ReplaceRemoteDescription.
RTC_DCHECK_EQ(existing_remote_description, old_remote_description());
#endif
} else {
SetAsSessionError();
}
return ok();
}
bool UpdateChannels() {
RTC_DCHECK(ok());
RTC_DCHECK(!desc_) << "ReplaceRemoteDescription hasn't been called";
const auto* remote_description = handler_->remote_description();
const SessionDescription* session_desc = remote_description->description();
// Transport and Media channels will be created only when offer is set.
if (unified_plan_) {
error_ = handler_->UpdateTransceiversAndDataChannels(
CS_REMOTE, *remote_description, handler_->local_description(),
old_remote_description(), bundle_groups_by_mid_);
} else {
// Media channels will be created only when offer is set. These may use
// new transports just created by PushdownTransportDescription.
if (type_ == SdpType::kOffer) {
// TODO(mallinath) - Handle CreateChannel failure, as new local
// description is applied. Restore back to old description.
error_ = handler_->CreateChannels(*session_desc);
}
// Remove unused channels if MediaContentDescription is rejected.
handler_->RemoveUnusedChannels(session_desc);
}
return ok();
}
bool UpdateSessionState() {
RTC_DCHECK(ok());
error_ = handler_->UpdateSessionState(
type_, CS_REMOTE, handler_->remote_description()->description(),
bundle_groups_by_mid_);
if (!ok())
SetAsSessionError();
return ok();
}
bool UseCandidatesInRemoteDescription() {
RTC_DCHECK(ok());
if (handler_->local_description() &&
!handler_->UseCandidatesInRemoteDescription()) {
InvalidParam(kInvalidCandidates);
}
return ok();
}
// Convenience getter for desc_->GetType().
SdpType type() const { return type_; }
bool unified_plan() const { return unified_plan_; }
SessionDescription* description() { return desc_->description(); }
const SessionDescriptionInterface* old_remote_description() const {
RTC_DCHECK(!desc_) << "Called before replacing the remote description";
if (type_ == SdpType::kAnswer)
return replaced_remote_description_.get();
return replaced_remote_description_
? replaced_remote_description_.get()
: handler_->current_remote_description();
}
// Returns a reference to a cached map of bundle groups ordered by mid.
// Note that this will only be valid after a successful call to
// `IsDescriptionValid`.
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid()
const {
RTC_DCHECK(ok());
return bundle_groups_by_mid_;
}
private:
// Convenience methods for populating the embedded `error_` object.
void Unsupported(std::string message) {
SetError(RTCErrorType::UNSUPPORTED_OPERATION, std::move(message));
}
void InvalidParam(std::string message) {
SetError(RTCErrorType::INVALID_PARAMETER, std::move(message));
}
void InternalError(std::string message) {
SetError(RTCErrorType::INTERNAL_ERROR, std::move(message));
}
void SetError(RTCErrorType type, std::string message) {
RTC_DCHECK(ok()) << "Overwriting an existing error?";
error_ = RTCError(type, std::move(message));
}
// Called when the PeerConnection could be in an inconsistent state and we set
// the session error so that future calls to
// SetLocalDescription/SetRemoteDescription fail.
void SetAsSessionError() {
RTC_DCHECK(!ok());
handler_->SetSessionError(SessionError::kContent, error_.message());
}
SdpOfferAnswerHandler* const handler_;
std::unique_ptr<SessionDescriptionInterface> desc_;
// Keeps the replaced session description object alive while the operation
// is taking place since methods that depend on `old_remote_description()`
// for updating the state, need it.
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description_;
scoped_refptr<SetRemoteDescriptionObserverInterface> observer_;
std::function<void()> operations_chain_callback_;
RTCError error_ = RTCError::OK();
std::map<std::string, const ContentGroup*> bundle_groups_by_mid_;
const SdpType type_;
const bool unified_plan_;
};
// Used by parameterless SetLocalDescription() to create an offer or answer.
// Upon completion of creating the session description, SetLocalDescription() is
// invoked with the result.
class SdpOfferAnswerHandler::ImplicitCreateSessionDescriptionObserver
: public CreateSessionDescriptionObserver {
public:
ImplicitCreateSessionDescriptionObserver(
WeakPtr<SdpOfferAnswerHandler> sdp_handler,
scoped_refptr<SetLocalDescriptionObserverInterface>
set_local_description_observer)
: sdp_handler_(std::move(sdp_handler)),
set_local_description_observer_(
std::move(set_local_description_observer)) {}
~ImplicitCreateSessionDescriptionObserver() override {
RTC_DCHECK(was_called_);
}
void SetOperationCompleteCallback(
std::function<void()> operation_complete_callback) {
operation_complete_callback_ = std::move(operation_complete_callback);
}
bool was_called() const { return was_called_; }
void OnSuccess(SessionDescriptionInterface* desc_ptr) override {
RTC_DCHECK(!was_called_);
std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr);
was_called_ = true;
// Abort early if `pc_` is no longer valid.
if (!sdp_handler_) {
operation_complete_callback_();
return;
}
// DoSetLocalDescription() is a synchronous operation that invokes
// `set_local_description_observer_` with the result.
sdp_handler_->DoSetLocalDescription(
std::move(desc), std::move(set_local_description_observer_));
operation_complete_callback_();
}
void OnFailure(RTCError error) override {
RTC_DCHECK(!was_called_);
was_called_ = true;
set_local_description_observer_->OnSetLocalDescriptionComplete(RTCError(
error.type(), std::string("SetLocalDescription failed to create "
"session description - ") +
error.message()));
operation_complete_callback_();
}
private:
bool was_called_ = false;
WeakPtr<SdpOfferAnswerHandler> sdp_handler_;
scoped_refptr<SetLocalDescriptionObserverInterface>
set_local_description_observer_;
std::function<void()> operation_complete_callback_;
};
// Wraps a CreateSessionDescriptionObserver and an OperationsChain operation
// complete callback. When the observer is invoked, the wrapped observer is
// invoked followed by invoking the completion callback.
class CreateSessionDescriptionObserverOperationWrapper
: public CreateSessionDescriptionObserver {
public:
CreateSessionDescriptionObserverOperationWrapper(
scoped_refptr<CreateSessionDescriptionObserver> observer,
std::function<void()> operation_complete_callback)
: observer_(std::move(observer)),
operation_complete_callback_(std::move(operation_complete_callback)) {
RTC_DCHECK(observer_);
}
~CreateSessionDescriptionObserverOperationWrapper() override {
#if RTC_DCHECK_IS_ON
RTC_DCHECK(was_called_);
#endif
}
void OnSuccess(SessionDescriptionInterface* desc) override {
#if RTC_DCHECK_IS_ON
RTC_DCHECK(!was_called_);
was_called_ = true;
#endif // RTC_DCHECK_IS_ON
// Completing the operation before invoking the observer allows the observer
// to execute SetLocalDescription() without delay.
operation_complete_callback_();
desc->RelinquishThreadOwnership();
observer_->OnSuccess(desc);
}
void OnFailure(RTCError error) override {
#if RTC_DCHECK_IS_ON
RTC_DCHECK(!was_called_);
was_called_ = true;
#endif // RTC_DCHECK_IS_ON
operation_complete_callback_();
observer_->OnFailure(std::move(error));
}
private:
#if RTC_DCHECK_IS_ON
bool was_called_ = false;
#endif // RTC_DCHECK_IS_ON
scoped_refptr<CreateSessionDescriptionObserver> observer_;
std::function<void()> operation_complete_callback_;
};
// Wraps a session description observer so a Clone of the last created
// offer/answer can be stored.
// This object can be used only once.
class CreateDescriptionObserverWrapperWithCreationCallback
: public CreateSessionDescriptionObserver {
public:
CreateDescriptionObserverWrapperWithCreationCallback(
absl::AnyInvocable<void(std::unique_ptr<SessionDescriptionInterface>) &&>
callback,
scoped_refptr<CreateSessionDescriptionObserver> observer)
: callback_(std::move(callback)), observer_(observer) {
RTC_DCHECK(observer_);
}
void OnSuccess(SessionDescriptionInterface* desc) override {
// `OnSuccess` will be called on the signaling thread. That's what we want
// for SdpOfferAnswerHandler since that's where public methods are called.
// The wrapped observer might have different threading needs. Calling the
// `Clone()` method will require accessing internal state of `desc` and
// therefore will (at least semantically) attach its state to the current
// thread. So, since we need to call Clone(), keep `desc` as the copy that
// SdpOfferAnswerHandler will own, and give a pristine clone to the
// observer.
auto clone = desc->Clone();
std::move(callback_)(absl::WrapUnique(desc));
observer_->OnSuccess(clone.release());
}
void OnFailure(RTCError error) override {
std::move(callback_)(nullptr);
observer_->OnFailure(std::move(error));
}
private:
absl::AnyInvocable<void(std::unique_ptr<SessionDescriptionInterface>) &&>
callback_;
scoped_refptr<CreateSessionDescriptionObserver> observer_;
};
// Wrapper for SetSessionDescriptionObserver that invokes the success or failure
// callback in a posted message handled by the peer connection. This introduces
// a delay that prevents recursive API calls by the observer, but this also
// means that the PeerConnection can be modified before the observer sees the
// result of the operation. This is ill-advised for synchronizing states.
//
// Implements both the SetLocalDescriptionObserverInterface and the
// SetRemoteDescriptionObserverInterface.
class SdpOfferAnswerHandler::SetSessionDescriptionObserverAdapter
: public SetLocalDescriptionObserverInterface,
public SetRemoteDescriptionObserverInterface {
public:
SetSessionDescriptionObserverAdapter(
WeakPtr<SdpOfferAnswerHandler> handler,
scoped_refptr<SetSessionDescriptionObserver> inner_observer)
: handler_(std::move(handler)),
inner_observer_(std::move(inner_observer)) {}
// SetLocalDescriptionObserverInterface implementation.
void OnSetLocalDescriptionComplete(RTCError error) override {
OnSetDescriptionComplete(std::move(error));
}
// SetRemoteDescriptionObserverInterface implementation.
void OnSetRemoteDescriptionComplete(RTCError error) override {
OnSetDescriptionComplete(std::move(error));
}
private:
void OnSetDescriptionComplete(RTCError error) {
if (!handler_)
return;
if (error.ok()) {
handler_->pc_->message_handler()->PostSetSessionDescriptionSuccess(
inner_observer_.get());
} else {
handler_->pc_->message_handler()->PostSetSessionDescriptionFailure(
inner_observer_.get(), std::move(error));
}
}
WeakPtr<SdpOfferAnswerHandler> handler_;
scoped_refptr<SetSessionDescriptionObserver> inner_observer_;
};
class SdpOfferAnswerHandler::LocalIceCredentialsToReplace {
public:
// Sets the ICE credentials that need restarting to the ICE credentials of
// the current and pending descriptions.
void SetIceCredentialsFromLocalDescriptions(
const SessionDescriptionInterface* current_local_description,
const SessionDescriptionInterface* pending_local_description) {
ice_credentials_.clear();
if (current_local_description) {
AppendIceCredentialsFromSessionDescription(*current_local_description);
}
if (pending_local_description) {
AppendIceCredentialsFromSessionDescription(*pending_local_description);
}
}
void ClearIceCredentials() { ice_credentials_.clear(); }
// Returns true if we have ICE credentials that need restarting.
bool HasIceCredentials() const { return !ice_credentials_.empty(); }
// Returns true if `local_description` shares no ICE credentials with the
// ICE credentials that need restarting.
bool SatisfiesIceRestart(
const SessionDescriptionInterface& local_description) const {
for (const auto& transport_info :
local_description.description()->transport_infos()) {
if (ice_credentials_.find(std::make_pair(
transport_info.description.ice_ufrag,
transport_info.description.ice_pwd)) != ice_credentials_.end()) {
return false;
}
}
return true;
}
private:
void AppendIceCredentialsFromSessionDescription(
const SessionDescriptionInterface& desc) {
for (const auto& transport_info : desc.description()->transport_infos()) {
ice_credentials_.insert(
std::make_pair(transport_info.description.ice_ufrag,
transport_info.description.ice_pwd));
}
}
std::set<std::pair<std::string, std::string>> ice_credentials_;
};
SdpOfferAnswerHandler::SdpOfferAnswerHandler(const Environment& env,
PeerConnectionSdpMethods* pc,
ConnectionContext* context)
: env_(env),
pc_(pc),
context_(context),
local_streams_(StreamCollection::Create()),
remote_streams_(StreamCollection::Create()),
operations_chain_(OperationsChain::Create()),
rtcp_cname_(GenerateRtcpCname()),
local_ice_credentials_to_replace_(new LocalIceCredentialsToReplace()),
weak_ptr_factory_(this) {
operations_chain_->SetOnChainEmptyCallback(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr()]() {
if (!this_weak_ptr)
return;
this_weak_ptr->OnOperationsChainEmpty();
});
}
SdpOfferAnswerHandler::~SdpOfferAnswerHandler() {}
// Static
std::unique_ptr<SdpOfferAnswerHandler> SdpOfferAnswerHandler::Create(
const Environment& env,
PeerConnectionSdpMethods* pc,
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<RTCCertificateGeneratorInterface> cert_generator,
std::unique_ptr<VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory,
ConnectionContext* context,
CodecLookupHelper* codec_lookup_helper) {
auto handler = absl::WrapUnique(new SdpOfferAnswerHandler(env, pc, context));
handler->Initialize(configuration, std::move(cert_generator),
std::move(video_bitrate_allocator_factory), context,
codec_lookup_helper);
return handler;
}
void SdpOfferAnswerHandler::Initialize(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<RTCCertificateGeneratorInterface> cert_generator,
std::unique_ptr<VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory,
ConnectionContext* context,
CodecLookupHelper* codec_lookup_helper) {
RTC_DCHECK_RUN_ON(signaling_thread());
// 100 kbps is used by default, but can be overriden by a non-standard
// RTCConfiguration value (not available on Web).
video_options_.screencast_min_bitrate_kbps =
configuration.screencast_min_bitrate.value_or(100);
audio_options_.audio_jitter_buffer_max_packets =
configuration.audio_jitter_buffer_max_packets;
audio_options_.audio_jitter_buffer_fast_accelerate =
configuration.audio_jitter_buffer_fast_accelerate;
audio_options_.audio_jitter_buffer_min_delay_ms =
configuration.audio_jitter_buffer_min_delay_ms;
// Obtain a certificate from RTCConfiguration if any were provided (optional).
scoped_refptr<RTCCertificate> certificate;
if (!configuration.certificates.empty()) {
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
// just picking the first one. The decision should be made based on the DTLS
// handshake. The DTLS negotiations need to know about all certificates.
certificate = configuration.certificates[0];
}
webrtc_session_desc_factory_ =
std::make_unique<WebRtcSessionDescriptionFactory>(
context, this, pc_->session_id(), pc_->dtls_enabled(),
std::move(cert_generator), std::move(certificate),
[this](const scoped_refptr<RTCCertificate>& certificate) {
RTC_DCHECK_RUN_ON(signaling_thread());
transport_controller_s()->SetLocalCertificate(certificate);
},
codec_lookup_helper, pc_->trials());
if (pc_->options()->disable_encryption) {
RTC_LOG(LS_INFO)
<< "Disabling encryption. This should only be done in tests.";
webrtc_session_desc_factory_->SetInsecureForTesting();
}
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
pc_->GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
webrtc_session_desc_factory_->set_is_unified_plan(IsUnifiedPlan());
video_bitrate_allocator_factory_ = std::move(video_bitrate_allocator_factory);
if (!video_bitrate_allocator_factory_) {
video_bitrate_allocator_factory_ =
CreateBuiltinVideoBitrateAllocatorFactory();
}
codec_lookup_helper_ = codec_lookup_helper;
}
// ==================================================================
// Access to pc_ variables
MediaEngineInterface* SdpOfferAnswerHandler::media_engine() const {
RTC_DCHECK(context_);
return context_->media_engine();
}
TransceiverList* SdpOfferAnswerHandler::transceivers() {
if (!pc_->rtp_manager()) {
return nullptr;
}
return pc_->rtp_manager()->transceivers();
}
const TransceiverList* SdpOfferAnswerHandler::transceivers() const {
if (!pc_->rtp_manager()) {
return nullptr;
}
return pc_->rtp_manager()->transceivers();
}
JsepTransportController* SdpOfferAnswerHandler::transport_controller_s() {
return pc_->transport_controller_s();
}
JsepTransportController* SdpOfferAnswerHandler::transport_controller_n() {
return pc_->transport_controller_n();
}
const JsepTransportController* SdpOfferAnswerHandler::transport_controller_s()
const {
return pc_->transport_controller_s();
}
const JsepTransportController* SdpOfferAnswerHandler::transport_controller_n()
const {
return pc_->transport_controller_n();
}
DataChannelController* SdpOfferAnswerHandler::data_channel_controller() {
return pc_->data_channel_controller();
}
const DataChannelController* SdpOfferAnswerHandler::data_channel_controller()
const {
return pc_->data_channel_controller();
}
PortAllocator* SdpOfferAnswerHandler::port_allocator() {
return pc_->port_allocator();
}
const PortAllocator* SdpOfferAnswerHandler::port_allocator() const {
return pc_->port_allocator();
}
RtpTransmissionManager* SdpOfferAnswerHandler::rtp_manager() {
return pc_->rtp_manager();
}
const RtpTransmissionManager* SdpOfferAnswerHandler::rtp_manager() const {
return pc_->rtp_manager();
}
// ===================================================================
void SdpOfferAnswerHandler::PrepareForShutdown() {
RTC_DCHECK_RUN_ON(signaling_thread());
weak_ptr_factory_.InvalidateWeakPtrs();
}
void SdpOfferAnswerHandler::Close() {
ChangeSignalingState(PeerConnectionInterface::kClosed);
}
void SdpOfferAnswerHandler::RestartIce() {
RTC_DCHECK_RUN_ON(signaling_thread());
local_ice_credentials_to_replace_->SetIceCredentialsFromLocalDescriptions(
current_local_description(), pending_local_description());
UpdateNegotiationNeeded();
}
Thread* SdpOfferAnswerHandler::signaling_thread() const {
return context_->signaling_thread();
}
Thread* SdpOfferAnswerHandler::network_thread() const {
return context_->network_thread();
}
void SdpOfferAnswerHandler::CreateOffer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
// lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
observer_refptr =
scoped_refptr<CreateSessionDescriptionObserver>(observer),
options](std::function<void()> operations_chain_callback) {
// Abort early if `this_weak_ptr` is no longer valid.
if (!this_weak_ptr) {
observer_refptr->OnFailure(
RTCError(RTCErrorType::INTERNAL_ERROR,
"CreateOffer failed because the session was shut down"));
operations_chain_callback();
return;
}
// The operation completes asynchronously when the wrapper is invoked.
auto observer_wrapper =
make_ref_counted<CreateSessionDescriptionObserverOperationWrapper>(
std::move(observer_refptr),
std::move(operations_chain_callback));
this_weak_ptr->DoCreateOffer(options, observer_wrapper);
});
}
void SdpOfferAnswerHandler::SetLocalDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc_ptr);
RTC_DCHECK(observer);
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
// lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
observer_refptr = scoped_refptr<SetSessionDescriptionObserver>(observer),
desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
std::function<void()> operations_chain_callback) mutable {
// Abort early if `this_weak_ptr` is no longer valid.
if (!this_weak_ptr) {
// For consistency with SetSessionDescriptionObserverAdapter whose
// posted messages doesn't get processed when the PC is destroyed, we
// do not inform `observer_refptr` that the operation failed.
operations_chain_callback();
return;
}
// SetSessionDescriptionObserverAdapter takes care of making sure the
// `observer_refptr` is invoked in a posted message.
this_weak_ptr->DoSetLocalDescription(
std::move(desc),
make_ref_counted<SetSessionDescriptionObserverAdapter>(
this_weak_ptr, observer_refptr));
// For backwards-compatability reasons, we declare the operation as
// completed here (rather than in a post), so that the operation chain
// is not blocked by this operation when the observer is invoked. This
// allows the observer to trigger subsequent offer/answer operations
// synchronously if the operation chain is now empty.
operations_chain_callback();
});
}
void SdpOfferAnswerHandler::SetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
RTC_DCHECK(observer);
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
// lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), observer,
desc = std::move(desc)](
std::function<void()> operations_chain_callback) mutable {
// Abort early if `this_weak_ptr` is no longer valid.
if (!this_weak_ptr) {
observer->OnSetLocalDescriptionComplete(RTCError(
RTCErrorType::INTERNAL_ERROR,
"SetLocalDescription failed because the session was shut down"));
operations_chain_callback();
return;
}
this_weak_ptr->DoSetLocalDescription(std::move(desc), observer);
// DoSetLocalDescription() is implemented as a synchronous operation.
// The `observer` will already have been informed that it completed, and
// we can mark this operation as complete without any loose ends.
operations_chain_callback();
});
}
void SdpOfferAnswerHandler::SetLocalDescription(
SetSessionDescriptionObserver* observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(observer);
SetLocalDescription(make_ref_counted<SetSessionDescriptionObserverAdapter>(
weak_ptr_factory_.GetWeakPtr(),
scoped_refptr<SetSessionDescriptionObserver>(observer)));
}
void SdpOfferAnswerHandler::SetLocalDescription(
scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(observer);
// The `create_sdp_observer` handles performing DoSetLocalDescription() with
// the resulting description as well as completing the operation.
auto create_sdp_observer =
make_ref_counted<ImplicitCreateSessionDescriptionObserver>(
weak_ptr_factory_.GetWeakPtr(), observer);
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
// lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
create_sdp_observer](std::function<void()> operations_chain_callback) {
// The `create_sdp_observer` is responsible for completing the
// operation.
create_sdp_observer->SetOperationCompleteCallback(
std::move(operations_chain_callback));
// Abort early if `this_weak_ptr` is no longer valid. This triggers the
// same code path as if DoCreateOffer() or DoCreateAnswer() failed.
if (!this_weak_ptr) {
create_sdp_observer->OnFailure(RTCError(
RTCErrorType::INTERNAL_ERROR,
"SetLocalDescription failed because the session was shut down"));
return;
}
switch (this_weak_ptr->signaling_state()) {
case PeerConnectionInterface::kStable:
case PeerConnectionInterface::kHaveLocalOffer:
case PeerConnectionInterface::kHaveRemotePrAnswer:
// TODO(hbos): If [LastCreatedOffer] exists and still represents the
// current state of the system, use that instead of creating another
// offer.
this_weak_ptr->DoCreateOffer(
PeerConnectionInterface::RTCOfferAnswerOptions(),
create_sdp_observer);
break;
case PeerConnectionInterface::kHaveLocalPrAnswer:
case PeerConnectionInterface::kHaveRemoteOffer:
// TODO(hbos): If [LastCreatedAnswer] exists and still represents
// the current state of the system, use that instead of creating
// another answer.
this_weak_ptr->DoCreateAnswer(
PeerConnectionInterface::RTCOfferAnswerOptions(),
create_sdp_observer);
break;
case PeerConnectionInterface::kClosed:
create_sdp_observer->OnFailure(RTCError(
RTCErrorType::INVALID_STATE,
"SetLocalDescription called when PeerConnection is closed."));
break;
}
});
}
RTCError SdpOfferAnswerHandler::ApplyLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyLocalDescription");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
// Invalidate the stats caches to make sure that they get
// updated the next time getStats() gets called, as updating the session
// description affects the stats.
pc_->ClearStatsCache();
// Take a reference to the old local description since it's used below to
// compare against the new local description. When setting the new local
// description, grab ownership of the replaced session description in case it
// is the same as `old_local_description`, to keep it alive for the duration
// of the method.
const SessionDescriptionInterface* old_local_description =
local_description();
std::unique_ptr<SessionDescriptionInterface> replaced_local_description;
SdpType type = desc->GetType();
if (type == SdpType::kAnswer) {
replaced_local_description = pending_local_description_
? std::move(pending_local_description_)
: std::move(current_local_description_);
current_local_description_ = std::move(desc);
pending_local_description_ = nullptr;
current_remote_description_ = std::move(pending_remote_description_);
} else {
replaced_local_description = std::move(pending_local_description_);
pending_local_description_ = std::move(desc);
}
if (!initial_offerer_) {
initial_offerer_.emplace(type == SdpType::kOffer);
}
// The session description to apply now must be accessed by
// `local_description()`.
RTC_DCHECK(local_description());
if (!is_caller_) {
if (remote_description()) {
// Remote description was applied first, so this PC is the callee.
is_caller_ = false;
} else {
// Local description is applied first, so this PC is the caller.
is_caller_ = true;
}
}
RTCError error = PushdownTransportDescription(CS_LOCAL, type);
if (!error.ok()) {
return error;
}
if (IsUnifiedPlan()) {
error = UpdateTransceiversAndDataChannels(
CS_LOCAL, *local_description(), old_local_description,
remote_description(), bundle_groups_by_mid);
if (!error.ok()) {
RTC_LOG(LS_ERROR) << error.message() << " (" << type << ")";
return error;
}
if (ConfiguredForMedia()) {
std::vector<scoped_refptr<RtpTransceiverInterface>> remove_list;
std::vector<scoped_refptr<MediaStreamInterface>> removed_streams;
for (const auto& transceiver_ext : transceivers()->List()) {
auto transceiver = transceiver_ext->internal();
if (transceiver->stopped()) {
continue;
}
// 2.2.7.1.1.(6-9): Set sender and receiver's transport slots.
// Note that code paths that don't set MID won't be able to use
// information about DTLS transports.
if (transceiver->mid()) {
auto dtls_transport = LookupDtlsTransportByMid(
context_->network_thread(), transport_controller_s(),
*transceiver->mid());
transceiver->sender_internal()->set_transport(dtls_transport);
transceiver->receiver_internal()->set_transport(dtls_transport);
}
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
const MediaContentDescription* media_desc =
content->media_description();
// 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
// the following steps:
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
// 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
// transceiver's [[FiredDirection]] slot is either "sendrecv" or
// "recvonly", process the removal of a remote track for the media
// description, given transceiver, removeList, and muteTracks.
if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
(transceiver->fired_direction() &&
RtpTransceiverDirectionHasRecv(
*transceiver->fired_direction()))) {
ProcessRemovalOfRemoteTrack(transceiver_ext, &remove_list,
&removed_streams);
}
// 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
// [[FiredDirection]] slots to direction.
transceiver->set_current_direction(media_desc->direction());
transceiver->set_fired_direction(media_desc->direction());
}
}
pc_->RunWithObserver([&](auto observer) {
for (const auto& transceiver : remove_list) {
observer->OnRemoveTrack(transceiver->receiver());
}
for (const auto& stream : removed_streams) {
observer->OnRemoveStream(stream);
}
});
}
} else {
// Media channels will be created only when offer is set. These may use new
// transports just created by PushdownTransportDescription.
if (type == SdpType::kOffer) {
// TODO(bugs.webrtc.org/4676) - Handle CreateChannel failure, as new local
// description is applied. Restore back to old description.
error = CreateChannels(*local_description()->description());
if (!error.ok()) {
RTC_LOG(LS_ERROR) << error.message() << " (" << type << ")";
return error;
}
}
// Remove unused channels if MediaContentDescription is rejected.
RemoveUnusedChannels(local_description()->description());
}
error = UpdateSessionState(type, CS_LOCAL, local_description()->description(),
bundle_groups_by_mid);
if (!error.ok()) {
RTC_LOG(LS_ERROR) << error.message() << " (" << type << ")";
return error;
}
// Now that we have a local description, we can push down remote candidates.
UseCandidatesInRemoteDescription();
pending_ice_restarts_.clear();
if (session_error() != SessionError::kNone) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
AllocateSctpSids();
// Validate SSRCs, we do not allow duplicates.
if (ConfiguredForMedia()) {
std::set<uint32_t> used_ssrcs;
for (const auto& content : local_description()->description()->contents()) {
for (const auto& stream : content.media_description()->streams()) {
for (uint32_t ssrc : stream.ssrcs) {
auto result = used_ssrcs.insert(ssrc);
if (!result.second) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Duplicate ssrc " + absl::StrCat(ssrc) + " is not allowed");
}
}
}
}
}
if (IsUnifiedPlan()) {
if (ConfiguredForMedia()) {
// We must use List and not ListInternal here because
// transceivers()->StableState() is indexed by the non-internal refptr.
for (const auto& transceiver_ext : transceivers()->List()) {
auto transceiver = transceiver_ext->internal();
if (transceiver->stopped()) {
continue;
}
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, local_description());
if (!content) {
continue;
}
ChannelInterface* channel = transceiver->channel();
if (content->rejected || !channel || channel->local_streams().empty()) {
// 0 is a special value meaning "this sender has no associated send
// stream". Need to call this so the sender won't attempt to configure
// a no longer existing stream and run into DCHECKs in the lower
// layers.
transceiver->sender_internal()->SetSsrc(0);
} else {
// Get the StreamParams from the channel which could generate SSRCs.
const std::vector<StreamParams>& streams = channel->local_streams();
transceiver->sender_internal()->set_stream_ids(
streams[0].stream_ids());
auto encodings =
transceiver->sender_internal()->init_send_encodings();
transceiver->sender_internal()->SetSsrc(streams[0].first_ssrc());
if (!encodings.empty()) {
transceivers()
->StableState(transceiver_ext)
->SetInitSendEncodings(encodings);
}
}
}
}
} else {
// Plan B semantics.
// Update state and SSRC of local MediaStreams and DataChannels based on the
// local session description.
const ContentInfo* audio_content =
GetFirstAudioContent(local_description()->description());
if (audio_content) {
if (audio_content->rejected) {
RemoveSenders(MediaType::AUDIO);
} else {
const MediaContentDescription* audio_desc =
audio_content->media_description();
UpdateLocalSenders(audio_desc->streams(), audio_desc->type());
}
}
const ContentInfo* video_content =
GetFirstVideoContent(local_description()->description());
if (video_content) {
if (video_content->rejected) {
RemoveSenders(MediaType::VIDEO);
} else {
const MediaContentDescription* video_desc =
video_content->media_description();
UpdateLocalSenders(video_desc->streams(), video_desc->type());
}
}
}
// This function does nothing with data content.
if (type == SdpType::kAnswer &&
local_ice_credentials_to_replace_->SatisfiesIceRestart(
*current_local_description_)) {
local_ice_credentials_to_replace_->ClearIceCredentials();
}
if (IsUnifiedPlan()) {
UpdateRtpHeaderExtensionPreferencesFromSdpMunging(
local_description()->description(), transceivers());
}
return RTCError::OK();
}
void SdpOfferAnswerHandler::SetRemoteDescription(
SetSessionDescriptionObserver* observer,
SessionDescriptionInterface* desc_ptr) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(observer);
RTC_DCHECK(desc_ptr);
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
// lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
observer_refptr = scoped_refptr<SetSessionDescriptionObserver>(observer),
desc = std::unique_ptr<SessionDescriptionInterface>(desc_ptr)](
std::function<void()> operations_chain_callback) mutable {
// Abort early if `this_weak_ptr` is no longer valid.
if (!this_weak_ptr) {
// For consistency with SetSessionDescriptionObserverAdapter whose
// posted messages doesn't get processed when the PC is destroyed, we
// do not inform `observer_refptr` that the operation failed.
operations_chain_callback();
return;
}
// SetSessionDescriptionObserverAdapter takes care of making sure the
// `observer_refptr` is invoked in a posted message.
this_weak_ptr->DoSetRemoteDescription(
std::make_unique<RemoteDescriptionOperation>(
this_weak_ptr.get(), std::move(desc),
make_ref_counted<SetSessionDescriptionObserverAdapter>(
this_weak_ptr, observer_refptr),
std::move(operations_chain_callback)));
});
}
void SdpOfferAnswerHandler::SetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(desc);
RTC_DCHECK(observer);
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
// lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(), observer,
desc = std::move(desc)](
std::function<void()> operations_chain_callback) mutable {
// Abort early if `this_weak_ptr` is no longer valid.
if (!this_weak_ptr) {
observer->OnSetRemoteDescriptionComplete(RTCError(
RTCErrorType::INTERNAL_ERROR,
"SetRemoteDescription failed because the session was shut down"));
operations_chain_callback();
return;
}
this_weak_ptr->DoSetRemoteDescription(
std::make_unique<RemoteDescriptionOperation>(
this_weak_ptr.get(), std::move(desc), std::move(observer),
std::move(operations_chain_callback)));
});
}
RTCError SdpOfferAnswerHandler::ReplaceRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
SdpType sdp_type,
std::unique_ptr<SessionDescriptionInterface>* replaced_description) {
RTC_DCHECK(replaced_description);
if (sdp_type == SdpType::kAnswer) {
*replaced_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_description = std::move(pending_remote_description_);
pending_remote_description_ = std::move(desc);
}
// The session description to apply now must be accessed by
// `remote_description()`.
const SessionDescription* session_desc = remote_description()->description();
const auto* local = local_description();
// NOTE: This will perform a BlockingCall() to the network thread.
return transport_controller_s()->SetRemoteDescription(
sdp_type, local ? local->description() : nullptr, session_desc);
}
void SdpOfferAnswerHandler::ApplyRemoteDescription(
std::unique_ptr<RemoteDescriptionOperation> operation) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ApplyRemoteDescription");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(operation->description());
// Invalidate the stats caches to make sure that they get
// updated next time getStats() gets called, as updating the session
// description affects the stats.
pc_->ClearStatsCache();
if (!operation->ReplaceRemoteDescriptionAndCheckError())
return;
if (!operation->UpdateChannels())
return;
// NOTE: Candidates allocation will be initiated only when
// SetLocalDescription is called.
if (!operation->UpdateSessionState())
return;
if (!operation->UseCandidatesInRemoteDescription())
return;
if (operation->old_remote_description()) {
for (const ContentInfo& content :
operation->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(operation->old_remote_description(),
remote_description(), content.mid())) {
if (operation->type() == SdpType::kOffer) {
pending_ice_restarts_.insert(content.mid());
}
} 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(
operation->old_remote_description(), content.mid(),
mutable_remote_description());
}
}
}
if (operation->HaveSessionError())
return;
// Set the the ICE connection state to connecting since the connection may
// become writable with peer reflexive candidates before any remote candidate
// is signaled.
// TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
// is to have a new signal the indicates a change in checking state from the
// transport and expose a new checking() member from transport that can be
// read to determine the current checking state. The existing SignalConnecting
// actually means "gathering candidates", so cannot be be used here.
if (remote_description()->GetType() != SdpType::kOffer &&
remote_description()->number_of_mediasections() > 0u &&
pc_->ice_connection_state_internal() ==
PeerConnectionInterface::kIceConnectionNew) {
pc_->SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
}
// If setting the description decided our SSL role, allocate any necessary
// SCTP sids.
AllocateSctpSids();
if (operation->unified_plan()) {
ApplyRemoteDescriptionUpdateTransceiverState(operation->type());
}
remote_peer_supports_msid_ =
remote_description()->description()->msid_signaling() !=
kMsidSignalingNotUsed;
if (!operation->unified_plan()) {
PlanBUpdateSendersAndReceivers(
GetFirstAudioContent(remote_description()->description()),
GetFirstAudioContentDescription(remote_description()->description()),
GetFirstVideoContent(remote_description()->description()),
GetFirstVideoContentDescription(remote_description()->description()));
}
if (operation->type() == SdpType::kAnswer) {
if (local_ice_credentials_to_replace_->SatisfiesIceRestart(
*current_local_description_)) {
local_ice_credentials_to_replace_->ClearIceCredentials();
}
RemoveStoppedTransceivers();
}
// Consider the operation complete at this point.
operation->SignalCompletion();
SetRemoteDescriptionPostProcess(operation->type() == SdpType::kAnswer);
}
void SdpOfferAnswerHandler::ApplyRemoteDescriptionUpdateTransceiverState(
SdpType sdp_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
if (!ConfiguredForMedia()) {
return;
}
std::vector<scoped_refptr<RtpTransceiverInterface>>
now_receiving_transceivers;
std::vector<scoped_refptr<RtpTransceiverInterface>> remove_list;
std::vector<scoped_refptr<MediaStreamInterface>> added_streams;
std::vector<scoped_refptr<MediaStreamInterface>> removed_streams;
for (const auto& transceiver_ext : transceivers()->List()) {
const auto transceiver = transceiver_ext->internal();
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, remote_description());
if (!content) {
continue;
}
const MediaContentDescription* media_desc = content->media_description();
RtpTransceiverDirection local_direction =
RtpTransceiverDirectionReversed(media_desc->direction());
// Remember the previous remote streams if this is a remote offer. This
// makes it possible to rollback modifications to the streams.
if (sdp_type == SdpType::kOffer) {
transceivers()
->StableState(transceiver_ext)
->SetRemoteStreamIds(transceiver->receiver()->stream_ids());
}
// Roughly the same as steps 2.2.8.6 of section 4.4.1.6 "Set the
// RTCSessionDescription: Set the associated remote streams given
// transceiver.[[Receiver]], msids, addList, and removeList".
// https://w3c.github.io/webrtc-pc/#set-the-rtcsessiondescription
if (RtpTransceiverDirectionHasRecv(local_direction)) {
std::vector<std::string> stream_ids;
if (!media_desc->streams().empty()) {
// The remote description has signaled the stream IDs.
stream_ids = media_desc->streams()[0].stream_ids();
}
RTC_LOG(LS_INFO) << "Processing the MSIDs for MID=" << content->mid()
<< " (" << GetStreamIdsString(stream_ids) << ").";
SetAssociatedRemoteStreams(transceiver->receiver_internal(), stream_ids,
&added_streams, &removed_streams);
// From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6
// "Set the RTCSessionDescription: If direction is sendrecv or recvonly,
// and transceiver's current direction is neither sendrecv nor recvonly,
// process the addition of a remote track for the media description.
if (!transceiver->fired_direction() ||
!RtpTransceiverDirectionHasRecv(*transceiver->fired_direction())) {
RTC_LOG(LS_INFO) << "Processing the addition of a remote track for MID="
<< content->mid() << ".";
// Since the transceiver is passed to the user in an
// OnTrack event, we must use the proxied transceiver.
now_receiving_transceivers.push_back(transceiver_ext);
}
}
// 2.2.8.1.9: If direction is "sendonly" or "inactive", and transceiver's
// [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the
// removal of a remote track for the media description, given transceiver,
// removeList, and muteTracks.
if (!RtpTransceiverDirectionHasRecv(local_direction) &&
(transceiver->fired_direction() &&
RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
ProcessRemovalOfRemoteTrack(transceiver_ext, &remove_list,
&removed_streams);
}
// 2.2.8.1.10: Set transceiver's [[FiredDirection]] slot to direction.
if (sdp_type == SdpType::kOffer) {
// Remember the previous fired direction if this is a remote offer. This
// makes it possible to rollback modifications to [[FiredDirection]],
// which is necessary for "ontrack" to fire in or after rollback.
transceivers()
->StableState(transceiver_ext)
->SetFiredDirection(transceiver->fired_direction());
}
transceiver->set_fired_direction(local_direction);
// 2.2.8.1.11: If description is of type "answer" or "pranswer", then run
// the following steps:
if (sdp_type == SdpType::kPrAnswer || sdp_type == SdpType::kAnswer) {
// 2.2.8.1.11.1: Set transceiver's [[CurrentDirection]] slot to
// direction.
transceiver->set_current_direction(local_direction);
// 2.2.8.1.11.[3-6]: Set the transport internal slots.
if (transceiver->mid()) {
auto dtls_transport = LookupDtlsTransportByMid(
context_->network_thread(), transport_controller_s(),
*transceiver->mid());
transceiver->sender_internal()->set_transport(dtls_transport);
transceiver->receiver_internal()->set_transport(dtls_transport);
}
}
// 2.2.8.1.12: If the media description is rejected, and transceiver is
// not already stopped, stop the RTCRtpTransceiver transceiver.
if (content->rejected && !transceiver->stopped()) {
RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->mid()
<< " since the media section was rejected.";
transceiver->StopTransceiverProcedure();
}
if (!content->rejected && RtpTransceiverDirectionHasRecv(local_direction)) {
if (!media_desc->streams().empty() &&
media_desc->streams()[0].has_ssrcs()) {
uint32_t ssrc = media_desc->streams()[0].first_ssrc();
transceiver->receiver_internal()->SetupMediaChannel(ssrc);
} else {
transceiver->receiver_internal()->SetupUnsignaledMediaChannel();
}
}
}
// Once all processing has finished, fire off callbacks.
pc_->RunWithObserver([&](auto observer) {
for (const auto& transceiver : now_receiving_transceivers) {
pc_->legacy_stats()->AddTrack(transceiver->receiver()->track().get());
observer->OnTrack(transceiver);
observer->OnAddTrack(transceiver->receiver(),
transceiver->receiver()->streams());
}
for (const auto& stream : added_streams) {
observer->OnAddStream(stream);
}
for (const auto& transceiver : remove_list) {
observer->OnRemoveTrack(transceiver->receiver());
}
for (const auto& stream : removed_streams) {
observer->OnRemoveStream(stream);
}
});
}
void SdpOfferAnswerHandler::PlanBUpdateSendersAndReceivers(
const ContentInfo* audio_content,
const AudioContentDescription* audio_desc,
const ContentInfo* video_content,
const VideoContentDescription* video_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsUnifiedPlan());
// We wait to signal new streams until we finish processing the description,
// since only at that point will new streams have all their tracks.
scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
// TODO(steveanton): When removing RTP senders/receivers in response to a
// rejected media section, there is some cleanup logic that expects the
// voice/ video channel to still be set. But in this method the voice/video
// channel would have been destroyed by the SetRemoteDescription caller
// above so the cleanup that relies on them fails to run. The RemoveSenders
// calls should be moved to right before the DestroyChannel calls to fix
// this.
// Find all audio rtp streams and create corresponding remote AudioTracks
// and MediaStreams.
if (audio_content) {
if (audio_content->rejected) {
RemoveSenders(MediaType::AUDIO);
} else {
bool default_audio_track_needed =
!remote_peer_supports_msid_ &&
RtpTransceiverDirectionHasSend(audio_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(audio_desc),
default_audio_track_needed, audio_desc->type(),
new_streams.get());
}
}
// Find all video rtp streams and create corresponding remote VideoTracks
// and MediaStreams.
if (video_content) {
if (video_content->rejected) {
RemoveSenders(MediaType::VIDEO);
} else {
bool default_video_track_needed =
!remote_peer_supports_msid_ &&
RtpTransceiverDirectionHasSend(video_desc->direction());
UpdateRemoteSendersList(GetActiveStreams(video_desc),
default_video_track_needed, video_desc->type(),
new_streams.get());
}
}
// Iterate new_streams and notify the observer about new MediaStreams.
pc_->RunWithObserver([&](auto observer) {
for (size_t i = 0; i < new_streams->count(); ++i) {
MediaStreamInterface* new_stream = new_streams->at(i);
pc_->legacy_stats()->AddStream(new_stream);
observer->OnAddStream(scoped_refptr<MediaStreamInterface>(new_stream));
}
});
UpdateEndedRemoteMediaStreams();
}
void SdpOfferAnswerHandler::ReportInitialSdpMunging(bool had_local_description,
SdpType type) {
// Report SDP munging of the initial call to setLocalDescription separately.
if (!had_local_description) {
switch (type) {
case SdpType::kOffer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.Offer.Initial",
last_sdp_munging_type_, SdpMungingType::kMaxValue);
break;
case SdpType::kAnswer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.Answer.Initial",
last_sdp_munging_type_, SdpMungingType::kMaxValue);
break;
case SdpType::kPrAnswer:
RTC_HISTOGRAM_ENUMERATION(
"WebRTC.PeerConnection.SdpMunging.PrAnswer.Initial",
last_sdp_munging_type_, SdpMungingType::kMaxValue);
break;
case SdpType::kRollback:
// Rollback does not have SDP so can not be munged.
break;
}
}
}
void SdpOfferAnswerHandler::DoSetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoSetLocalDescription");
RTC_DCHECK(desc);
RTC_DCHECK(observer);
// If a session error has occurred the PeerConnection is in a possibly
// inconsistent state so fail right away.
if (session_error() != SessionError::kNone) {
std::string error_message = GetSessionErrorMsg();
RTC_LOG(LS_ERROR) << "SetLocalDescription: " << error_message;
observer->OnSetLocalDescriptionComplete(
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
// For SLD we support only explicit rollback.
if (desc->GetType() == SdpType::kRollback) {
if (IsUnifiedPlan()) {
observer->OnSetLocalDescriptionComplete(Rollback(desc->GetType()));
} else {
observer->OnSetLocalDescriptionComplete(
RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
"Rollback not supported in Plan B"));
}
return;
}
std::map<std::string, const ContentGroup*> bundle_groups_by_mid =
GetBundleGroupsByMid(desc->description());
RTCError error =
ValidateSessionDescription(desc.get(), CS_LOCAL, bundle_groups_by_mid);
if (!error.ok()) {
std::string error_message =
GetSetDescriptionErrorMessage(CS_LOCAL, desc->GetType(), error);
RTC_LOG(LS_ERROR) << error_message;
observer->OnSetLocalDescriptionComplete(
RTCError(error.type(), std::move(error_message)));
return;
}
SessionDescriptionInterface* last_created_desc =
desc->GetType() == SdpType::kOffer ? last_created_offer_.get()
: last_created_answer_.get();
// Determine if SDP munging was done. This is not yet acted upon.
bool had_local_description = !!local_description();
SdpMungingType sdp_munging_type =
DetermineSdpMungingType(desc.get(), last_created_desc);
if (!disable_sdp_munging_checks_) {
bool reject_error = false;
if (HasUfragSdpMunging(desc.get(), last_created_desc)) {
has_sdp_munged_ufrag_ = true;
if (pc_->trials().IsEnabled("WebRTC-NoSdpMangleUfrag")) {
RTC_LOG(LS_ERROR) << "Rejecting SDP because of ufrag modification";
reject_error = true;
}
} else {
reject_error = !IsSdpMungingAllowed(sdp_munging_type, pc_->trials());
}
SdpMungingOutcome outcome = reject_error ? SdpMungingOutcome::kRejected
: SdpMungingOutcome::kAccepted;
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpMunging.Outcome",
static_cast<int>(outcome),
static_cast<int>(SdpMungingOutcome::kMaxValue));
if (reject_error) {
observer->OnSetLocalDescriptionComplete(
RTCError(RTCErrorType::INVALID_MODIFICATION,
"SDP is modified in a non-acceptable way"));
last_sdp_munging_type_ = sdp_munging_type;
ReportInitialSdpMunging(had_local_description, desc->GetType());
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SdpMunging.SdpOutcome.Rejected",
sdp_munging_type, SdpMungingType::kMaxValue);
return;
}
if (sdp_munging_type != kNoModification) {
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.SdpMunging.SdpOutcome.Accepted",
sdp_munging_type, SdpMungingType::kMaxValue);
}
}
// Workaround for isses.webrtc.org/412904801 - detect if packetization:raw
// was added in SDP munging, and if so, push it back into the codec vendor.
std::vector<std::pair<Codec, Codec>> codecs_mangled_to_raw;
// Ignore if other mungings have been applied.
if (last_created_desc &&
last_created_desc->description()->contents().size() ==
desc->description()->contents().size()) {
for (size_t i = 0; i < desc->description()->contents().size(); ++i) {
auto last_created_codecs = last_created_desc->description()
->contents()[i]
.media_description()
->codecs();
auto new_codecs =
desc->description()->contents()[i].media_description()->codecs();
if (last_created_codecs.size() != new_codecs.size()) {
continue; // Ignore added codecs
}
for (size_t j = 0; j < new_codecs.size(); ++j) {
if (new_codecs[j].type == Codec::Type::kVideo &&
new_codecs[j].name == last_created_codecs[j].name &&
new_codecs[j].id == last_created_codecs[j].id &&
!last_created_codecs[j].packetization &&
new_codecs[j].packetization == kPacketizationParamRaw) {
codecs_mangled_to_raw.push_back(
std::pair(last_created_codecs[j], new_codecs[j]));
}
}
}
}
// Grab the description type before moving ownership to
// ApplyLocalDescription, which may destroy it before returning.
const SdpType type = desc->GetType();
error = ApplyLocalDescription(std::move(desc), bundle_groups_by_mid);
// `desc` may be destroyed at this point.
if (!error.ok()) {
// If ApplyLocalDescription fails, the PeerConnection could be in an
// inconsistent state, so act conservatively here and set the session error
// so that future calls to SetLocalDescription/SetRemoteDescription fail.
SetSessionError(SessionError::kContent, error.message());
std::string error_message =
GetSetDescriptionErrorMessage(CS_LOCAL, type, error);
RTC_LOG(LS_ERROR) << error_message;
observer->OnSetLocalDescriptionComplete(
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
RTC_DCHECK(local_description());
// Workaround for issues.webrtc.org/412904801 part 2.
// If application was successful, we change the codec vendor's codec
// tables according to the mangle.
// Note that this depends on there being a single codec vendor for all MIDs.
codec_lookup_helper_->GetCodecVendor()->ModifyVideoCodecs(
codecs_mangled_to_raw);
if (local_description()->GetType() == SdpType::kAnswer) {
RemoveStoppedTransceivers();
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
context_->network_thread()->BlockingCall(
[this] { port_allocator()->DiscardCandidatePool(); });
}
// Clear last created offer/answer and update SDP munging type.
last_created_offer_.reset(nullptr);
last_created_answer_.reset(nullptr);
last_sdp_munging_type_ = sdp_munging_type;
// Report SDP munging of the initial call to setLocalDescription separately.
ReportInitialSdpMunging(had_local_description,
local_description()->GetType());
observer->OnSetLocalDescriptionComplete(RTCError::OK());
pc_->NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED);
// Check if negotiation is needed. We must do this after informing the
// observer that SetLocalDescription() has completed to ensure negotiation
// is not needed prior to the promise resolving.
if (IsUnifiedPlan()) {
bool was_negotiation_needed = is_negotiation_needed_;
UpdateNegotiationNeeded();
if (signaling_state() == PeerConnectionInterface::kStable &&
was_negotiation_needed && is_negotiation_needed_) {
// Legacy version.
pc_->RunWithObserver(
[&](auto observer) { observer->OnRenegotiationNeeded(); });
// Spec-compliant version; the event may get invalidated before firing.
GenerateNegotiationNeededEvent();
}
}
// MaybeStartGathering needs to be called after informing the observer so
// that we don't signal any candidates before signaling that
// SetLocalDescription completed.
transport_controller_s()->MaybeStartGathering();
}
void SdpOfferAnswerHandler::DoCreateOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
scoped_refptr<CreateSessionDescriptionObserver> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoCreateOffer");
if (!observer) {
RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
return;
}
if (pc_->IsClosed()) {
std::string error = "CreateOffer called when PeerConnection is closed.";
RTC_LOG(LS_ERROR) << error;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer.get(),
RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
return;
}
// If a session error has occurred the PeerConnection is in a possibly
// inconsistent state so fail right away.
if (session_error() != SessionError::kNone) {
std::string error_message = GetSessionErrorMsg();
RTC_LOG(LS_ERROR) << "CreateOffer: " << error_message;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer.get(),
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
if (!ValidateOfferAnswerOptions(options)) {
std::string error = "CreateOffer called with invalid options.";
RTC_LOG(LS_ERROR) << error;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer.get(),
RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
return;
}
// Legacy handling for offer_to_receive_audio and offer_to_receive_video.
// Specified in WebRTC section 4.4.3.2 "Legacy configuration extensions".
if (IsUnifiedPlan()) {
RTCError error = HandleLegacyOfferOptions(options);
if (!error.ok()) {
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer.get(), std::move(error));
return;
}
}
MediaSessionOptions session_options;
GetOptionsForOffer(options, &session_options);
auto observer_wrapper =
make_ref_counted<CreateDescriptionObserverWrapperWithCreationCallback>(
[this](std::unique_ptr<SessionDescriptionInterface> desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
last_created_offer_ = std::move(desc);
},
std::move(observer));
webrtc_session_desc_factory_->CreateOffer(observer_wrapper.get(), options,
session_options);
}
void SdpOfferAnswerHandler::CreateAnswer(
CreateSessionDescriptionObserver* observer,
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateAnswer");
RTC_DCHECK_RUN_ON(signaling_thread());
// Chain this operation. If asynchronous operations are pending on the chain,
// this operation will be queued to be invoked, otherwise the contents of the
// lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
observer_refptr =
scoped_refptr<CreateSessionDescriptionObserver>(observer),
options](std::function<void()> operations_chain_callback) {
// Abort early if `this_weak_ptr` is no longer valid.
if (!this_weak_ptr) {
observer_refptr->OnFailure(RTCError(
RTCErrorType::INTERNAL_ERROR,
"CreateAnswer failed because the session was shut down"));
operations_chain_callback();
return;
}
// The operation completes asynchronously when the wrapper is invoked.
auto observer_wrapper =
make_ref_counted<CreateSessionDescriptionObserverOperationWrapper>(
std::move(observer_refptr),
std::move(operations_chain_callback));
this_weak_ptr->DoCreateAnswer(options, observer_wrapper);
});
}
void SdpOfferAnswerHandler::DoCreateAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& options,
scoped_refptr<CreateSessionDescriptionObserver> observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoCreateAnswer");
if (!observer) {
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
return;
}
// If a session error has occurred the PeerConnection is in a possibly
// inconsistent state so fail right away.
if (session_error() != SessionError::kNone) {
std::string error_message = GetSessionErrorMsg();
RTC_LOG(LS_ERROR) << "CreateAnswer: " << error_message;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer.get(),
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
return;
}
if (!(signaling_state_ == PeerConnectionInterface::kHaveRemoteOffer ||
signaling_state_ == PeerConnectionInterface::kHaveLocalPrAnswer)) {
std::string error =
"PeerConnection cannot create an answer in a state other than "
"have-remote-offer or have-local-pranswer.";
RTC_LOG(LS_ERROR) << error;
pc_->message_handler()->PostCreateSessionDescriptionFailure(
observer.get(),
RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
return;
}
// The remote description should be set if we're in the right state.
RTC_DCHECK(remote_description());
if (IsUnifiedPlan()) {
if (options.offer_to_receive_audio !=
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
"supported with Unified Plan semantics. Use the "
"RtpTransceiver API instead.";
}
if (options.offer_to_receive_video !=
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
"supported with Unified Plan semantics. Use the "
"RtpTransceiver API instead.";
}
}
MediaSessionOptions session_options;
GetOptionsForAnswer(options, &session_options);
auto observer_wrapper =
make_ref_counted<CreateDescriptionObserverWrapperWithCreationCallback>(
[this](std::unique_ptr<SessionDescriptionInterface> desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
last_created_answer_ = std::move(desc);
},
std::move(observer));
webrtc_session_desc_factory_->CreateAnswer(observer_wrapper.get(),
session_options);
}
void SdpOfferAnswerHandler::DoSetRemoteDescription(
std::unique_ptr<RemoteDescriptionOperation> operation) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoSetRemoteDescription");
if (!operation->ok())
return;
if (operation->HaveSessionError())
return;
if (operation->MaybeRollback())
return;
operation->ReportOfferAnswerUma();
// Handle remote descriptions missing a=mid lines for interop with legacy
// end points.
FillInMissingRemoteMids(operation->description());
if (!operation->IsDescriptionValid())
return;
ApplyRemoteDescription(std::move(operation));
}
// Called after a DoSetRemoteDescription operation completes.
void SdpOfferAnswerHandler::SetRemoteDescriptionPostProcess(bool was_answer) {
RTC_DCHECK(remote_description());
if (was_answer) {
// TODO(deadbeef): We already had to hop to the network thread for
// MaybeStartGathering...
context_->network_thread()->BlockingCall(
[this] { port_allocator()->DiscardCandidatePool(); });
}
pc_->NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_SUCCEEDED);
// Check if negotiation is needed. We must do this after informing the
// observer that SetRemoteDescription() has completed to ensure negotiation
// is not needed prior to the promise resolving.
if (IsUnifiedPlan()) {
bool was_negotiation_needed = is_negotiation_needed_;
UpdateNegotiationNeeded();
if (signaling_state() == PeerConnectionInterface::kStable &&
was_negotiation_needed && is_negotiation_needed_) {
// Legacy version.
pc_->RunWithObserver(
[&](auto observer) { observer->OnRenegotiationNeeded(); });
// Spec-compliant version; the event may get invalidated before firing.
GenerateNegotiationNeededEvent();
}
}
}
void SdpOfferAnswerHandler::SetAssociatedRemoteStreams(
scoped_refptr<RtpReceiverInternal> receiver,
const std::vector<std::string>& stream_ids,
std::vector<scoped_refptr<MediaStreamInterface>>* added_streams,
std::vector<scoped_refptr<MediaStreamInterface>>* removed_streams) {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<scoped_refptr<MediaStreamInterface>> media_streams;
for (const std::string& stream_id : stream_ids) {
scoped_refptr<MediaStreamInterface> stream(
remote_streams_->find(stream_id));
if (!stream) {
stream = MediaStreamProxy::Create(Thread::Current(),
MediaStream::Create(stream_id));
remote_streams_->AddStream(stream);
added_streams->push_back(stream);
}
media_streams.push_back(stream);
}
// Special case: "a=msid" missing, use random stream ID.
if (media_streams.empty() &&
!(remote_description()->description()->msid_signaling() &
kMsidSignalingMediaSection)) {
if (!missing_msid_default_stream_) {
missing_msid_default_stream_ = MediaStreamProxy::Create(
Thread::Current(), MediaStream::Create(CreateRandomUuid()));
added_streams->push_back(missing_msid_default_stream_);
}
media_streams.push_back(missing_msid_default_stream_);
}
std::vector<scoped_refptr<MediaStreamInterface>> previous_streams =
receiver->streams();
// SetStreams() will add/remove the receiver's track to/from the streams.
// This differs from the spec - the spec uses an "addList" and "removeList"
// to update the stream-track relationships in a later step. We do this
// earlier, changing the order of things, but the end-result is the same.
// TODO(hbos): When we remove remote_streams(), use set_stream_ids()
// instead. https://crbug.com/webrtc/9480
receiver->SetStreams(media_streams);
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
}
bool SdpOfferAnswerHandler::AddIceCandidate(const IceCandidate* ice_candidate) {
const AddIceCandidateResult result = AddIceCandidateInternal(ice_candidate);
NoteAddIceCandidateResult(result);
// If the return value is kAddIceCandidateFailNotReady, the candidate has
// been added, although not 'ready', but that's a success.
return result == kAddIceCandidateSuccess ||
result == kAddIceCandidateFailNotReady;
}
AddIceCandidateResult SdpOfferAnswerHandler::AddIceCandidateInternal(
const IceCandidate* ice_candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AddIceCandidate");
if (pc_->IsClosed()) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: PeerConnection is closed.";
return kAddIceCandidateFailClosed;
}
if (!remote_description()) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: ICE candidates can't be added "
"without any remote session description.";
return kAddIceCandidateFailNoRemoteDescription;
}
if (!ice_candidate) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate is null.";
return kAddIceCandidateFailNullCandidate;
}
bool valid = false;
bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
if (!valid) {
return kAddIceCandidateFailNotValid;
}
// Add this candidate to the remote session description.
if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate cannot be used.";
return kAddIceCandidateFailInAddition;
}
if (!ready) {
RTC_LOG(LS_INFO) << "AddIceCandidate: Not ready to use candidate.";
return kAddIceCandidateFailNotReady;
}
if (!UseCandidate(ice_candidate)) {
return kAddIceCandidateFailNotUsable;
}
pc_->NoteUsageEvent(UsageEvent::ADD_ICE_CANDIDATE_SUCCEEDED);
return kAddIceCandidateSuccess;
}
void SdpOfferAnswerHandler::AddIceCandidate(
std::unique_ptr<IceCandidate> candidate,
std::function<void(RTCError)> callback) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AddIceCandidate");
RTC_DCHECK_RUN_ON(signaling_thread());
// Chain this operation. If asynchronous operations are pending on the
// chain, this operation will be queued to be invoked, otherwise the
// contents of the lambda will execute immediately.
operations_chain_->ChainOperation(
[this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
candidate = std::move(candidate), callback = std::move(callback)](
std::function<void()> operations_chain_callback) {
auto result =
this_weak_ptr
? this_weak_ptr->AddIceCandidateInternal(candidate.get())
: kAddIceCandidateFailClosed;
NoteAddIceCandidateResult(result);
operations_chain_callback();
switch (result) {
case AddIceCandidateResult::kAddIceCandidateSuccess:
case AddIceCandidateResult::kAddIceCandidateFailNotReady:
// Success!
callback(RTCError::OK());
break;
case AddIceCandidateResult::kAddIceCandidateFailClosed:
// Note that the spec says to just abort without resolving the
// promise in this case, but this layer must return an RTCError.
callback(RTCError(
RTCErrorType::INVALID_STATE,
"AddIceCandidate failed because the session was shut down"));
break;
case AddIceCandidateResult::kAddIceCandidateFailNoRemoteDescription:
// Spec: "If remoteDescription is null return a promise rejected
// with a newly created InvalidStateError."
callback(RTCError(RTCErrorType::INVALID_STATE,
"The remote description was null"));
break;
case AddIceCandidateResult::kAddIceCandidateFailNullCandidate:
// TODO(https://crbug.com/935898): Handle end-of-candidates instead
// of treating null candidate as an error.
callback(RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
"Error processing ICE candidate"));
break;
case AddIceCandidateResult::kAddIceCandidateFailNotValid:
case AddIceCandidateResult::kAddIceCandidateFailInAddition:
case AddIceCandidateResult::kAddIceCandidateFailNotUsable:
// Spec: "If candidate could not be successfully added [...] Reject
// p with a newly created OperationError and abort these steps."
// UNSUPPORTED_OPERATION maps to OperationError.
callback(RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
"Error processing ICE candidate"));
break;
default:
RTC_DCHECK_NOTREACHED();
}
});
}
bool SdpOfferAnswerHandler::RemoveIceCandidate(const IceCandidate* candidate) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::RemoveIceCandidate");
RTC_DCHECK_RUN_ON(signaling_thread());
if (pc_->IsClosed() || !remote_description() || !candidate) {
RTC_LOG(LS_ERROR) << "RemoveIceCandidate: PeerConnection is closed.";
return false;
}
bool removed = mutable_remote_description()->RemoveCandidate(candidate);
// Remove the candidates from the transport controller.
// This involves a hop to the network thread - should we rather do this
// asynchronously?
transport_controller_s()->RemoveRemoteCandidate(candidate);
return removed;
}
void SdpOfferAnswerHandler::AddLocalIceCandidate(
const IceCandidate* candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (local_description()) {
mutable_local_description()->AddCandidate(candidate);
}
}
void SdpOfferAnswerHandler::RemoveLocalIceCandidates(
absl::string_view mid,
const std::vector<Candidate>& candidates) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (local_description()) {
for (const auto& c : candidates) {
IceCandidate ice_candidate(mid, -1, c);
mutable_local_description()->RemoveCandidate(&ice_candidate);
}
}
}
const SessionDescriptionInterface* SdpOfferAnswerHandler::local_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_local_description_ ? pending_local_description_.get()
: current_local_description_.get();
}
const SessionDescriptionInterface* SdpOfferAnswerHandler::remote_description()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_remote_description_ ? pending_remote_description_.get()
: current_remote_description_.get();
}
const SessionDescriptionInterface*
SdpOfferAnswerHandler::current_local_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return current_local_description_.get();
}
const SessionDescriptionInterface*
SdpOfferAnswerHandler::current_remote_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return current_remote_description_.get();
}
const SessionDescriptionInterface*
SdpOfferAnswerHandler::pending_local_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_local_description_.get();
}
const SessionDescriptionInterface*
SdpOfferAnswerHandler::pending_remote_description() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_remote_description_.get();
}
PeerConnectionInterface::SignalingState SdpOfferAnswerHandler::signaling_state()
const {
RTC_DCHECK_RUN_ON(signaling_thread());
return signaling_state_;
}
void SdpOfferAnswerHandler::ChangeSignalingState(
PeerConnectionInterface::SignalingState signaling_state) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::ChangeSignalingState");
RTC_DCHECK_RUN_ON(signaling_thread());
if (signaling_state_ == signaling_state) {
return;
}
RTC_LOG(LS_INFO) << "Session: " << pc_->session_id() << " Old state: "
<< PeerConnectionInterface::AsString(signaling_state_)
<< " New state: "
<< PeerConnectionInterface::AsString(signaling_state);
signaling_state_ = signaling_state;
pc_->RunWithObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
observer->OnSignalingChange(signaling_state_);
});
}
RTCError SdpOfferAnswerHandler::UpdateSessionState(
SdpType type,
ContentSource source,
const SessionDescription* description,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
RTC_DCHECK_RUN_ON(signaling_thread());
// If there's already a pending error then no state transition should
// happen. But all call-sites should be verifying this before calling us!
RTC_DCHECK(session_error() == SessionError::kNone);
// If this is answer-ish we're ready to let media flow.
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
EnableSending();
}
// Update the signaling state according to the specified state machine (see
// https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum).
if (type == SdpType::kOffer) {
ChangeSignalingState(source == CS_LOCAL
? PeerConnectionInterface::kHaveLocalOffer
: PeerConnectionInterface::kHaveRemoteOffer);
} else if (type == SdpType::kPrAnswer) {
ChangeSignalingState(source == CS_LOCAL
? PeerConnectionInterface::kHaveLocalPrAnswer
: PeerConnectionInterface::kHaveRemotePrAnswer);
} else {
RTC_DCHECK(type == SdpType::kAnswer);
ChangeSignalingState(PeerConnectionInterface::kStable);
if (ConfiguredForMedia()) {
transceivers()->DiscardStableStates();
}
}
// Update internal objects according to the session description's media
// descriptions.
return PushdownMediaDescription(type, source, bundle_groups_by_mid);
}
bool SdpOfferAnswerHandler::ShouldFireNegotiationNeededEvent(
uint32_t event_id) {
RTC_DCHECK_RUN_ON(signaling_thread());
// Plan B? Always fire to conform with useless legacy behavior.
if (!IsUnifiedPlan()) {
return true;
}
// The event ID has been invalidated. Either negotiation is no longer needed
// or a newer negotiation needed event has been generated.
if (event_id != negotiation_needed_event_id_) {
return false;
}
// The chain is no longer empty, update negotiation needed when it becomes
// empty. This should generate a newer negotiation needed event, making this
// one obsolete.
if (!operations_chain_->IsEmpty()) {
// Since we just suppressed an event that would have been fired, if
// negotiation is still needed by the time the chain becomes empty again,
// we must make sure to generate another event if negotiation is needed
// then. This happens when `is_negotiation_needed_` goes from false to
// true, so we set it to false until UpdateNegotiationNeeded() is called.
is_negotiation_needed_ = false;
update_negotiation_needed_on_empty_chain_ = true;
return false;
}
// We must not fire if the signaling state is no longer "stable". If
// negotiation is still needed when we return to "stable", a new negotiation
// needed event will be generated, so this one can safely be suppressed.
if (signaling_state_ != PeerConnectionInterface::kStable) {
return false;
}
// All checks have passed - please fire "negotiationneeded" now!
return true;
}
scoped_refptr<StreamCollectionInterface>
SdpOfferAnswerHandler::local_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
"Plan SdpSemantics. Please use GetSenders "
"instead.";
return local_streams_;
}
scoped_refptr<StreamCollectionInterface>
SdpOfferAnswerHandler::remote_streams() {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
"Plan SdpSemantics. Please use GetReceivers "
"instead.";
return remote_streams_;
}
bool SdpOfferAnswerHandler::AddStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
"SdpSemantics. Please use AddTrack instead.";
if (pc_->IsClosed()) {
return false;
}
if (!CanAddLocalMediaStream(local_streams_.get(), local_stream)) {
return false;
}
local_streams_->AddStream(scoped_refptr<MediaStreamInterface>(local_stream));
auto observer = std::make_unique<MediaStreamObserver>(
local_stream,
[this](AudioTrackInterface* audio_track,
MediaStreamInterface* media_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
OnAudioTrackAdded(audio_track, media_stream);
},
[this](AudioTrackInterface* audio_track,
MediaStreamInterface* media_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
OnAudioTrackRemoved(audio_track, media_stream);
},
[this](VideoTrackInterface* video_track,
MediaStreamInterface* media_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
OnVideoTrackAdded(video_track, media_stream);
},
[this](VideoTrackInterface* video_track,
MediaStreamInterface* media_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
OnVideoTrackRemoved(video_track, media_stream);
});
stream_observers_.push_back(std::move(observer));
for (const auto& track : local_stream->GetAudioTracks()) {
rtp_manager()->AddAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
rtp_manager()->AddVideoTrack(track.get(), local_stream);
}
pc_->legacy_stats()->AddStream(local_stream);
UpdateNegotiationNeeded();
return true;
}
void SdpOfferAnswerHandler::RemoveStream(MediaStreamInterface* local_stream) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
"Plan SdpSemantics. Please use RemoveTrack "
"instead.";
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
if (!pc_->IsClosed()) {
for (const auto& track : local_stream->GetAudioTracks()) {
rtp_manager()->RemoveAudioTrack(track.get(), local_stream);
}
for (const auto& track : local_stream->GetVideoTracks()) {
rtp_manager()->RemoveVideoTrack(track.get(), local_stream);
}
}
local_streams_->RemoveStream(local_stream);
std::erase_if(
stream_observers_,
[local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
return observer->stream()->id().compare(local_stream->id()) == 0;
});
if (pc_->IsClosed()) {
return;
}
UpdateNegotiationNeeded();
}
void SdpOfferAnswerHandler::OnAudioTrackAdded(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (pc_->IsClosed()) {
return;
}
rtp_manager()->AddAudioTrack(track, stream);
UpdateNegotiationNeeded();
}
void SdpOfferAnswerHandler::OnAudioTrackRemoved(AudioTrackInterface* track,
MediaStreamInterface* stream) {
if (pc_->IsClosed()) {
return;
}
rtp_manager()->RemoveAudioTrack(track, stream);
UpdateNegotiationNeeded();
}
void SdpOfferAnswerHandler::OnVideoTrackAdded(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (pc_->IsClosed()) {
return;
}
rtp_manager()->AddVideoTrack(track, stream);
UpdateNegotiationNeeded();
}
void SdpOfferAnswerHandler::OnVideoTrackRemoved(VideoTrackInterface* track,
MediaStreamInterface* stream) {
if (pc_->IsClosed()) {
return;
}
rtp_manager()->RemoveVideoTrack(track, stream);
UpdateNegotiationNeeded();
}
RTCError SdpOfferAnswerHandler::Rollback(SdpType desc_type) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::Rollback");
auto state = signaling_state();
if (state != PeerConnectionInterface::kHaveLocalOffer &&
state != PeerConnectionInterface::kHaveRemoteOffer) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_STATE,
(StringBuilder("Called in wrong signalingState: ")
<< (PeerConnectionInterface::AsString(signaling_state())))
.Release());
}
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
std::vector<scoped_refptr<RtpTransceiverInterface>>
now_receiving_transceivers;
std::vector<scoped_refptr<MediaStreamInterface>> all_added_streams;
std::vector<scoped_refptr<MediaStreamInterface>> all_removed_streams;
std::vector<scoped_refptr<RtpReceiverInterface>> removed_receivers;
for (auto&& transceivers_stable_state_pair : transceivers()->StableStates()) {
auto transceiver = transceivers_stable_state_pair.first;
auto stable_state = transceivers_stable_state_pair.second;
if (stable_state.did_set_fired_direction()) {
// If this rollback triggers going from not receiving to receving again,
// we need to fire "ontrack".
bool previously_fired_direction_is_recv =
transceiver->fired_direction().has_value() &&
RtpTransceiverDirectionHasRecv(*transceiver->fired_direction());
bool currently_fired_direction_is_recv =
stable_state.fired_direction().has_value() &&
RtpTransceiverDirectionHasRecv(
stable_state.fired_direction().value());
if (!previously_fired_direction_is_recv &&
currently_fired_direction_is_recv) {
now_receiving_transceivers.push_back(transceiver);
}
transceiver->internal()->set_fired_direction(
stable_state.fired_direction());
}
if (stable_state.remote_stream_ids()) {
std::vector<scoped_refptr<MediaStreamInterface>> added_streams;
std::vector<scoped_refptr<MediaStreamInterface>> removed_streams;
SetAssociatedRemoteStreams(transceiver->internal()->receiver_internal(),
stable_state.remote_stream_ids().value(),
&added_streams, &removed_streams);
all_added_streams.insert(all_added_streams.end(), added_streams.begin(),
added_streams.end());
all_removed_streams.insert(all_removed_streams.end(),
removed_streams.begin(),
removed_streams.end());
if (!stable_state.has_m_section() && !stable_state.newly_created()) {
continue;
}
}
// Due to the above `continue` statement, the below code only runs if there
// is a change in mid association (has_m_section), if the transceiver was
// newly created (newly_created) or if remote streams were not set.
RTC_DCHECK(transceiver->internal()->mid().has_value());
transceiver->internal()->ClearChannel();
if (signaling_state() == PeerConnectionInterface::kHaveRemoteOffer &&
transceiver->receiver()) {
removed_receivers.push_back(transceiver->receiver());
}
if (stable_state.newly_created()) {
if (transceiver->internal()->reused_for_addtrack()) {
transceiver->internal()->set_created_by_addtrack(true);
} else {
transceiver->internal()->StopTransceiverProcedure();
transceivers()->Remove(transceiver);
}
}
if (stable_state.init_send_encodings()) {
transceiver->internal()->sender_internal()->set_init_send_encodings(
stable_state.init_send_encodings().value());
}
transceiver->internal()->sender_internal()->set_transport(nullptr);
transceiver->internal()->receiver_internal()->set_transport(nullptr);
if (stable_state.has_m_section()) {
transceiver->internal()->set_mid(stable_state.mid());
transceiver->internal()->set_mline_index(stable_state.mline_index());
}
}
RTCError e = transport_controller_s()->RollbackTransports();
if (!e.ok()) {
return e;
}
transceivers()->DiscardStableStates();
pending_local_description_.reset();
pending_remote_description_.reset();
ChangeSignalingState(PeerConnectionInterface::kStable);
// Once all processing has finished, fire off callbacks.
pc_->RunWithObserver([&](auto observer) {
for (const auto& transceiver : now_receiving_transceivers) {
observer->OnTrack(transceiver);
observer->OnAddTrack(transceiver->receiver(),
transceiver->receiver()->streams());
}
for (const auto& receiver : removed_receivers) {
observer->OnRemoveTrack(receiver);
}
for (const auto& stream : all_added_streams) {
observer->OnAddStream(stream);
}
for (const auto& stream : all_removed_streams) {
observer->OnRemoveStream(stream);
}
});
// The assumption is that in case of implicit rollback
// UpdateNegotiationNeeded gets called in SetRemoteDescription.
if (desc_type == SdpType::kRollback) {
UpdateNegotiationNeeded();
if (is_negotiation_needed_) {
// Legacy version.
pc_->RunWithObserver(
[&](auto observer) { observer->OnRenegotiationNeeded(); });
// Spec-compliant version; the event may get invalidated before firing.
GenerateNegotiationNeededEvent();
}
}
return RTCError::OK();
}
bool SdpOfferAnswerHandler::IsUnifiedPlan() const {
return pc_->IsUnifiedPlan();
}
void SdpOfferAnswerHandler::OnOperationsChainEmpty() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (pc_->IsClosed() || !update_negotiation_needed_on_empty_chain_)
return;
update_negotiation_needed_on_empty_chain_ = false;
// Firing when chain is empty is only supported in Unified Plan to avoid
// Plan B regressions. (In Plan B, onnegotiationneeded is already broken
// anyway, so firing it even more might just be confusing.)
if (IsUnifiedPlan()) {
UpdateNegotiationNeeded();
}
}
std::optional<bool> SdpOfferAnswerHandler::is_caller() const {
RTC_DCHECK_RUN_ON(signaling_thread());
return is_caller_;
}
bool SdpOfferAnswerHandler::HasNewIceCredentials() {
RTC_DCHECK_RUN_ON(signaling_thread());
return local_ice_credentials_to_replace_->HasIceCredentials();
}
bool SdpOfferAnswerHandler::IceRestartPending(
const std::string& content_name) const {
RTC_DCHECK_RUN_ON(signaling_thread());
return pending_ice_restarts_.find(content_name) !=
pending_ice_restarts_.end();
}
bool SdpOfferAnswerHandler::NeedsIceRestart(
const std::string& content_name) const {
return pc_->NeedsIceRestart(content_name);
}
std::optional<SSLRole> SdpOfferAnswerHandler::GetDtlsRole(
const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
return transport_controller_s()->GetDtlsRole(mid);
}
void SdpOfferAnswerHandler::UpdateNegotiationNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!IsUnifiedPlan()) {
pc_->RunWithObserver(
[&](auto observer) { observer->OnRenegotiationNeeded(); });
GenerateNegotiationNeededEvent();
return;
}
// In the spec, a task is queued here to run the following steps - this is
// meant to ensure we do not fire onnegotiationneeded prematurely if
// multiple changes are being made at once. In order to support Chromium's
// implementation where the JavaScript representation of the PeerConnection
// lives on a separate thread though, the queuing of a task is instead
// performed by the PeerConnectionObserver posting from the signaling thread
// to the JavaScript main thread that negotiation is needed. And because the
// Operations Chain lives on the WebRTC signaling thread,
// ShouldFireNegotiationNeededEvent() must be called before firing the event
// to ensure the Operations Chain is still empty and the event has not been
// invalidated.
// If connection's [[IsClosed]] slot is true, abort these steps.
if (pc_->IsClosed())
return;
// If connection's signaling state is not "stable", abort these steps.
if (signaling_state() != PeerConnectionInterface::kStable)
return;
// NOTE
// The negotiation-needed flag will be updated once the state transitions to
// "stable", as part of the steps for setting an RTCSessionDescription.
// If the result of checking if negotiation is needed is false, clear the
// negotiation-needed flag by setting connection's [[NegotiationNeeded]]
// slot to false, and abort these steps.
bool is_negotiation_needed = CheckIfNegotiationIsNeeded();
if (!is_negotiation_needed) {
is_negotiation_needed_ = false;
// Invalidate any negotiation needed event that may previosuly have been
// generated.
++negotiation_needed_event_id_;
return;
}
// If connection's [[NegotiationNeeded]] slot is already true, abort these
// steps.
if (is_negotiation_needed_)
return;
// Set connection's [[NegotiationNeeded]] slot to true.
is_negotiation_needed_ = true;
// Queue a task that runs the following steps:
// If connection's [[IsClosed]] slot is true, abort these steps.
// If connection's [[NegotiationNeeded]] slot is false, abort these steps.
// Fire an event named negotiationneeded at connection.
pc_->RunWithObserver(
[&](auto observer) { observer->OnRenegotiationNeeded(); });
// Fire the spec-compliant version; when ShouldFireNegotiationNeededEvent()
// is used in the task queued by the observer, this event will only fire
// when the chain is empty.
GenerateNegotiationNeededEvent();
}
void SdpOfferAnswerHandler::AllocateSctpSids() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!local_description() || !remote_description()) {
RTC_DLOG(LS_VERBOSE)
<< "Local and Remote descriptions must be applied to get the "
"SSL Role of the SCTP transport.";
return;
}
std::optional<SSLRole> guessed_role = GuessSslRole();
network_thread()->BlockingCall(
[&, data_channel_controller = data_channel_controller()] {
RTC_DCHECK_RUN_ON(network_thread());
std::optional<SSLRole> role = pc_->GetSctpSslRole_n();
if (!role)
role = guessed_role;
if (role)
data_channel_controller->AllocateSctpSids(*role);
});
}
std::optional<SSLRole> SdpOfferAnswerHandler::GuessSslRole() const {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!pc_->sctp_mid())
return std::nullopt;
// TODO(bugs.webrtc.org/13668): This guesswork is guessing wrong (returning
// SSL_CLIENT = ACTIVE) if remote offer has role ACTIVE, but we'll be able
// to detect that by looking at the SDP.
//
// The phases of establishing an SCTP session are:
//
// Offerer:
//
// * Before negotiation: Neither is_caller nor sctp_mid exists.
// * After setting an offer as local description: is_caller is known (true),
// sctp_mid is known, but we don't know the SSL role for sure (or if we'll
// eventually get an SCTP session).
// * After setting an answer as the remote description: We know is_caller,
// sctp_mid and that we'll get the SCTP channel established (m-section
// wasn't rejected).
// * Special case: The SCTP m-section was rejected: Close datachannels.
// * We MAY know the SSL role if we offered actpass and got back active or
// passive; if the other end is a webrtc implementation, it will be active.
// * After the TLS handshake: We have a definitive answer on the SSL role.
//
// Answerer:
//
// * After setting an offer as remote description: We know is_caller (false).
// * If there was an SCTP session, we know the SCTP mid. We also know the
// SSL role, since if the remote offer was actpass or passive, we'll answer
// active, and if the remote offer was active, we're passive.
// * Special case: No SCTP m= line. We don't know for sure if the remote
// doesn't support it or just didn't offer it. Not sure what we do in this
// case (logic would suggest fire a `negotiationneeded` event and generate a
// subsequent offer, but this needs to be tested).
// * After the TLS handshake: We know that TLS obeyed the protocol. There
// should be an error surfaced somewhere if it didn't.
// * "Guessing" should always be correct if we get an SCTP session and are not
// the offerer.
return is_caller() ? SSL_SERVER : SSL_CLIENT;
}
bool SdpOfferAnswerHandler::CheckIfNegotiationIsNeeded() {
RTC_DCHECK_RUN_ON(signaling_thread());
// 1. If any implementation-specific negotiation is required, as described
// at the start of this section, return true.
// 2. If connection.[[LocalIceCredentialsToReplace]] is not empty, return
// true.
if (local_ice_credentials_to_replace_->HasIceCredentials()) {
return true;
}
// 3. Let description be connection.[[CurrentLocalDescription]].
const SessionDescriptionInterface* description = current_local_description();
if (!description)
return true;
// 4. If connection has created any RTCDataChannels, and no m= section in
// description has been negotiated yet for data, return true.
if (data_channel_controller()->HasUsedDataChannels()) {
const ContentInfo* data_content =
GetFirstDataContent(description->description()->contents());
if (!data_content) {
return true;
}
// The remote end might have rejected the data content.
const ContentInfo* remote_data_content =
current_remote_description()
? current_remote_description()->description()->GetContentByName(
data_content->mid())
: nullptr;
if (remote_data_content && remote_data_content->rejected) {
return true;
}
}
if (!ConfiguredForMedia()) {
return false;
}
// 5. For each transceiver in connection's set of transceivers, perform the
// following checks:
for (const auto& transceiver : transceivers()->ListInternal()) {
const ContentInfo* current_local_msection =
FindTransceiverMSection(transceiver, description);
const ContentInfo* current_remote_msection =
FindTransceiverMSection(transceiver, current_remote_description());
// 5.4 If transceiver is stopped and is associated with an m= section,
// but the associated m= section is not yet rejected in
// connection.[[CurrentLocalDescription]] or
// connection.[[CurrentRemoteDescription]], return true.
if (transceiver->stopped()) {
RTC_DCHECK(transceiver->stopping());
if (current_local_msection && !current_local_msection->rejected &&
((current_remote_msection && !current_remote_msection->rejected) ||
!current_remote_msection)) {
return true;
}
continue;
}
// 5.1 If transceiver.[[Stopping]] is true and transceiver.[[Stopped]] is
// false, return true.
if (transceiver->stopping() && !transceiver->stopped())
return true;
// 5.2 If transceiver isn't stopped and isn't yet associated with an m=
// section in description, return true.
if (!current_local_msection)
return true;
const MediaContentDescription* current_local_media_description =
current_local_msection->media_description();
// 5.3 If transceiver isn't stopped and is associated with an m= section
// in description then perform the following checks:
// 5.3.1 If transceiver.[[Direction]] is "sendrecv" or "sendonly", and the
// associated m= section in description either doesn't contain a single
// "a=msid" line, or the number of MSIDs from the "a=msid" lines in this
// m= section, or the MSID values themselves, differ from what is in
// transceiver.sender.[[AssociatedMediaStreamIds]], return true.
if (RtpTransceiverDirectionHasSend(transceiver->direction())) {
if (current_local_media_description->streams().empty())
return true;
std::vector<std::string> msection_msids;
for (const auto& stream : current_local_media_description->streams()) {
for (const std::string& msid : stream.stream_ids())
msection_msids.push_back(msid);
}
std::vector<std::string> transceiver_msids =
transceiver->sender()->stream_ids();
if (msection_msids.size() != transceiver_msids.size())
return true;
absl::c_sort(transceiver_msids);
absl::c_sort(msection_msids);
if (transceiver_msids != msection_msids)
return true;
}
// 5.3.2 If description is of type "offer", and the direction of the
// associated m= section in neither connection.[[CurrentLocalDescription]]
// nor connection.[[CurrentRemoteDescription]] matches
// transceiver.[[Direction]], return true.
if (description->GetType() == SdpType::kOffer) {
if (!current_remote_description())
return true;
if (!current_remote_msection)
return true;
RtpTransceiverDirection current_local_direction =
current_local_media_description->direction();
RtpTransceiverDirection current_remote_direction =
current_remote_msection->media_description()->direction();
if (transceiver->direction() != current_local_direction &&
transceiver->direction() !=
RtpTransceiverDirectionReversed(current_remote_direction)) {
return true;
}
}
// 5.3.3 If description is of type "answer", and the direction of the
// associated m= section in the description does not match
// transceiver.[[Direction]] intersected with the offered direction (as
// described in [JSEP] (section 5.3.1.)), return true.
if (description->GetType() == SdpType::kAnswer) {
if (!remote_description())
return true;
const ContentInfo* offered_remote_msection =
FindTransceiverMSection(transceiver, remote_description());
RtpTransceiverDirection offered_direction =
offered_remote_msection
? offered_remote_msection->media_description()->direction()
: RtpTransceiverDirection::kInactive;
if (current_local_media_description->direction() !=
(RtpTransceiverDirectionIntersection(
transceiver->direction(),
RtpTransceiverDirectionReversed(offered_direction)))) {
return true;
}
}
}
// If all the preceding checks were performed and true was not returned,
// nothing remains to be negotiated; return false.
return false;
}
void SdpOfferAnswerHandler::GenerateNegotiationNeededEvent() {
RTC_DCHECK_RUN_ON(signaling_thread());
++negotiation_needed_event_id_;
pc_->RunWithObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
observer->OnNegotiationNeededEvent(negotiation_needed_event_id_);
});
}
RTCError SdpOfferAnswerHandler::ValidateSessionDescription(
const SessionDescriptionInterface* sdesc,
ContentSource source,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
// An assumption is that a check for session error is done at a higher level.
RTC_DCHECK_EQ(SessionError::kNone, session_error());
if (!sdesc || !sdesc->description()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
}
SdpType type = sdesc->GetType();
if ((source == CS_LOCAL && !ExpectSetLocalDescription(type)) ||
(source == CS_REMOTE && !ExpectSetRemoteDescription(type))) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_STATE,
(StringBuilder("Called in wrong state: ")
<< PeerConnectionInterface::AsString(signaling_state()))
.Release());
}
RTCError error = ValidateMids(*sdesc->description());
if (!error.ok()) {
return error;
}
// Verify crypto settings.
if (pc_->dtls_enabled()) {
RTCError crypto_error = VerifyCrypto(
sdesc->description(), pc_->dtls_enabled(), bundle_groups_by_mid);
if (!crypto_error.ok()) {
return crypto_error;
}
}
// Verify ice-ufrag and ice-pwd.
if (!VerifyIceUfragPwdPresent(sdesc->description(), bundle_groups_by_mid)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kSdpWithoutIceUfragPwd);
}
// Validate that there are no collisions of bundled payload types.
error = ValidateBundledPayloadTypes(*sdesc->description());
// TODO(bugs.webrtc.org/14420): actually reject.
RTC_HISTOGRAM_BOOLEAN("WebRTC.PeerConnection.ValidBundledPayloadTypes",
error.ok());
// Validate that there are no collisions of bundled header extensions ids.
error = ValidateBundledRtpHeaderExtensions(*sdesc->description());
if (!error.ok()) {
return error;
}
// Validate the SSRC groups.
error = ValidateSsrcGroups(*sdesc->description());
if (!error.ok()) {
return error;
}
if (!pc_->ValidateBundleSettings(sdesc->description(),
bundle_groups_by_mid)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kBundleWithoutRtcpMux);
}
error = ValidatePayloadTypes(*sdesc->description());
if (!error.ok()) {
return error;
}
// TODO(skvlad): When the local rtcp-mux policy is Require, reject any
// m-lines that do not rtcp-mux enabled.
// Verify m-lines in Answer when compared against Offer.
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
// With an answer we want to compare the new answer session description
// with the offer's session description from the current negotiation.
const SessionDescription* offer_desc =
(source == CS_LOCAL) ? remote_description()->description()
: local_description()->description();
if (!MediaSectionsHaveSameCount(*offer_desc, *sdesc->description()) ||
!MediaSectionsInSameOrder(*offer_desc, nullptr, *sdesc->description(),
type)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kMlineMismatchInAnswer);
}
} else {
// The re-offers should respect the order of m= sections in current
// description. See RFC3264 Section 8 paragraph 4 for more details.
// With a re-offer, either the current local or current remote
// descriptions could be the most up to date, so we would like to check
// against both of them if they exist. It could be the case that one of
// them has a 0 port for a media section, but the other does not. This is
// important to check against in the case that we are recycling an m=
// section.
const SessionDescription* current_desc = nullptr;
const SessionDescription* secondary_current_desc = nullptr;
if (local_description()) {
current_desc = local_description()->description();
if (remote_description()) {
secondary_current_desc = remote_description()->description();
}
} else if (remote_description()) {
current_desc = remote_description()->description();
}
if (current_desc &&
!MediaSectionsInSameOrder(*current_desc, secondary_current_desc,
*sdesc->description(), type)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
kMlineMismatchInSubsequentOffer);
}
}
if (IsUnifiedPlan()) {
// Ensure that each audio and video media section has at most one
// "StreamParams". This will return an error if receiving a session
// description from a "Plan B" endpoint which adds multiple tracks of the
// same type. With Unified Plan, there can only be at most one track per
// media section.
for (const ContentInfo& content : sdesc->description()->contents()) {
const MediaContentDescription& desc = *content.media_description();
if ((desc.type() == MediaType::AUDIO ||
desc.type() == MediaType::VIDEO) &&
desc.streams().size() > 1u) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Media section has more than one track specified with a=ssrc lines "
"which is not supported with Unified Plan.");
}
}
// Validate spec-simulcast which only works if the remote end negotiated the
// mid and rid header extension.
error = ValidateRtpHeaderExtensionsForSpecSimulcast(*sdesc->description());
if (!error.ok()) {
return error;
}
}
return RTCError::OK();
}
RTCError SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels(
ContentSource source,
const SessionDescriptionInterface& new_session,
const SessionDescriptionInterface* old_local_description,
const SessionDescriptionInterface* old_remote_description,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
TRACE_EVENT0("webrtc",
"SdpOfferAnswerHandler::UpdateTransceiversAndDataChannels");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
if (new_session.GetType() == SdpType::kOffer) {
// If the BUNDLE policy is max-bundle, then we know for sure that all
// transports will be bundled from the start. Return an error if
// max-bundle is specified but the session description does not have a
// BUNDLE group.
if (pc_->configuration()->bundle_policy ==
PeerConnectionInterface::kBundlePolicyMaxBundle &&
bundle_groups_by_mid.empty()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"max-bundle configured but session description has no BUNDLE group");
}
}
const ContentInfos& new_contents = new_session.description()->contents();
for (size_t i = 0; i < new_contents.size(); ++i) {
const ContentInfo& new_content = new_contents[i];
MediaType media_type = new_content.media_description()->type();
mid_generator_.AddKnownId(new_content.mid());
auto it = bundle_groups_by_mid.find(new_content.mid());
const ContentGroup* bundle_group =
it != bundle_groups_by_mid.end() ? it->second : nullptr;
if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) {
const ContentInfo* old_local_content = nullptr;
if (old_local_description &&
i < old_local_description->description()->contents().size()) {
old_local_content =
&old_local_description->description()->contents()[i];
}
const ContentInfo* old_remote_content = nullptr;
if (old_remote_description &&
i < old_remote_description->description()->contents().size()) {
old_remote_content =
&old_remote_description->description()->contents()[i];
}
auto transceiver_or_error =
AssociateTransceiver(source, new_session.GetType(), i, new_content,
old_local_content, old_remote_content);
if (!transceiver_or_error.ok()) {
// In the case where a transceiver is rejected locally prior to being
// associated, we don't expect to find a transceiver, but might find it
// in the case where state is still "stopping", not "stopped".
if (new_content.rejected) {
continue;
}
return transceiver_or_error.MoveError();
}
auto transceiver = transceiver_or_error.MoveValue();
RTCError error =
UpdateTransceiverChannel(transceiver, new_content, bundle_group);
// Handle locally rejected content. This code path is only needed for apps
// that SDP munge. Remote rejected content is handled in
// ApplyRemoteDescriptionUpdateTransceiverState().
if (source == ContentSource::CS_LOCAL && new_content.rejected) {
// Local offer.
if (new_session.GetType() == SdpType::kOffer) {
// If the RtpTransceiver API was used, it would already have made the
// transceiver stopping. But if the rejection was caused by SDP
// munging then we need to ensure the transceiver is stopping here.
if (!transceiver->internal()->stopping()) {
transceiver->internal()->StopStandard();
}
RTC_DCHECK(transceiver->internal()->stopping());
} else {
// Local answer.
RTC_DCHECK(new_session.GetType() == SdpType::kAnswer ||
new_session.GetType() == SdpType::kPrAnswer);
// When RtpTransceiver API is used, rejection happens in the offer and
// the transceiver will already be stopped at local answer time
// (calling stop between SRD(offer) and SLD(answer) would not reject
// the content in the answer - instead this would trigger a follow-up
// O/A exchange). So if the content was rejected but the transceiver
// is not already stopped, SDP munging has happened and we need to
// ensure the transceiver is stopped.
if (!transceiver->internal()->stopped()) {
transceiver->internal()->StopTransceiverProcedure();
}
RTC_DCHECK(transceiver->internal()->stopped());
}
}
if (!error.ok()) {
return error;
}
} else if (media_type == MediaType::DATA) {
const auto data_mid = pc_->sctp_mid();
if (data_mid && new_content.mid() != data_mid.value()) {
// Ignore all but the first data section.
RTC_LOG(LS_INFO) << "Ignoring data media section with MID="
<< new_content.mid();
continue;
}
RTCError error =
UpdateDataChannelTransport(source, new_content, bundle_group);
if (!error.ok()) {
return error;
}
} else if (media_type == MediaType::UNSUPPORTED) {
RTC_LOG(LS_INFO) << "Ignoring unsupported media type";
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Unknown section type.");
}
}
return RTCError::OK();
}
RTCErrorOr<scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
SdpOfferAnswerHandler::AssociateTransceiver(
ContentSource source,
SdpType type,
size_t mline_index,
const ContentInfo& content,
const ContentInfo* old_local_content,
const ContentInfo* old_remote_content) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::AssociateTransceiver");
RTC_DCHECK(IsUnifiedPlan());
#if RTC_DCHECK_IS_ON
// If this is an offer then the m= section might be recycled. If the m=
// section is being recycled (defined as: rejected in the current local or
// remote description and not rejected in new description), the transceiver
// should have been removed by RemoveStoppedtransceivers()->
if (IsMediaSectionBeingRecycled(type, content, old_local_content,
old_remote_content)) {
const auto old_mid = (old_local_content && old_local_content->rejected)
? old_local_content->mid()
: old_remote_content->mid();
auto old_transceiver = transceivers()->FindByMid(old_mid);
// The transceiver should be disassociated in RemoveStoppedTransceivers()
RTC_DCHECK(!old_transceiver);
}
#endif
const MediaContentDescription* media_desc = content.media_description();
auto transceiver = transceivers()->FindByMid(content.mid());
if (source == CS_LOCAL) {
// Find the RtpTransceiver that corresponds to this m= section, using the
// mapping between transceivers and m= section indices established when
// creating the offer.
if (!transceiver) {
transceiver = transceivers()->FindByMLineIndex(mline_index);
}
if (!transceiver) {
// This may happen normally when media sections are rejected.
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Transceiver not found based on m-line index");
}
} else {
RTC_DCHECK_EQ(source, CS_REMOTE);
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers
// of the same type...
// When simulcast is requested, a transceiver cannot be associated because
// AddTrack cannot be called to initialize it.
if (!transceiver &&
RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
!media_desc->HasSimulcast()) {
transceiver = FindAvailableTransceiverToReceive(media_desc->type());
}
// If no RtpTransceiver was found in the previous step, create one with a
// recvonly direction.
if (!transceiver) {
RTC_LOG(LS_INFO) << "Adding " << MediaTypeToString(media_desc->type())
<< " transceiver for MID=" << content.mid()
<< " at i=" << mline_index
<< " in response to the remote description.";
std::string sender_id = CreateRandomUuid();
std::vector<RtpEncodingParameters> send_encodings =
GetSendEncodingsFromRemoteDescription(*media_desc);
auto sender = rtp_manager()->CreateSender(media_desc->type(), sender_id,
nullptr, {}, send_encodings);
std::string receiver_id;
if (!media_desc->streams().empty()) {
receiver_id = media_desc->streams()[0].id;
} else {
receiver_id = CreateRandomUuid();
}
auto receiver =
rtp_manager()->CreateReceiver(media_desc->type(), receiver_id);
transceiver = rtp_manager()->CreateAndAddTransceiver(sender, receiver);
transceiver->internal()->set_direction(
RtpTransceiverDirection::kRecvOnly);
if (type == SdpType::kOffer) {
transceivers()->StableState(transceiver)->set_newly_created();
}
}
RTC_DCHECK(transceiver);
// Check if the offer indicated simulcast but the answer rejected it.
// This can happen when simulcast is not supported on the remote party.
if (SimulcastIsRejected(old_local_content, *media_desc,
pc_->GetCryptoOptions()
.srtp.enable_encrypted_rtp_header_extensions)) {
RTCError error =
DisableSimulcastInSender(transceiver->internal()->sender_internal());
if (!error.ok()) {
RTC_LOG(LS_ERROR) << "Failed to remove rejected simulcast.";
return std::move(error);
}
}
}
if (transceiver->media_type() != media_desc->type()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Transceiver type does not match media description type.");
}
if (media_desc->HasSimulcast()) {
std::vector<SimulcastLayer> layers =
source == CS_LOCAL
? media_desc->simulcast_description().send_layers().GetAllLayers()
: media_desc->simulcast_description()
.receive_layers()
.GetAllLayers();
RTCError error = UpdateSimulcastLayerStatusInSender(
layers, transceiver->internal()->sender_internal());
if (!error.ok()) {
RTC_LOG(LS_ERROR) << "Failed updating status for simulcast layers.";
return std::move(error);
}
}
if (type == SdpType::kOffer) {
bool state_changes = transceiver->internal()->mid() != content.mid() ||
transceiver->internal()->mline_index() != mline_index;
if (state_changes) {
transceivers()
->StableState(transceiver)
->SetMSectionIfUnset(transceiver->internal()->mid(),
transceiver->internal()->mline_index());
}
}
// Associate the found or created RtpTransceiver with the m= section by
// setting the value of the RtpTransceiver's mid property to the MID of the m=
// section, and establish a mapping between the transceiver and the index of
// the m= section.
transceiver->internal()->set_mid(content.mid());
transceiver->internal()->set_mline_index(mline_index);
return std::move(transceiver);
}
RTCError SdpOfferAnswerHandler::UpdateTransceiverChannel(
scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> transceiver,
const ContentInfo& content,
const ContentGroup* bundle_group) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateTransceiverChannel");
RTC_DCHECK(IsUnifiedPlan());
RTC_DCHECK(transceiver);
ChannelInterface* channel = transceiver->internal()->channel();
if (content.rejected) {
if (channel) {
transceiver->internal()->ClearChannel();
}
} else {
if (!channel) {
auto error = transceiver->internal()->CreateChannel(
content.mid(), pc_->call_ptr(), pc_->configuration()->media_config,
pc_->SrtpRequired(), pc_->GetCryptoOptions(), audio_options(),
video_options(), video_bitrate_allocator_factory_.get(),
[&](absl::string_view mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
if (!error.ok()) {
return error;
}
}
}
return RTCError::OK();
}
RTCError SdpOfferAnswerHandler::UpdateDataChannelTransport(
ContentSource source,
const ContentInfo& content,
const ContentGroup* bundle_group) {
if (content.rejected) {
RTC_LOG(LS_INFO) << "Rejected data channel transport with mid="
<< content.mid();
StringBuilder sb;
sb << "Rejected data channel transport with mid=" << content.mid();
RTCError error(RTCErrorType::OPERATION_ERROR_WITH_DATA, sb.Release());
error.set_error_detail(RTCErrorDetailType::DATA_CHANNEL_FAILURE);
pc_->DestroyDataChannelTransport(error);
} else if (!pc_->CreateDataChannelTransport(content.mid())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create data channel.");
}
return RTCError::OK();
}
bool SdpOfferAnswerHandler::ExpectSetLocalDescription(SdpType type) {
PeerConnectionInterface::SignalingState state = signaling_state();
if (type == SdpType::kOffer) {
return (state == PeerConnectionInterface::kStable) ||
(state == PeerConnectionInterface::kHaveLocalOffer);
} else {
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
return (state == PeerConnectionInterface::kHaveRemoteOffer) ||
(state == PeerConnectionInterface::kHaveLocalPrAnswer);
}
}
bool SdpOfferAnswerHandler::ExpectSetRemoteDescription(SdpType type) {
PeerConnectionInterface::SignalingState state = signaling_state();
if (type == SdpType::kOffer) {
return (state == PeerConnectionInterface::kStable) ||
(state == PeerConnectionInterface::kHaveRemoteOffer);
} else {
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
return (state == PeerConnectionInterface::kHaveLocalOffer) ||
(state == PeerConnectionInterface::kHaveRemotePrAnswer);
}
}
void SdpOfferAnswerHandler::FillInMissingRemoteMids(
SessionDescription* new_remote_description) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(new_remote_description);
const ContentInfos no_infos;
const ContentInfos& local_contents =
(local_description() ? local_description()->description()->contents()
: no_infos);
const ContentInfos& remote_contents =
(remote_description() ? remote_description()->description()->contents()
: no_infos);
for (size_t i = 0; i < new_remote_description->contents().size(); ++i) {
ContentInfo& content = new_remote_description->contents()[i];
if (!content.mid().empty()) {
continue;
}
std::string new_mid;
absl::string_view source_explanation;
if (IsUnifiedPlan()) {
if (i < local_contents.size()) {
new_mid = local_contents[i].mid();
source_explanation = "from the matching local media section";
} else if (i < remote_contents.size()) {
new_mid = remote_contents[i].mid();
source_explanation = "from the matching previous remote media section";
} else {
new_mid = mid_generator_.GenerateString();
source_explanation = "generated just now";
}
} else {
new_mid = GetDefaultMidForPlanB(content.media_description()->type());
source_explanation = "to match pre-existing behavior";
}
RTC_DCHECK(!new_mid.empty());
content.set_mid(new_mid);
new_remote_description->transport_infos()[i].content_name = new_mid;
RTC_LOG(LS_INFO) << "SetRemoteDescription: Remote media section at i=" << i
<< " is missing an a=mid line. Filling in the value '"
<< new_mid << "' " << source_explanation << ".";
}
}
scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
SdpOfferAnswerHandler::FindAvailableTransceiverToReceive(
MediaType media_type) const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
// From JSEP section 5.10 (Applying a Remote Description):
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers of
// the same type that were added to the PeerConnection by addTrack and are not
// associated with any m= section and are not stopped, find the first such
// RtpTransceiver.
for (auto transceiver : transceivers()->List()) {
if (transceiver->media_type() == media_type &&
transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
!transceiver->stopped()) {
return transceiver;
}
}
return nullptr;
}
const ContentInfo* SdpOfferAnswerHandler::FindMediaSectionForTransceiver(
const RtpTransceiver* transceiver,
const SessionDescriptionInterface* sdesc) const {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(transceiver);
RTC_DCHECK(sdesc);
if (IsUnifiedPlan()) {
if (!transceiver->mid()) {
// This transceiver is not associated with a media section yet.
return nullptr;
}
return sdesc->description()->GetContentByName(*transceiver->mid());
} else {
// Plan B only allows at most one audio and one video section, so use the
// first media section of that type.
return GetFirstMediaContent(sdesc->description()->contents(),
transceiver->media_type());
}
}
void SdpOfferAnswerHandler::GetOptionsForOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
MediaSessionOptions* session_options) {
RTC_DCHECK_RUN_ON(signaling_thread());
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
if (IsUnifiedPlan()) {
GetOptionsForUnifiedPlanOffer(offer_answer_options, session_options);
} else {
GetOptionsForPlanBOffer(offer_answer_options, session_options);
}
// Apply ICE restart flag and renomination flag.
bool ice_restart = offer_answer_options.ice_restart || HasNewIceCredentials();
for (auto& options : session_options->media_description_options) {
options.transport_options.ice_restart = ice_restart;
options.transport_options.enable_ice_renomination =
pc_->configuration()->enable_ice_renomination;
}
session_options->rtcp_cname = rtcp_cname_;
session_options->crypto_options = pc_->GetCryptoOptions();
session_options->pooled_ice_credentials =
context_->network_thread()->BlockingCall(
[this] { return port_allocator()->GetPooledIceCredentials(); });
session_options->offer_extmap_allow_mixed =
pc_->configuration()->offer_extmap_allow_mixed;
// Allow fallback for using obsolete SCTP syntax.
// Note that the default in `session_options` is true, while
// the default in `options` is false.
session_options->use_obsolete_sctp_sdp =
offer_answer_options.use_obsolete_sctp_sdp;
}
void SdpOfferAnswerHandler::GetOptionsForPlanBOffer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
MediaSessionOptions* session_options) {
bool offer_new_data_description =
data_channel_controller()->HasUsedDataChannels();
bool send_audio = false;
bool send_video = false;
bool recv_audio = false;
bool recv_video = false;
if (ConfiguredForMedia()) {
// Figure out transceiver directional preferences.
send_audio =
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
send_video =
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
// By default, generate sendrecv/recvonly m= sections.
recv_audio = true;
recv_video = true;
}
// By default, only offer a new m= section if we have media to send with it.
bool offer_new_audio_description = send_audio;
bool offer_new_video_description = send_video;
if (ConfiguredForMedia()) {
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
offer_new_audio_description =
offer_new_audio_description ||
(offer_answer_options.offer_to_receive_audio > 0);
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
offer_new_video_description =
offer_new_video_description ||
(offer_answer_options.offer_to_receive_video > 0);
}
}
std::optional<size_t> audio_index;
std::optional<size_t> video_index;
std::optional<size_t> data_index;
// If a current description exists, generate m= sections in the same order,
// using the first audio/video/data section that appears and rejecting
// extraneous ones.
if (local_description()) {
GenerateMediaDescriptionOptions(
local_description(),
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
&audio_index, &video_index, &data_index, session_options);
}
if (ConfiguredForMedia()) {
// Add audio/video/data m= sections to the end if needed.
if (!audio_index && offer_new_audio_description) {
MediaDescriptionOptions options(
MediaType::AUDIO, GetDefaultMidForPlanB(MediaType::AUDIO),
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false);
options.header_extensions =
media_engine()->voice().GetRtpHeaderExtensions(&env_.field_trials());
session_options->media_description_options.push_back(options);
audio_index = session_options->media_description_options.size() - 1;
}
if (!video_index && offer_new_video_description) {
MediaDescriptionOptions options(
MediaType::VIDEO, GetDefaultMidForPlanB(MediaType::VIDEO),
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false);
options.header_extensions =
media_engine()->video().GetRtpHeaderExtensions(&env_.field_trials());
session_options->media_description_options.push_back(options);
video_index = session_options->media_description_options.size() - 1;
}
MediaDescriptionOptions* audio_media_description_options =
!audio_index
? nullptr
: &session_options->media_description_options[*audio_index];
MediaDescriptionOptions* video_media_description_options =
!video_index
? nullptr
: &session_options->media_description_options[*video_index];
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
}
if (!data_index && offer_new_data_description) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(
GetDefaultMidForPlanB(MediaType::DATA)));
}
}
void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanOffer(
const RTCOfferAnswerOptions& offer_answer_options,
MediaSessionOptions* session_options) {
// Rules for generating an offer are dictated by JSEP sections 5.2.1 (Initial
// Offers) and 5.2.2 (Subsequent Offers).
RTC_DCHECK_EQ(session_options->media_description_options.size(), 0);
const ContentInfos no_infos;
const ContentInfos& local_contents =
(local_description() ? local_description()->description()->contents()
: no_infos);
const ContentInfos& remote_contents =
(remote_description() ? remote_description()->description()->contents()
: no_infos);
// The mline indices that can be recycled. New transceivers should reuse these
// slots first.
std::queue<size_t> recycleable_mline_indices;
// First, go through each media section that exists in either the local or
// remote description and generate a media section in this offer for the
// associated transceiver. If a media section can be recycled, generate a
// default, rejected media section here that can be later overwritten.
for (size_t i = 0;
i < std::max(local_contents.size(), remote_contents.size()); ++i) {
// Either `local_content` or `remote_content` is non-null.
const ContentInfo* local_content =
(i < local_contents.size() ? &local_contents[i] : nullptr);
const ContentInfo* current_local_content =
GetContentByIndex(current_local_description(), i);
const ContentInfo* remote_content =
(i < remote_contents.size() ? &remote_contents[i] : nullptr);
const ContentInfo* current_remote_content =
GetContentByIndex(current_remote_description(), i);
bool had_been_rejected =
(current_local_content && current_local_content->rejected) ||
(current_remote_content && current_remote_content->rejected);
const std::string& mid =
(local_content ? local_content->mid() : remote_content->mid());
MediaType media_type =
(local_content ? local_content->media_description()->type()
: remote_content->media_description()->type());
if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) {
// A media section is considered eligible for recycling if it is marked as
// rejected in either the current local or current remote description.
auto transceiver = transceivers()->FindByMid(mid);
if (!transceiver) {
// No associated transceiver. The media section has been stopped.
recycleable_mline_indices.push(i);
session_options->media_description_options.push_back(
MediaDescriptionOptions(media_type, mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
// NOTE: a stopping transceiver should be treated as a stopped one in
// createOffer as specified in
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer.
if (had_been_rejected && transceiver->stopping()) {
session_options->media_description_options.push_back(
MediaDescriptionOptions(transceiver->media_type(), mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
recycleable_mline_indices.push(i);
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(
transceiver->internal(), mid,
/*is_create_offer=*/true));
// CreateOffer shouldn't really cause any state changes in
// PeerConnection, but we need a way to match new transceivers to new
// media sections in SetLocalDescription and JSEP specifies this is
// done by recording the index of the media section generated for the
// transceiver in the offer.
transceiver->internal()->set_mline_index(i);
}
}
} else if (media_type == MediaType::UNSUPPORTED) {
RTC_DCHECK(local_content->rejected);
session_options->media_description_options.push_back(
MediaDescriptionOptions(media_type, mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
RTC_CHECK_EQ(MediaType::DATA, media_type);
if (had_been_rejected) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(mid));
} else {
const auto data_mid = pc_->sctp_mid();
if (data_mid.has_value() && mid == data_mid.value()) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(mid));
} else {
if (!data_mid.has_value()) {
RTC_LOG(LS_ERROR) << "Datachannel transport not available: " << mid;
}
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(mid));
}
}
}
}
// Next, look for transceivers that are newly added (that is, are not stopped
// and not associated). Reuse media sections marked as recyclable first,
// otherwise append to the end of the offer. New media sections should be
// added in the order they were added to the PeerConnection.
if (ConfiguredForMedia()) {
for (const auto& transceiver : transceivers()->ListInternal()) {
if (transceiver->mid() || transceiver->stopping()) {
continue;
}
size_t mline_index;
if (!recycleable_mline_indices.empty()) {
mline_index = recycleable_mline_indices.front();
recycleable_mline_indices.pop();
session_options->media_description_options[mline_index] =
GetMediaDescriptionOptionsForTransceiver(
transceiver, mid_generator_.GenerateString(),
/*is_create_offer=*/true);
} else {
mline_index = session_options->media_description_options.size();
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(
transceiver, mid_generator_.GenerateString(),
/*is_create_offer=*/true));
}
// See comment above for why CreateOffer changes the transceiver's state.
transceiver->set_mline_index(mline_index);
}
}
// Lastly, add a m-section if we have requested local data channels and an
// m section does not already exist.
if (!pc_->sctp_mid() && data_channel_controller()->HasDataChannels()) {
// Attempt to recycle a stopped m-line.
// TODO(crbug.com/1442604): sctp_mid() should return the mid if one was
// ever created but rejected.
bool recycled = false;
for (size_t i = 0; i < session_options->media_description_options.size();
i++) {
auto media_description = session_options->media_description_options[i];
if (media_description.type == MediaType::DATA &&
media_description.stopped) {
session_options->media_description_options[i] =
GetMediaDescriptionOptionsForActiveData(media_description.mid);
recycled = true;
break;
}
}
if (!recycled) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(
mid_generator_.GenerateString()));
}
}
}
void SdpOfferAnswerHandler::GetOptionsForAnswer(
const RTCOfferAnswerOptions& offer_answer_options,
MediaSessionOptions* session_options) {
RTC_DCHECK_RUN_ON(signaling_thread());
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
if (IsUnifiedPlan()) {
GetOptionsForUnifiedPlanAnswer(offer_answer_options, session_options);
} else {
GetOptionsForPlanBAnswer(offer_answer_options, session_options);
}
// Apply ICE renomination flag.
for (auto& options : session_options->media_description_options) {
options.transport_options.enable_ice_renomination =
pc_->configuration()->enable_ice_renomination;
}
session_options->rtcp_cname = rtcp_cname_;
session_options->crypto_options = pc_->GetCryptoOptions();
session_options->pooled_ice_credentials =
context_->network_thread()->BlockingCall(
[this] { return port_allocator()->GetPooledIceCredentials(); });
}
void SdpOfferAnswerHandler::GetOptionsForPlanBAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
MediaSessionOptions* session_options) {
bool send_audio = false;
bool recv_audio = false;
bool send_video = false;
bool recv_video = false;
if (ConfiguredForMedia()) {
// Figure out transceiver directional preferences.
send_audio =
!rtp_manager()->GetAudioTransceiver()->internal()->senders().empty();
send_video =
!rtp_manager()->GetVideoTransceiver()->internal()->senders().empty();
// By default, generate sendrecv/recvonly m= sections. The direction is also
// restricted by the direction in the offer.
recv_audio = true;
recv_video = true;
// The "offer_to_receive_X" options allow those defaults to be overridden.
if (offer_answer_options.offer_to_receive_audio !=
RTCOfferAnswerOptions::kUndefined) {
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
}
if (offer_answer_options.offer_to_receive_video !=
RTCOfferAnswerOptions::kUndefined) {
recv_video = (offer_answer_options.offer_to_receive_video > 0);
}
}
std::optional<size_t> audio_index;
std::optional<size_t> video_index;
std::optional<size_t> data_index;
// Generate m= sections that match those in the offer.
// Note that mediasession.cc will handle intersection our preferred
// direction with the offered direction.
GenerateMediaDescriptionOptions(
remote_description(),
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), &audio_index,
&video_index, &data_index, session_options);
MediaDescriptionOptions* audio_media_description_options =
!audio_index ? nullptr
: &session_options->media_description_options[*audio_index];
MediaDescriptionOptions* video_media_description_options =
!video_index ? nullptr
: &session_options->media_description_options[*video_index];
if (ConfiguredForMedia()) {
AddPlanBRtpSenderOptions(rtp_manager()->GetSendersInternal(),
audio_media_description_options,
video_media_description_options,
offer_answer_options.num_simulcast_layers);
}
}
void SdpOfferAnswerHandler::GetOptionsForUnifiedPlanAnswer(
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
MediaSessionOptions* session_options) {
// Rules for generating an answer are dictated by JSEP sections 5.3.1 (Initial
// Answers) and 5.3.2 (Subsequent Answers).
RTC_DCHECK(remote_description());
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
for (const ContentInfo& content :
remote_description()->description()->contents()) {
MediaType media_type = content.media_description()->type();
if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) {
auto transceiver = transceivers()->FindByMid(content.mid());
if (transceiver) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForTransceiver(
transceiver->internal(), content.mid(),
/*is_create_offer=*/false));
} else {
// This should only happen with rejected transceivers.
RTC_DCHECK(content.rejected);
session_options->media_description_options.push_back(
MediaDescriptionOptions(media_type, content.mid(),
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
}
} else if (media_type == MediaType::UNSUPPORTED) {
RTC_DCHECK(content.rejected);
session_options->media_description_options.push_back(
MediaDescriptionOptions(media_type, content.mid(),
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
RTC_CHECK_EQ(MediaType::DATA, media_type);
// Reject all data sections if data channels are disabled.
// Reject a data section if it has already been rejected.
// Reject all data sections except for the first one.
auto sctp_mid = pc_->sctp_mid();
if (content.rejected ||
(sctp_mid.has_value() && content.mid() != *sctp_mid)) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(content.mid()));
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(content.mid()));
}
}
}
}
const char* SdpOfferAnswerHandler::SessionErrorToString(
SessionError error) const {
switch (error) {
case SessionError::kNone:
return "ERROR_NONE";
case SessionError::kContent:
return "ERROR_CONTENT";
case SessionError::kTransport:
return "ERROR_TRANSPORT";
}
RTC_DCHECK_NOTREACHED();
return "";
}
std::string SdpOfferAnswerHandler::GetSessionErrorMsg() {
RTC_DCHECK_RUN_ON(signaling_thread());
StringBuilder desc;
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
desc << kSessionErrorDesc << session_error_desc() << ".";
return desc.Release();
}
void SdpOfferAnswerHandler::SetSessionError(SessionError error,
const std::string& error_desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (error != session_error_) {
session_error_ = error;
session_error_desc_ = error_desc;
}
}
RTCError SdpOfferAnswerHandler::HandleLegacyOfferOptions(
const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(IsUnifiedPlan());
if (options.offer_to_receive_audio == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(MediaType::AUDIO);
} else if (options.offer_to_receive_audio == 1) {
AddUpToOneReceivingTransceiverOfType(MediaType::AUDIO);
} else if (options.offer_to_receive_audio > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_audio > 1 is not supported.");
}
if (options.offer_to_receive_video == 0) {
RemoveRecvDirectionFromReceivingTransceiversOfType(MediaType::VIDEO);
} else if (options.offer_to_receive_video == 1) {
AddUpToOneReceivingTransceiverOfType(MediaType::VIDEO);
} else if (options.offer_to_receive_video > 1) {
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
"offer_to_receive_video > 1 is not supported.");
}
return RTCError::OK();
}
void SdpOfferAnswerHandler::RemoveRecvDirectionFromReceivingTransceiversOfType(
MediaType media_type) {
for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
RtpTransceiverDirection new_direction =
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
if (new_direction != transceiver->direction()) {
RTC_LOG(LS_INFO) << "Changing " << MediaTypeToString(media_type)
<< " transceiver (MID="
<< transceiver->mid().value_or("<not set>") << ") from "
<< RtpTransceiverDirectionToString(
transceiver->direction())
<< " to "
<< RtpTransceiverDirectionToString(new_direction)
<< " since CreateOffer specified offer_to_receive=0";
transceiver->internal()->set_direction(new_direction);
}
}
}
void SdpOfferAnswerHandler::AddUpToOneReceivingTransceiverOfType(
MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (GetReceivingTransceiversOfType(media_type).empty()) {
RTC_LOG(LS_INFO)
<< "Adding one recvonly " << MediaTypeToString(media_type)
<< " transceiver since CreateOffer specified offer_to_receive=1";
RtpTransceiverInit init;
init.direction = RtpTransceiverDirection::kRecvOnly;
pc_->AddTransceiver(media_type, nullptr, init,
/*update_negotiation_needed=*/false);
}
}
std::vector<scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
SdpOfferAnswerHandler::GetReceivingTransceiversOfType(MediaType media_type) {
std::vector<scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
receiving_transceivers;
for (const auto& transceiver : transceivers()->List()) {
if (!transceiver->stopped() && transceiver->media_type() == media_type &&
RtpTransceiverDirectionHasRecv(transceiver->direction())) {
receiving_transceivers.push_back(transceiver);
}
}
return receiving_transceivers;
}
void SdpOfferAnswerHandler::ProcessRemovalOfRemoteTrack(
scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> transceiver,
std::vector<scoped_refptr<RtpTransceiverInterface>>* remove_list,
std::vector<scoped_refptr<MediaStreamInterface>>* removed_streams) {
RTC_DCHECK(transceiver->mid());
RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
<< *transceiver->mid();
std::vector<scoped_refptr<MediaStreamInterface>> previous_streams =
transceiver->internal()->receiver_internal()->streams();
// This will remove the remote track from the streams.
transceiver->internal()->receiver_internal()->set_stream_ids({});
remove_list->push_back(transceiver);
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
}
void SdpOfferAnswerHandler::RemoveRemoteStreamsIfEmpty(
const std::vector<scoped_refptr<MediaStreamInterface>>& remote_streams,
std::vector<scoped_refptr<MediaStreamInterface>>* removed_streams) {
RTC_DCHECK_RUN_ON(signaling_thread());
// TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead of
// streams, see if the stream was removed by checking if this was the last
// receiver with that stream ID.
for (const auto& remote_stream : remote_streams) {
if (remote_stream->GetAudioTracks().empty() &&
remote_stream->GetVideoTracks().empty()) {
remote_streams_->RemoveStream(remote_stream.get());
removed_streams->push_back(remote_stream);
}
}
}
void SdpOfferAnswerHandler::RemoveSenders(MediaType media_type) {
RTC_DCHECK_RUN_ON(signaling_thread());
UpdateLocalSenders(std::vector<StreamParams>(), media_type);
UpdateRemoteSendersList(std::vector<StreamParams>(), false, media_type,
nullptr);
}
void SdpOfferAnswerHandler::UpdateLocalSenders(
const std::vector<StreamParams>& streams,
MediaType media_type) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateLocalSenders");
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<RtpSenderInfo>* current_senders =
rtp_manager()->GetLocalSenderInfos(media_type);
// Find removed tracks. I.e., tracks where the track id, stream id or ssrc
// don't match the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const StreamParams* params = GetStreamBySsrc(streams, info.first_ssrc);
if (!params || params->id != info.sender_id ||
params->first_stream_id() != info.stream_id) {
rtp_manager()->OnLocalSenderRemoved(info, media_type);
sender_it = current_senders->erase(sender_it);
} else {
++sender_it;
}
}
// Find new and active senders.
for (const StreamParams& params : streams) {
// The sync_label is the MediaStream label and the `stream.id` is the
// sender id.
const std::string& stream_id = params.first_stream_id();
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
const RtpSenderInfo* sender_info =
rtp_manager()->FindSenderInfo(*current_senders, stream_id, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
rtp_manager()->OnLocalSenderAdded(current_senders->back(), media_type);
}
}
}
void SdpOfferAnswerHandler::UpdateRemoteSendersList(
const StreamParamsVec& streams,
bool default_sender_needed,
MediaType media_type,
StreamCollection* new_streams) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::UpdateRemoteSendersList");
RTC_DCHECK_RUN_ON(signaling_thread());
RTC_DCHECK(!IsUnifiedPlan());
std::vector<RtpSenderInfo>* current_senders =
rtp_manager()->GetRemoteSenderInfos(media_type);
// Find removed senders. I.e., senders where the sender id or ssrc don't match
// the new StreamParam.
for (auto sender_it = current_senders->begin();
sender_it != current_senders->end();
/* incremented manually */) {
const RtpSenderInfo& info = *sender_it;
const StreamParams* params = GetStreamBySsrc(streams, info.first_ssrc);
std::string params_stream_id;
if (params) {
params_stream_id =
(!params->first_stream_id().empty() ? params->first_stream_id()
: kDefaultStreamId);
}
bool sender_exists = params && params->id == info.sender_id &&
params_stream_id == info.stream_id;
// If this is a default track, and we still need it, don't remove it.
if ((info.stream_id == kDefaultStreamId && default_sender_needed) ||
sender_exists) {
++sender_it;
} else {
rtp_manager()->OnRemoteSenderRemoved(
info, remote_streams_->find(info.stream_id), media_type);
sender_it = current_senders->erase(sender_it);
}
}
// Find new and active senders.
for (const StreamParams& params : streams) {
if (!params.has_ssrcs()) {
// The remote endpoint has streams, but didn't signal ssrcs. For an active
// sender, this means it is coming from a Unified Plan endpoint,so we just
// create a default.
default_sender_needed = true;
break;
}
// `params.id` is the sender id and the stream id uses the first of
// `params.stream_ids`. The remote description could come from a Unified
// Plan endpoint, with multiple or no stream_ids() signaled. Since this is
// not supported in Plan B, we just take the first here and create the
// default stream ID if none is specified.
const std::string& stream_id =
(!params.first_stream_id().empty() ? params.first_stream_id()
: kDefaultStreamId);
const std::string& sender_id = params.id;
uint32_t ssrc = params.first_ssrc();
scoped_refptr<MediaStreamInterface> stream(
remote_streams_->find(stream_id));
if (!stream) {
// This is a new MediaStream. Create a new remote MediaStream.
stream = MediaStreamProxy::Create(Thread::Current(),
MediaStream::Create(stream_id));
remote_streams_->AddStream(stream);
new_streams->AddStream(stream);
}
const RtpSenderInfo* sender_info =
rtp_manager()->FindSenderInfo(*current_senders, stream_id, sender_id);
if (!sender_info) {
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
rtp_manager()->OnRemoteSenderAdded(current_senders->back(), stream.get(),
media_type);
}
}
// Add default sender if necessary.
if (default_sender_needed) {
scoped_refptr<MediaStreamInterface> default_stream(
remote_streams_->find(kDefaultStreamId));
if (!default_stream) {
// Create the new default MediaStream.
default_stream = MediaStreamProxy::Create(
Thread::Current(), MediaStream::Create(kDefaultStreamId));
remote_streams_->AddStream(default_stream);
new_streams->AddStream(default_stream);
}
std::string default_sender_id = (media_type == MediaType::AUDIO)
? kDefaultAudioSenderId
: kDefaultVideoSenderId;
const RtpSenderInfo* default_sender_info = rtp_manager()->FindSenderInfo(
*current_senders, kDefaultStreamId, default_sender_id);
if (!default_sender_info) {
current_senders->push_back(
RtpSenderInfo(kDefaultStreamId, default_sender_id, /*ssrc=*/0));
rtp_manager()->OnRemoteSenderAdded(current_senders->back(),
default_stream.get(), media_type);
}
}
}
void SdpOfferAnswerHandler::EnableSending() {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::EnableSending");
RTC_DCHECK_RUN_ON(signaling_thread());
if (!ConfiguredForMedia()) {
return;
}
for (const auto& transceiver : transceivers()->ListInternal()) {
ChannelInterface* channel = transceiver->channel();
if (channel) {
channel->Enable(true);
}
}
}
RTCError SdpOfferAnswerHandler::PushdownMediaDescription(
SdpType type,
ContentSource source,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownMediaDescription");
RTC_DCHECK_RUN_ON(signaling_thread());
const SessionDescriptionInterface* sdesc =
(source == CS_LOCAL ? local_description() : remote_description());
RTC_DCHECK(sdesc);
if (ConfiguredForMedia()) {
// Note: This will perform a BlockingCall over to the worker thread, which
// we'll also do in a loop below.
if (!UpdatePayloadTypeDemuxingState(source, bundle_groups_by_mid)) {
// Note that this is never expected to fail, since RtpDemuxer doesn't
// return an error when changing payload type demux criteria, which is all
// this does.
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to update payload type demuxing state.");
}
// Push down the new SDP media section for each audio/video transceiver.
auto rtp_transceivers = transceivers()->ListInternal();
std::vector<std::pair<ChannelInterface*, const MediaContentDescription*>>
channels;
std::optional<RtcpFeedbackType> preferred_rtcp_cc_ack_type;
for (const auto& transceiver : rtp_transceivers) {
const ContentInfo* content_info =
FindMediaSectionForTransceiver(transceiver, sdesc);
ChannelInterface* channel = transceiver->channel();
if (!channel || !content_info || content_info->rejected) {
continue;
}
const MediaContentDescription* content_desc =
content_info->media_description();
if (!content_desc) {
continue;
}
if (preferred_rtcp_cc_ack_type.has_value()) {
// RFC 8888 says that the ccfb must be consistent across the
// description.
if (preferred_rtcp_cc_ack_type == RtcpFeedbackType::CCFB &&
preferred_rtcp_cc_ack_type !=
content_desc->preferred_rtcp_cc_ack_type()) {
RTC_LOG(LS_ERROR)
<< "Warning: Inconsistent CCFB flag - ack type changed to "
<< content_desc->preferred_rtcp_cc_ack_type();
preferred_rtcp_cc_ack_type =
content_desc->preferred_rtcp_cc_ack_type();
}
} else {
preferred_rtcp_cc_ack_type = content_desc->preferred_rtcp_cc_ack_type();
}
transceiver->OnNegotiationUpdate(type, content_desc);
channels.push_back(std::make_pair(channel, content_desc));
}
// This for-loop of invokes helps audio impairment during re-negotiations.
// One of the causes is that downstairs decoder creation is synchronous at
// the moment, and that a decoder is created for each codec listed in the
// SDP.
//
// TODO(bugs.webrtc.org/12840): consider merging the invokes again after
// these projects have shipped:
// - bugs.webrtc.org/12462
// - crbug.com/1157227
// - crbug.com/1187289
for (const auto& entry : channels) {
std::string error;
bool success = context_->worker_thread()->BlockingCall([&]() {
return (source == CS_LOCAL)
? entry.first->SetLocalContent(entry.second, type, error)
: entry.first->SetRemoteContent(entry.second, type, error);
});
if (!success) {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
}
}
// If local and remote are both set, we assume that it's safe to trigger
// CCFB.
if (pc_->trials().IsEnabled("WebRTC-RFC8888CongestionControlFeedback")) {
if (type == SdpType::kAnswer && local_description() &&
remote_description()) {
std::optional<RtcpFeedbackType> remote_preferred_rtcp_cc_ack_type;
// Verify that the remote agrees on congestion control feedback format.
for (const auto& content :
remote_description()->description()->contents()) {
if (content.type != MediaProtocolType::kRtp || content.rejected ||
content.media_description() == nullptr) {
continue;
}
if (!remote_preferred_rtcp_cc_ack_type.has_value()) {
remote_preferred_rtcp_cc_ack_type =
content.media_description()->preferred_rtcp_cc_ack_type();
}
if (content.media_description()
->preferred_rtcp_cc_ack_type()
.has_value() &&
remote_preferred_rtcp_cc_ack_type !=
content.media_description()->preferred_rtcp_cc_ack_type()) {
RTC_LOG(LS_ERROR) << "Warning: Inconsistent remote congestion "
"control feedback types. ";
remote_preferred_rtcp_cc_ack_type = std::nullopt;
break;
}
}
if (preferred_rtcp_cc_ack_type.has_value() &&
preferred_rtcp_cc_ack_type == remote_preferred_rtcp_cc_ack_type) {
// The call and the congestion controller live on the worker thread.
context_->worker_thread()->PostTask(
[call = pc_->call_ptr(),
preferred_rtcp_cc_ack_type = *preferred_rtcp_cc_ack_type] {
call->SetPreferredRtcpCcAckType(preferred_rtcp_cc_ack_type);
});
} else {
RTC_LOG(LS_WARNING)
<< "Inconsistent Congestion Control feedback types: "
<< preferred_rtcp_cc_ack_type << " vs "
<< remote_preferred_rtcp_cc_ack_type << " Using default.";
}
}
}
}
// 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 (pc_->sctp_mid() && local_description() && remote_description()) {
auto local_sctp_description =
GetFirstSctpDataContentDescription(local_description()->description());
auto remote_sctp_description =
GetFirstSctpDataContentDescription(remote_description()->description());
if (local_sctp_description && remote_sctp_description) {
int max_message_size;
// A remote max message size of zero means "any size supported".
// We configure the connection with our own max message size.
if (remote_sctp_description->max_message_size() == 0) {
max_message_size = local_sctp_description->max_message_size();
} else {
max_message_size =
std::min(local_sctp_description->max_message_size(),
remote_sctp_description->max_message_size());
}
pc_->StartSctpTransport({.local_port = local_sctp_description->port(),
.remote_port = remote_sctp_description->port(),
.max_message_size = max_message_size});
}
}
return RTCError::OK();
}
RTCError SdpOfferAnswerHandler::PushdownTransportDescription(
ContentSource source,
SdpType type) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::PushdownTransportDescription");
RTC_DCHECK_RUN_ON(signaling_thread());
if (source == CS_LOCAL) {
const SessionDescriptionInterface* sdesc = local_description();
RTC_DCHECK(sdesc);
const auto* remote = remote_description();
return transport_controller_s()->SetLocalDescription(
type, sdesc->description(), remote ? remote->description() : nullptr);
} else {
const SessionDescriptionInterface* sdesc = remote_description();
RTC_DCHECK(sdesc);
const auto* local = local_description();
return transport_controller_s()->SetRemoteDescription(
type, local ? local->description() : nullptr, sdesc->description());
}
}
void SdpOfferAnswerHandler::RemoveStoppedTransceivers() {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::RemoveStoppedTransceivers");
RTC_DCHECK_RUN_ON(signaling_thread());
// 3.2.10.1: For each transceiver in the connection's set of transceivers
// run the following steps:
if (!IsUnifiedPlan())
return;
if (!ConfiguredForMedia()) {
return;
}
// Traverse a copy of the transceiver list.
auto transceiver_list = transceivers()->List();
for (auto transceiver : transceiver_list) {
// 3.2.10.1.1: If transceiver is stopped, associated with an m= section
// and the associated m= section is rejected in
// connection.[[CurrentLocalDescription]] or
// connection.[[CurrentRemoteDescription]], remove the
// transceiver from the connection's set of transceivers.
if (!transceiver->stopped()) {
continue;
}
const ContentInfo* local_content = FindMediaSectionForTransceiver(
transceiver->internal(), local_description());
const ContentInfo* remote_content = FindMediaSectionForTransceiver(
transceiver->internal(), remote_description());
if ((local_content && local_content->rejected) ||
(remote_content && remote_content->rejected)) {
RTC_LOG(LS_INFO) << "Dissociating transceiver"
" since the media section is being recycled.";
transceiver->internal()->set_mid(std::nullopt);
transceiver->internal()->set_mline_index(std::nullopt);
} else if (!local_content && !remote_content) {
// TODO(bugs.webrtc.org/11973): Consider if this should be removed already
// See https://github.com/w3c/webrtc-pc/issues/2576
RTC_LOG(LS_INFO)
<< "Dropping stopped transceiver that was never associated";
}
transceivers()->Remove(transceiver);
}
}
void SdpOfferAnswerHandler::RemoveUnusedChannels(
const SessionDescription* desc) {
RTC_DCHECK_RUN_ON(signaling_thread());
if (ConfiguredForMedia()) {
// Destroy video channel first since it may have a pointer to the
// voice channel.
const ContentInfo* video_info = GetFirstVideoContent(desc);
if (!video_info || video_info->rejected) {
rtp_manager()->GetVideoTransceiver()->internal()->ClearChannel();
}
const ContentInfo* audio_info = GetFirstAudioContent(desc);
if (!audio_info || audio_info->rejected) {
rtp_manager()->GetAudioTransceiver()->internal()->ClearChannel();
}
}
const ContentInfo* data_info = GetFirstDataContent(desc);
if (!data_info) {
RTCError error(RTCErrorType::OPERATION_ERROR_WITH_DATA,
"No data channel section in the description.");
error.set_error_detail(RTCErrorDetailType::DATA_CHANNEL_FAILURE);
pc_->DestroyDataChannelTransport(error);
} else if (data_info->rejected) {
StringBuilder sb;
sb << "Rejected data channel with mid=" << data_info->mid() << ".";
RTCError error(RTCErrorType::OPERATION_ERROR_WITH_DATA, sb.Release());
error.set_error_detail(RTCErrorDetailType::DATA_CHANNEL_FAILURE);
pc_->DestroyDataChannelTransport(error);
}
}
void SdpOfferAnswerHandler::UpdateEndedRemoteMediaStreams() {
RTC_DCHECK_RUN_ON(signaling_thread());
std::vector<scoped_refptr<MediaStreamInterface>> streams_to_remove;
for (size_t i = 0; i < remote_streams_->count(); ++i) {
MediaStreamInterface* stream = remote_streams_->at(i);
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
streams_to_remove.push_back(scoped_refptr<MediaStreamInterface>(stream));
}
}
pc_->RunWithObserver([&](auto observer) {
RTC_DCHECK_RUN_ON(signaling_thread());
for (auto& stream : streams_to_remove) {
remote_streams_->RemoveStream(stream.get());
observer->OnRemoveStream(std::move(stream));
}
});
}
bool SdpOfferAnswerHandler::UseCandidatesInRemoteDescription() {
RTC_DCHECK_RUN_ON(signaling_thread());
auto* remote_desc = remote_description();
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 IceCandidate* candidate = candidates->at(n);
bool valid = false;
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
if (valid) {
RTC_LOG(LS_INFO)
<< "UseCandidatesInRemoteDescription: Not ready to use "
"candidate.";
}
continue;
}
ret = UseCandidate(candidate);
if (!ret) {
break;
}
}
}
return ret;
}
bool SdpOfferAnswerHandler::UseCandidate(const IceCandidate* candidate) {
RTC_DCHECK_RUN_ON(signaling_thread());
Thread::ScopedDisallowBlockingCalls no_blocking_calls;
RTCErrorOr<const ContentInfo*> result =
FindContentInfo(remote_description(), candidate);
if (!result.ok())
return false;
const Candidate& c = candidate->candidate();
RTCError error = VerifyCandidate(c);
if (!error.ok()) {
RTC_LOG(LS_WARNING) << "Invalid candidate: " << c.ToString();
return true;
}
pc_->AddRemoteCandidate(result.value()->mid(), c);
return true;
}
// 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 SdpOfferAnswerHandler::ReadyToUseRemoteCandidate(
const IceCandidate* candidate,
const SessionDescriptionInterface* remote_desc,
bool* valid) {
RTC_DCHECK_RUN_ON(signaling_thread());
*valid = true;
const SessionDescriptionInterface* current_remote_desc =
remote_desc ? remote_desc : remote_description();
if (!current_remote_desc) {
return false;
}
RTCErrorOr<const ContentInfo*> result =
FindContentInfo(current_remote_desc, candidate);
if (!result.ok()) {
RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate. "
<< result.error().message();
*valid = false;
return false;
}
if (has_sdp_munged_ufrag_) {
// The format is
// restricted_addresses:<host-with-wildcard>:<port>|<other-host>:<other-port>
// The host can contain wildcards that match anything that comes after it.
// The port should either be a wildcard or a specific port.
const std::string restricted_addresses =
pc_->trials().Lookup("WebRTC-NoSdpMangleUfragRestrictedAddresses");
const std::string port = candidate->candidate().address().PortAsString();
const std::string host = candidate->candidate().address().HostAsURIString();
const std::vector<absl::string_view> restricted_address_list =
absl::StrSplit(restricted_addresses, '|');
bool allowed = true;
for (const absl::string_view restricted_address : restricted_address_list) {
const std::pair<absl::string_view, absl::string_view> address =
absl::StrSplit(restricted_address, ':');
if ((address.second == port || address.second == "*") &&
WildcardHostPrefixMatch(host, address.first)) {
RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Candidate not valid "
"because of SDP munging.";
*valid = false;
// There might be other types of SDP munging, but here we're only
// interested in IceUfrag and IcePwd.
SdpMungingType sdp_munging_type =
last_sdp_munging_type_ == SdpMungingType::kIcePwd
? SdpMungingType::kIcePwd
: SdpMungingType::kIceUfrag;
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.RestrictedCandidates.SdpMungingType",
sdp_munging_type, SdpMungingType::kMaxValue);
RTC_HISTOGRAM_ENUMERATION_SPARSE(
"WebRTC.PeerConnection.RestrictedCandidates.Port",
candidate->candidate().address().port(), 65536);
allowed = false;
break;
}
}
RTC_HISTOGRAM_BOOLEAN(
"WebRTC.PeerConnection.RestrictedCandidates.MungeAllowed", allowed);
return allowed;
}
return true;
}
RTCErrorOr<const ContentInfo*> SdpOfferAnswerHandler::FindContentInfo(
const SessionDescriptionInterface* description,
const IceCandidate* candidate) {
if (!candidate->sdp_mid().empty()) {
auto& contents = description->description()->contents();
auto it =
absl::c_find_if(contents, [candidate](const ContentInfo& content_info) {
return content_info.mid() == candidate->sdp_mid();
});
if (it == contents.end()) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INVALID_PARAMETER,
"Mid " + candidate->sdp_mid() +
" specified but no media section with that mid found.");
} else {
return &*it;
}
} else if (candidate->sdp_mline_index() >= 0) {
size_t mediacontent_index =
static_cast<size_t>(candidate->sdp_mline_index());
size_t content_size = description->description()->contents().size();
if (mediacontent_index < content_size) {
return &description->description()->contents()[mediacontent_index];
} else {
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
"Media line index (" +
absl::StrCat(candidate->sdp_mline_index()) +
") out of range (number of mlines: " +
absl::StrCat(content_size) + ").");
}
}
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
"Neither sdp_mline_index nor sdp_mid specified.");
}
RTCError SdpOfferAnswerHandler::CreateChannels(const SessionDescription& desc) {
TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::CreateChannels");
// Creating the media channels. Transports should already have been created
// at this point.
RTC_DCHECK_RUN_ON(signaling_thread());
const ContentInfo* voice = GetFirstAudioContent(&desc);
if (voice && !voice->rejected &&
!rtp_manager()->GetAudioTransceiver()->internal()->channel()) {
auto error =
rtp_manager()->GetAudioTransceiver()->internal()->CreateChannel(
voice->mid(), pc_->call_ptr(), pc_->configuration()->media_config,
pc_->SrtpRequired(), pc_->GetCryptoOptions(), audio_options(),
video_options(), video_bitrate_allocator_factory_.get(),
[&](absl::string_view mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
if (!error.ok()) {
return error;
}
}
const ContentInfo* video = GetFirstVideoContent(&desc);
if (video && !video->rejected &&
!rtp_manager()->GetVideoTransceiver()->internal()->channel()) {
auto error =
rtp_manager()->GetVideoTransceiver()->internal()->CreateChannel(
video->mid(), pc_->call_ptr(), pc_->configuration()->media_config,
pc_->SrtpRequired(), pc_->GetCryptoOptions(),
audio_options(), video_options(),
video_bitrate_allocator_factory_.get(), [&](absl::string_view mid) {
RTC_DCHECK_RUN_ON(network_thread());
return transport_controller_n()->GetRtpTransport(mid);
});
if (!error.ok()) {
return error;
}
}
const ContentInfo* data = GetFirstDataContent(&desc);
if (data && !data->rejected &&
!pc_->CreateDataChannelTransport(data->mid())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create data channel.");
}
return RTCError::OK();
}
void SdpOfferAnswerHandler::DestroyMediaChannels() {
RTC_DCHECK_RUN_ON(signaling_thread());
if (!transceivers()) {
return;
}
RTC_LOG_THREAD_BLOCK_COUNT();
// Destroy video channels first since they may have a pointer to a voice
// channel.
auto list = transceivers()->List();
RTC_DCHECK_BLOCK_COUNT_NO_MORE_THAN(0);
for (const auto& transceiver : list) {
if (transceiver->media_type() == MediaType::VIDEO) {
transceiver->internal()->ClearChannel();
}
}
for (const auto& transceiver : list) {
if (transceiver->media_type() == MediaType::AUDIO) {
transceiver->internal()->ClearChannel();
}
}
}
void SdpOfferAnswerHandler::GenerateMediaDescriptionOptions(
const SessionDescriptionInterface* session_desc,
RtpTransceiverDirection audio_direction,
RtpTransceiverDirection video_direction,
std::optional<size_t>* audio_index,
std::optional<size_t>* video_index,
std::optional<size_t>* data_index,
MediaSessionOptions* session_options) {
RTC_DCHECK_RUN_ON(signaling_thread());
for (const ContentInfo& content : session_desc->description()->contents()) {
if (IsAudioContent(&content)) {
// If we already have an audio m= section, reject this extra one.
if (*audio_index) {
session_options->media_description_options.push_back(
MediaDescriptionOptions(MediaType::AUDIO, content.mid(),
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
bool stopped = (audio_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
MediaDescriptionOptions(MediaType::AUDIO, content.mid(),
audio_direction, stopped));
*audio_index = session_options->media_description_options.size() - 1;
}
session_options->media_description_options.back().header_extensions =
media_engine()->voice().GetRtpHeaderExtensions(&env_.field_trials());
} else if (IsVideoContent(&content)) {
// If we already have an video m= section, reject this extra one.
if (*video_index) {
session_options->media_description_options.push_back(
MediaDescriptionOptions(MediaType::VIDEO, content.mid(),
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
bool stopped = (video_direction == RtpTransceiverDirection::kInactive);
session_options->media_description_options.push_back(
MediaDescriptionOptions(MediaType::VIDEO, content.mid(),
video_direction, stopped));
*video_index = session_options->media_description_options.size() - 1;
}
session_options->media_description_options.back().header_extensions =
media_engine()->video().GetRtpHeaderExtensions(&env_.field_trials());
} else if (IsUnsupportedContent(&content)) {
session_options->media_description_options.push_back(
MediaDescriptionOptions(MediaType::UNSUPPORTED, content.mid(),
RtpTransceiverDirection::kInactive,
/*stopped=*/true));
} else {
RTC_DCHECK(IsDataContent(&content));
// If we already have an data m= section, reject this extra one.
if (*data_index) {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForRejectedData(content.mid()));
} else {
session_options->media_description_options.push_back(
GetMediaDescriptionOptionsForActiveData(content.mid()));
*data_index = session_options->media_description_options.size() - 1;
}
}
}
}
MediaDescriptionOptions
SdpOfferAnswerHandler::GetMediaDescriptionOptionsForActiveData(
const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
// Direction for data sections is meaningless, but legacy endpoints might
// expect sendrecv.
MediaDescriptionOptions options(MediaType::DATA, mid,
RtpTransceiverDirection::kSendRecv,
/*stopped=*/false);
return options;
}
MediaDescriptionOptions
SdpOfferAnswerHandler::GetMediaDescriptionOptionsForRejectedData(
const std::string& mid) const {
RTC_DCHECK_RUN_ON(signaling_thread());
MediaDescriptionOptions options(MediaType::DATA, mid,
RtpTransceiverDirection::kInactive,
/*stopped=*/true);
return options;
}
bool SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState(
ContentSource source,
const std::map<std::string, const ContentGroup*>& bundle_groups_by_mid) {
TRACE_EVENT0("webrtc",
"SdpOfferAnswerHandler::UpdatePayloadTypeDemuxingState");
RTC_DCHECK_RUN_ON(signaling_thread());
// We may need to delete any created default streams and disable creation of
// new ones on the basis of payload type. This is needed to avoid SSRC
// collisions in Call's RtpDemuxer, in the case that a transceiver has
// created a default stream, and then some other channel gets the SSRC
// signaled in the corresponding Unified Plan "m=" section. Specifically, we
// need to disable payload type based demuxing when two bundled "m=" sections
// are using the same payload type(s). For more context
// see https://bugs.chromium.org/p/webrtc/issues/detail?id=11477
const SessionDescriptionInterface* sdesc =
(source == CS_LOCAL ? local_description() : remote_description());
struct PayloadTypes {
std::set<int> audio_payload_types;
std::set<int> video_payload_types;
bool pt_demuxing_possible_audio = true;
bool pt_demuxing_possible_video = true;
};
std::map<const ContentGroup*, PayloadTypes> payload_types_by_bundle;
// If the MID is missing from *any* receiving m= section, this is set to true.
bool mid_header_extension_missing_audio = false;
bool mid_header_extension_missing_video = false;
for (auto& content_info : sdesc->description()->contents()) {
auto it = bundle_groups_by_mid.find(content_info.mid());
const ContentGroup* bundle_group =
it != bundle_groups_by_mid.end() ? it->second : nullptr;
// If this m= section isn't bundled, it's safe to demux by payload type
// since other m= sections using the same payload type will also be using
// different transports.
if (!bundle_group) {
continue;
}
PayloadTypes* payload_types = &payload_types_by_bundle[bundle_group];
if (content_info.rejected ||
(source == ContentSource::CS_LOCAL &&
!RtpTransceiverDirectionHasRecv(
content_info.media_description()->direction())) ||
(source == ContentSource::CS_REMOTE &&
!RtpTransceiverDirectionHasSend(
content_info.media_description()->direction()))) {
// Ignore transceivers that are not receiving.
continue;
}
const MediaType media_type = content_info.media_description()->type();
if (media_type == MediaType::AUDIO || media_type == MediaType::VIDEO) {
if (media_type == MediaType::AUDIO &&
!mid_header_extension_missing_audio) {
mid_header_extension_missing_audio =
!ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
} else if (media_type == MediaType::VIDEO &&
!mid_header_extension_missing_video) {
mid_header_extension_missing_video =
!ContentHasHeaderExtension(content_info, RtpExtension::kMidUri);
}
const MediaContentDescription* media_desc =
content_info.media_description();
for (const Codec& codec : media_desc->codecs()) {
if (media_type == MediaType::AUDIO) {
if (payload_types->audio_payload_types.count(codec.id)) {
// Two m= sections are using the same payload type, thus demuxing
// by payload type is not possible.
if (media_type == MediaType::AUDIO) {
payload_types->pt_demuxing_possible_audio = false;
}
}
payload_types->audio_payload_types.insert(codec.id);
} else if (media_type == MediaType::VIDEO) {
if (payload_types->video_payload_types.count(codec.id)) {
// Two m= sections are using the same payload type, thus demuxing
// by payload type is not possible.
payload_types->pt_demuxing_possible_video = false;
}
payload_types->video_payload_types.insert(codec.id);
}
}
}
}
// In Unified Plan, payload type demuxing is useful for legacy endpoints that
// don't support the MID header extension, but it can also cause incorrrect
// forwarding of packets when going from one m= section to multiple m=
// sections in the same BUNDLE. This only happens if media arrives prior to
// negotiation, but this can cause missing video and unsignalled ssrc bugs
// severe enough to warrant disabling PT demuxing in such cases. Therefore, if
// a MID header extension is present on all m= sections for a given kind
// (audio/video) then we use that as an OK to disable payload type demuxing in
// BUNDLEs of that kind. However if PT demuxing was ever turned on (e.g. MID
// was ever removed on ANY m= section of that kind) then we continue to allow
// PT demuxing in order to prevent disabling it in follow-up O/A exchanges and
// allowing early media by PT.
bool bundled_pt_demux_allowed_audio = !IsUnifiedPlan() ||
mid_header_extension_missing_audio ||
pt_demuxing_has_been_used_audio_;
bool bundled_pt_demux_allowed_video = !IsUnifiedPlan() ||
mid_header_extension_missing_video ||
pt_demuxing_has_been_used_video_;
// Gather all updates ahead of time so that all channels can be updated in a
// single BlockingCall; necessary due to thread guards.
std::vector<std::pair<bool, ChannelInterface*>> channels_to_update;
for (const auto& transceiver : transceivers()->ListInternal()) {
ChannelInterface* channel = transceiver->channel();
const ContentInfo* content =
FindMediaSectionForTransceiver(transceiver, sdesc);
if (!channel || !content) {
continue;
}
const MediaType media_type = channel->media_type();
if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) {
continue;
}
RtpTransceiverDirection local_direction =
content->media_description()->direction();
if (source == CS_REMOTE) {
local_direction = RtpTransceiverDirectionReversed(local_direction);
}
auto bundle_it = bundle_groups_by_mid.find(channel->mid());
const ContentGroup* bundle_group =
bundle_it != bundle_groups_by_mid.end() ? bundle_it->second : nullptr;
bool pt_demux_enabled = RtpTransceiverDirectionHasRecv(local_direction);
if (media_type == MediaType::AUDIO) {
pt_demux_enabled &=
!bundle_group ||
(bundled_pt_demux_allowed_audio &&
payload_types_by_bundle[bundle_group].pt_demuxing_possible_audio);
if (pt_demux_enabled) {
pt_demuxing_has_been_used_audio_ = true;
}
} else {
RTC_DCHECK_EQ(media_type, MediaType::VIDEO);
pt_demux_enabled &=
!bundle_group ||
(bundled_pt_demux_allowed_video &&
payload_types_by_bundle[bundle_group].pt_demuxing_possible_video);
if (pt_demux_enabled) {
pt_demuxing_has_been_used_video_ = true;
}
}
channels_to_update.emplace_back(pt_demux_enabled, transceiver->channel());
}
if (channels_to_update.empty()) {
return true;
}
// TODO(bugs.webrtc.org/11993): This BlockingCall() will also block on the
// network thread for every demuxer sink that needs to be updated. The demuxer
// state needs to be fully (and only) managed on the network thread and once
// that's the case, there's no need to stop by on the worker. Ideally we could
// also do this without blocking.
return context_->worker_thread()->BlockingCall([&channels_to_update]() {
for (const auto& it : channels_to_update) {
if (!it.second->SetPayloadTypeDemuxingEnabled(it.first)) {
// Note that the state has already been irrevocably changed at this
// point. Is it useful to stop the loop?
return false;
}
}
return true;
});
}
bool SdpOfferAnswerHandler::ConfiguredForMedia() const {
return context_->media_engine();
}
} // namespace webrtc