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webrtc / src / d2098933e1c392e248f603b5601305c290b5b901 / . / pc / media_session.cc
blob: a763919c160855547ac388f0d6b24050eea50e3b [file] [log] [blame]
/*
* Copyright 2004 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/media_session.h"
#include <stddef.h>
#include <algorithm>
#include <map>
#include <string>
#include <unordered_map>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/crypto_params.h"
#include "media/base/codec.h"
#include "media/base/media_constants.h"
#include "media/base/media_engine.h"
#include "media/base/sdp_video_format_utils.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "pc/media_protocol_names.h"
#include "pc/rtp_media_utils.h"
#include "pc/used_ids.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/third_party/base64/base64.h"
#include "rtc_base/unique_id_generator.h"
namespace {
using rtc::UniqueRandomIdGenerator;
using webrtc::RTCError;
using webrtc::RTCErrorType;
using webrtc::RtpTransceiverDirection;
const char kInline[] = "inline:";
void GetSupportedSdesCryptoSuiteNames(
void (*func)(const webrtc::CryptoOptions&, std::vector<int>*),
const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* names) {
std::vector<int> crypto_suites;
func(crypto_options, &crypto_suites);
for (const auto crypto : crypto_suites) {
names->push_back(rtc::SrtpCryptoSuiteToName(crypto));
}
}
webrtc::RtpExtension RtpExtensionFromCapability(
const webrtc::RtpHeaderExtensionCapability& capability) {
return webrtc::RtpExtension(capability.uri,
capability.preferred_id.value_or(1));
}
cricket::RtpHeaderExtensions RtpHeaderExtensionsFromCapabilities(
const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities) {
cricket::RtpHeaderExtensions exts;
for (const auto& capability : capabilities) {
exts.push_back(RtpExtensionFromCapability(capability));
}
return exts;
}
std::vector<webrtc::RtpHeaderExtensionCapability>
UnstoppedRtpHeaderExtensionCapabilities(
std::vector<webrtc::RtpHeaderExtensionCapability> capabilities) {
capabilities.erase(
std::remove_if(
capabilities.begin(), capabilities.end(),
[](const webrtc::RtpHeaderExtensionCapability& capability) {
return capability.direction == RtpTransceiverDirection::kStopped;
}),
capabilities.end());
return capabilities;
}
bool IsCapabilityPresent(const webrtc::RtpHeaderExtensionCapability& capability,
const cricket::RtpHeaderExtensions& extensions) {
return std::find_if(extensions.begin(), extensions.end(),
[&capability](const webrtc::RtpExtension& extension) {
return capability.uri == extension.uri;
}) != extensions.end();
}
cricket::RtpHeaderExtensions UnstoppedOrPresentRtpHeaderExtensions(
const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities,
const cricket::RtpHeaderExtensions& unencrypted,
const cricket::RtpHeaderExtensions& encrypted) {
cricket::RtpHeaderExtensions extensions;
for (const auto& capability : capabilities) {
if (capability.direction != RtpTransceiverDirection::kStopped ||
IsCapabilityPresent(capability, unencrypted) ||
IsCapabilityPresent(capability, encrypted)) {
extensions.push_back(RtpExtensionFromCapability(capability));
}
}
return extensions;
}
} // namespace
namespace cricket {
namespace {
bool IsRtxCodec(const Codec& codec) {
return absl::EqualsIgnoreCase(codec.name, kRtxCodecName);
}
bool IsRtxCodec(const webrtc::RtpCodecCapability& capability) {
return absl::EqualsIgnoreCase(capability.name, kRtxCodecName);
}
bool ContainsRtxCodec(const std::vector<Codec>& codecs) {
for (const auto& codec : codecs) {
if (IsRtxCodec(codec)) {
return true;
}
}
return false;
}
bool IsRedCodec(const Codec& codec) {
return absl::EqualsIgnoreCase(codec.name, kRedCodecName);
}
bool IsRedCodec(const webrtc::RtpCodecCapability& capability) {
return absl::EqualsIgnoreCase(capability.name, kRedCodecName);
}
bool IsFlexfecCodec(const Codec& codec) {
return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
}
bool ContainsFlexfecCodec(const std::vector<Codec>& codecs) {
for (const auto& codec : codecs) {
if (IsFlexfecCodec(codec)) {
return true;
}
}
return false;
}
bool IsUlpfecCodec(const Codec& codec) {
return absl::EqualsIgnoreCase(codec.name, kUlpfecCodecName);
}
bool IsComfortNoiseCodec(const Codec& codec) {
return absl::EqualsIgnoreCase(codec.name, kComfortNoiseCodecName);
}
RtpTransceiverDirection NegotiateRtpTransceiverDirection(
RtpTransceiverDirection offer,
RtpTransceiverDirection wants) {
bool offer_send = webrtc::RtpTransceiverDirectionHasSend(offer);
bool offer_recv = webrtc::RtpTransceiverDirectionHasRecv(offer);
bool wants_send = webrtc::RtpTransceiverDirectionHasSend(wants);
bool wants_recv = webrtc::RtpTransceiverDirectionHasRecv(wants);
return webrtc::RtpTransceiverDirectionFromSendRecv(offer_recv && wants_send,
offer_send && wants_recv);
}
bool IsMediaContentOfType(const ContentInfo* content, MediaType media_type) {
if (!content || !content->media_description()) {
return false;
}
return content->media_description()->type() == media_type;
}
bool CreateCryptoParams(int tag,
const std::string& cipher,
CryptoParams* crypto_out) {
int key_len;
int salt_len;
if (!rtc::GetSrtpKeyAndSaltLengths(rtc::SrtpCryptoSuiteFromName(cipher),
&key_len, &salt_len)) {
return false;
}
int master_key_len = key_len + salt_len;
std::string master_key;
if (!rtc::CreateRandomData(master_key_len, &master_key)) {
return false;
}
RTC_CHECK_EQ(master_key_len, master_key.size());
std::string key = rtc::Base64::Encode(master_key);
crypto_out->tag = tag;
crypto_out->crypto_suite = cipher;
crypto_out->key_params = kInline;
crypto_out->key_params += key;
return true;
}
bool AddCryptoParams(const std::string& crypto_suite,
CryptoParamsVec* cryptos_out) {
int size = static_cast<int>(cryptos_out->size());
cryptos_out->resize(size + 1);
return CreateCryptoParams(size, crypto_suite, &cryptos_out->at(size));
}
void AddMediaCryptos(const CryptoParamsVec& cryptos,
MediaContentDescription* media) {
for (const CryptoParams& crypto : cryptos) {
media->AddCrypto(crypto);
}
}
bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites,
MediaContentDescription* media) {
CryptoParamsVec cryptos;
for (const std::string& crypto_suite : crypto_suites) {
if (!AddCryptoParams(crypto_suite, &cryptos)) {
return false;
}
}
AddMediaCryptos(cryptos, media);
return true;
}
const CryptoParamsVec* GetCryptos(const ContentInfo* content) {
if (!content || !content->media_description()) {
return nullptr;
}
return &content->media_description()->cryptos();
}
bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
const CryptoParams& crypto,
CryptoParams* crypto_out) {
auto it = absl::c_find_if(
cryptos, [&crypto](const CryptoParams& c) { return crypto.Matches(c); });
if (it == cryptos.end()) {
return false;
}
*crypto_out = *it;
return true;
}
// For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the
// low overhead.
void GetSupportedAudioSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
if (crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher) {
crypto_suites->push_back(rtc::kSrtpAes128CmSha1_32);
}
crypto_suites->push_back(rtc::kSrtpAes128CmSha1_80);
if (crypto_options.srtp.enable_gcm_crypto_suites) {
crypto_suites->push_back(rtc::kSrtpAeadAes256Gcm);
crypto_suites->push_back(rtc::kSrtpAeadAes128Gcm);
}
}
void GetSupportedAudioSdesCryptoSuiteNames(
const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedSdesCryptoSuiteNames(GetSupportedAudioSdesCryptoSuites,
crypto_options, crypto_suite_names);
}
void GetSupportedVideoSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
crypto_suites->push_back(rtc::kSrtpAes128CmSha1_80);
if (crypto_options.srtp.enable_gcm_crypto_suites) {
crypto_suites->push_back(rtc::kSrtpAeadAes256Gcm);
crypto_suites->push_back(rtc::kSrtpAeadAes128Gcm);
}
}
void GetSupportedVideoSdesCryptoSuiteNames(
const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites,
crypto_options, crypto_suite_names);
}
// Support any GCM cipher (if enabled through options). For video support only
// 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated (if enabled) unless
// bundle is enabled because it is low overhead.
// Pick the crypto in the list that is supported.
bool SelectCrypto(const MediaContentDescription* offer,
bool bundle,
const webrtc::CryptoOptions& crypto_options,
CryptoParams* crypto_out) {
bool audio = offer->type() == MEDIA_TYPE_AUDIO;
const CryptoParamsVec& cryptos = offer->cryptos();
for (const CryptoParams& crypto : cryptos) {
if ((crypto_options.srtp.enable_gcm_crypto_suites &&
rtc::IsGcmCryptoSuiteName(crypto.crypto_suite)) ||
rtc::kCsAesCm128HmacSha1_80 == crypto.crypto_suite ||
(rtc::kCsAesCm128HmacSha1_32 == crypto.crypto_suite && audio &&
!bundle && crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher)) {
return CreateCryptoParams(crypto.tag, crypto.crypto_suite, crypto_out);
}
}
return false;
}
// Finds all StreamParams of all media types and attach them to stream_params.
StreamParamsVec GetCurrentStreamParams(
const std::vector<const ContentInfo*>& active_local_contents) {
StreamParamsVec stream_params;
for (const ContentInfo* content : active_local_contents) {
for (const StreamParams& params : content->media_description()->streams()) {
stream_params.push_back(params);
}
}
return stream_params;
}
StreamParams CreateStreamParamsForNewSenderWithSsrcs(
const SenderOptions& sender,
const std::string& rtcp_cname,
bool include_rtx_streams,
bool include_flexfec_stream,
UniqueRandomIdGenerator* ssrc_generator,
const webrtc::FieldTrialsView& field_trials) {
StreamParams result;
result.id = sender.track_id;
// TODO(brandtr): Update when we support multistream protection.
if (include_flexfec_stream && sender.num_sim_layers > 1) {
include_flexfec_stream = false;
RTC_LOG(LS_WARNING)
<< "Our FlexFEC implementation only supports protecting "
"a single media streams. This session has multiple "
"media streams however, so no FlexFEC SSRC will be generated.";
}
if (include_flexfec_stream && !field_trials.IsEnabled("WebRTC-FlexFEC-03")) {
include_flexfec_stream = false;
RTC_LOG(LS_WARNING)
<< "WebRTC-FlexFEC trial is not enabled, not sending FlexFEC";
}
result.GenerateSsrcs(sender.num_sim_layers, include_rtx_streams,
include_flexfec_stream, ssrc_generator);
result.cname = rtcp_cname;
result.set_stream_ids(sender.stream_ids);
return result;
}
bool ValidateSimulcastLayers(const std::vector<RidDescription>& rids,
const SimulcastLayerList& simulcast_layers) {
return absl::c_all_of(
simulcast_layers.GetAllLayers(), [&rids](const SimulcastLayer& layer) {
return absl::c_any_of(rids, [&layer](const RidDescription& rid) {
return rid.rid == layer.rid;
});
});
}
StreamParams CreateStreamParamsForNewSenderWithRids(
const SenderOptions& sender,
const std::string& rtcp_cname) {
RTC_DCHECK(!sender.rids.empty());
RTC_DCHECK_EQ(sender.num_sim_layers, 0)
<< "RIDs are the compliant way to indicate simulcast.";
RTC_DCHECK(ValidateSimulcastLayers(sender.rids, sender.simulcast_layers));
StreamParams result;
result.id = sender.track_id;
result.cname = rtcp_cname;
result.set_stream_ids(sender.stream_ids);
// More than one rid should be signaled.
if (sender.rids.size() > 1) {
result.set_rids(sender.rids);
}
return result;
}
// Adds SimulcastDescription if indicated by the media description options.
// MediaContentDescription should already be set up with the send rids.
void AddSimulcastToMediaDescription(
const MediaDescriptionOptions& media_description_options,
MediaContentDescription* description) {
RTC_DCHECK(description);
// Check if we are using RIDs in this scenario.
if (absl::c_all_of(description->streams(), [](const StreamParams& params) {
return !params.has_rids();
})) {
return;
}
RTC_DCHECK_EQ(1, description->streams().size())
<< "RIDs are only supported in Unified Plan semantics.";
RTC_DCHECK_EQ(1, media_description_options.sender_options.size());
RTC_DCHECK(description->type() == MediaType::MEDIA_TYPE_AUDIO ||
description->type() == MediaType::MEDIA_TYPE_VIDEO);
// One RID or less indicates that simulcast is not needed.
if (description->streams()[0].rids().size() <= 1) {
return;
}
// Only negotiate the send layers.
SimulcastDescription simulcast;
simulcast.send_layers() =
media_description_options.sender_options[0].simulcast_layers;
description->set_simulcast_description(simulcast);
}
// Adds a StreamParams for each SenderOptions in `sender_options` to
// content_description.
// `current_params` - All currently known StreamParams of any media type.
bool AddStreamParams(const std::vector<SenderOptions>& sender_options,
const std::string& rtcp_cname,
UniqueRandomIdGenerator* ssrc_generator,
StreamParamsVec* current_streams,
MediaContentDescription* content_description,
const webrtc::FieldTrialsView& field_trials) {
// SCTP streams are not negotiated using SDP/ContentDescriptions.
if (IsSctpProtocol(content_description->protocol())) {
return true;
}
const bool include_rtx_streams =
ContainsRtxCodec(content_description->codecs());
const bool include_flexfec_stream =
ContainsFlexfecCodec(content_description->codecs());
for (const SenderOptions& sender : sender_options) {
StreamParams* param = GetStreamByIds(*current_streams, sender.track_id);
if (!param) {
// This is a new sender.
StreamParams stream_param =
sender.rids.empty()
?
// Signal SSRCs and legacy simulcast (if requested).
CreateStreamParamsForNewSenderWithSsrcs(
sender, rtcp_cname, include_rtx_streams,
include_flexfec_stream, ssrc_generator, field_trials)
:
// Signal RIDs and spec-compliant simulcast (if requested).
CreateStreamParamsForNewSenderWithRids(sender, rtcp_cname);
content_description->AddStream(stream_param);
// Store the new StreamParams in current_streams.
// This is necessary so that we can use the CNAME for other media types.
current_streams->push_back(stream_param);
} else {
// Use existing generated SSRCs/groups, but update the sync_label if
// necessary. This may be needed if a MediaStreamTrack was moved from one
// MediaStream to another.
param->set_stream_ids(sender.stream_ids);
content_description->AddStream(*param);
}
}
return true;
}
// Updates the transport infos of the `sdesc` according to the given
// `bundle_group`. The transport infos of the content names within the
// `bundle_group` should be updated to use the ufrag, pwd and DTLS role of the
// first content within the `bundle_group`.
bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
SessionDescription* sdesc) {
// The bundle should not be empty.
if (!sdesc || !bundle_group.FirstContentName()) {
return false;
}
// We should definitely have a transport for the first content.
const std::string& selected_content_name = *bundle_group.FirstContentName();
const TransportInfo* selected_transport_info =
sdesc->GetTransportInfoByName(selected_content_name);
if (!selected_transport_info) {
return false;
}
// Set the other contents to use the same ICE credentials.
const std::string& selected_ufrag =
selected_transport_info->description.ice_ufrag;
const std::string& selected_pwd =
selected_transport_info->description.ice_pwd;
ConnectionRole selected_connection_role =
selected_transport_info->description.connection_role;
for (TransportInfo& transport_info : sdesc->transport_infos()) {
if (bundle_group.HasContentName(transport_info.content_name) &&
transport_info.content_name != selected_content_name) {
transport_info.description.ice_ufrag = selected_ufrag;
transport_info.description.ice_pwd = selected_pwd;
transport_info.description.connection_role = selected_connection_role;
}
}
return true;
}
// Gets the CryptoParamsVec of the given `content_name` from `sdesc`, and
// sets it to `cryptos`.
bool GetCryptosByName(const SessionDescription* sdesc,
const std::string& content_name,
CryptoParamsVec* cryptos) {
if (!sdesc || !cryptos) {
return false;
}
const ContentInfo* content = sdesc->GetContentByName(content_name);
if (!content || !content->media_description()) {
return false;
}
*cryptos = content->media_description()->cryptos();
return true;
}
// Prunes the `target_cryptos` by removing the crypto params (crypto_suite)
// which are not available in `filter`.
void PruneCryptos(const CryptoParamsVec& filter,
CryptoParamsVec* target_cryptos) {
if (!target_cryptos) {
return;
}
target_cryptos->erase(
std::remove_if(target_cryptos->begin(), target_cryptos->end(),
// Returns true if the `crypto`'s crypto_suite is not
// found in `filter`.
[&filter](const CryptoParams& crypto) {
for (const CryptoParams& entry : filter) {
if (entry.crypto_suite == crypto.crypto_suite)
return false;
}
return true;
}),
target_cryptos->end());
}
bool IsRtpContent(SessionDescription* sdesc, const std::string& content_name) {
bool is_rtp = false;
ContentInfo* content = sdesc->GetContentByName(content_name);
if (content && content->media_description()) {
is_rtp = IsRtpProtocol(content->media_description()->protocol());
}
return is_rtp;
}
// Updates the crypto parameters of the `sdesc` according to the given
// `bundle_group`. The crypto parameters of all the contents within the
// `bundle_group` should be updated to use the common subset of the
// available cryptos.
bool UpdateCryptoParamsForBundle(const ContentGroup& bundle_group,
SessionDescription* sdesc) {
// The bundle should not be empty.
if (!sdesc || !bundle_group.FirstContentName()) {
return false;
}
bool common_cryptos_needed = false;
// Get the common cryptos.
const ContentNames& content_names = bundle_group.content_names();
CryptoParamsVec common_cryptos;
bool first = true;
for (const std::string& content_name : content_names) {
if (!IsRtpContent(sdesc, content_name)) {
continue;
}
// The common cryptos are needed if any of the content does not have DTLS
// enabled.
if (!sdesc->GetTransportInfoByName(content_name)->description.secure()) {
common_cryptos_needed = true;
}
if (first) {
first = false;
// Initial the common_cryptos with the first content in the bundle group.
if (!GetCryptosByName(sdesc, content_name, &common_cryptos)) {
return false;
}
if (common_cryptos.empty()) {
// If there's no crypto params, we should just return.
return true;
}
} else {
CryptoParamsVec cryptos;
if (!GetCryptosByName(sdesc, content_name, &cryptos)) {
return false;
}
PruneCryptos(cryptos, &common_cryptos);
}
}
if (common_cryptos.empty() && common_cryptos_needed) {
return false;
}
// Update to use the common cryptos.
for (const std::string& content_name : content_names) {
if (!IsRtpContent(sdesc, content_name)) {
continue;
}
ContentInfo* content = sdesc->GetContentByName(content_name);
if (IsMediaContent(content)) {
MediaContentDescription* media_desc = content->media_description();
if (!media_desc) {
return false;
}
media_desc->set_cryptos(common_cryptos);
}
}
return true;
}
std::vector<const ContentInfo*> GetActiveContents(
const SessionDescription& description,
const MediaSessionOptions& session_options) {
std::vector<const ContentInfo*> active_contents;
for (size_t i = 0; i < description.contents().size(); ++i) {
RTC_DCHECK_LT(i, session_options.media_description_options.size());
const ContentInfo& content = description.contents()[i];
const MediaDescriptionOptions& media_options =
session_options.media_description_options[i];
if (!content.rejected && !media_options.stopped &&
content.name == media_options.mid) {
active_contents.push_back(&content);
}
}
return active_contents;
}
// Create a media content to be offered for the given `sender_options`,
// according to the given options.rtcp_mux, session_options.is_muc, codecs,
// secure_transport, crypto, and current_streams. If we don't currently have
// crypto (in current_cryptos) and it is enabled (in secure_policy), crypto is
// created (according to crypto_suites). The created content is added to the
// offer.
RTCError CreateContentOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const SecurePolicy& secure_policy,
const CryptoParamsVec* current_cryptos,
const std::vector<std::string>& crypto_suites,
const RtpHeaderExtensions& rtp_extensions,
UniqueRandomIdGenerator* ssrc_generator,
StreamParamsVec* current_streams,
MediaContentDescription* offer) {
offer->set_rtcp_mux(session_options.rtcp_mux_enabled);
if (offer->type() == cricket::MEDIA_TYPE_VIDEO) {
offer->set_rtcp_reduced_size(true);
}
// Build the vector of header extensions with directions for this
// media_description's options.
RtpHeaderExtensions extensions;
for (auto extension_with_id : rtp_extensions) {
for (const auto& extension : media_description_options.header_extensions) {
if (extension_with_id.uri == extension.uri) {
// TODO(crbug.com/1051821): Configure the extension direction from
// the information in the media_description_options extension
// capability.
if (extension.direction != RtpTransceiverDirection::kStopped) {
extensions.push_back(extension_with_id);
}
}
}
}
offer->set_rtp_header_extensions(extensions);
AddSimulcastToMediaDescription(media_description_options, offer);
if (secure_policy != SEC_DISABLED) {
if (current_cryptos) {
AddMediaCryptos(*current_cryptos, offer);
}
if (offer->cryptos().empty()) {
if (!CreateMediaCryptos(crypto_suites, offer)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create crypto parameters");
}
}
}
if (secure_policy == SEC_REQUIRED && offer->cryptos().empty()) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create crypto parameters");
}
return RTCError::OK();
}
RTCError CreateMediaContentOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const std::vector<Codec>& codecs,
const SecurePolicy& secure_policy,
const CryptoParamsVec* current_cryptos,
const std::vector<std::string>& crypto_suites,
const RtpHeaderExtensions& rtp_extensions,
UniqueRandomIdGenerator* ssrc_generator,
StreamParamsVec* current_streams,
MediaContentDescription* offer,
const webrtc::FieldTrialsView& field_trials) {
offer->AddCodecs(codecs);
if (!AddStreamParams(media_description_options.sender_options,
session_options.rtcp_cname, ssrc_generator,
current_streams, offer, field_trials)) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to add stream parameters");
}
return CreateContentOffer(media_description_options, session_options,
secure_policy, current_cryptos, crypto_suites,
rtp_extensions, ssrc_generator, current_streams,
offer);
}
bool ReferencedCodecsMatch(const std::vector<Codec>& codecs1,
const int codec1_id,
const std::vector<Codec>& codecs2,
const int codec2_id,
const webrtc::FieldTrialsView* field_trials) {
const Codec* codec1 = FindCodecById(codecs1, codec1_id);
const Codec* codec2 = FindCodecById(codecs2, codec2_id);
return codec1 != nullptr && codec2 != nullptr &&
codec1->Matches(*codec2, field_trials);
}
void NegotiatePacketization(const Codec& local_codec,
const Codec& remote_codec,
Codec* negotiated_codec) {
negotiated_codec->packetization =
(local_codec.packetization == remote_codec.packetization)
? local_codec.packetization
: absl::nullopt;
}
// Finds a codec in `codecs2` that matches `codec_to_match`, which is
// a member of `codecs1`. If `codec_to_match` is an RED or RTX codec, both
// the codecs themselves and their associated codecs must match.
absl::optional<Codec> FindMatchingCodec(
const std::vector<Codec>& codecs1,
const std::vector<Codec>& codecs2,
const Codec& codec_to_match,
const webrtc::FieldTrialsView* field_trials) {
// `codec_to_match` should be a member of `codecs1`, in order to look up
// RED/RTX codecs' associated codecs correctly. If not, that's a programming
// error.
RTC_DCHECK(absl::c_any_of(codecs1, [&codec_to_match](const Codec& codec) {
return &codec == &codec_to_match;
}));
for (const Codec& potential_match : codecs2) {
if (potential_match.Matches(codec_to_match, field_trials)) {
if (IsRtxCodec(codec_to_match)) {
int apt_value_1 = 0;
int apt_value_2 = 0;
if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType,
&apt_value_1) ||
!potential_match.GetParam(kCodecParamAssociatedPayloadType,
&apt_value_2)) {
RTC_LOG(LS_WARNING) << "RTX missing associated payload type.";
continue;
}
if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2, apt_value_2,
field_trials)) {
continue;
}
} else if (IsRedCodec(codec_to_match)) {
auto red_parameters_1 =
codec_to_match.params.find(kCodecParamNotInNameValueFormat);
auto red_parameters_2 =
potential_match.params.find(kCodecParamNotInNameValueFormat);
bool has_parameters_1 = red_parameters_1 != codec_to_match.params.end();
bool has_parameters_2 =
red_parameters_2 != potential_match.params.end();
if (has_parameters_1 && has_parameters_2) {
// Mixed reference codecs (i.e. 111/112) are not supported.
// Different levels of redundancy between offer and answer are
// since RED is considered to be declarative.
std::vector<absl::string_view> redundant_payloads_1 =
rtc::split(red_parameters_1->second, '/');
std::vector<absl::string_view> redundant_payloads_2 =
rtc::split(red_parameters_2->second, '/');
if (redundant_payloads_1.size() > 0 &&
redundant_payloads_2.size() > 0) {
bool consistent = true;
for (size_t i = 1; i < redundant_payloads_1.size(); i++) {
if (redundant_payloads_1[i] != redundant_payloads_1[0]) {
consistent = false;
break;
}
}
for (size_t i = 1; i < redundant_payloads_2.size(); i++) {
if (redundant_payloads_2[i] != redundant_payloads_2[0]) {
consistent = false;
break;
}
}
if (!consistent) {
continue;
}
int red_value_1;
int red_value_2;
if (rtc::FromString(redundant_payloads_1[0], &red_value_1) &&
rtc::FromString(redundant_payloads_2[0], &red_value_2)) {
if (!ReferencedCodecsMatch(codecs1, red_value_1, codecs2,
red_value_2, field_trials)) {
continue;
}
}
}
} else if (has_parameters_1 != has_parameters_2) {
continue;
}
}
return potential_match;
}
}
return absl::nullopt;
}
void NegotiateCodecs(const std::vector<Codec>& local_codecs,
const std::vector<Codec>& offered_codecs,
std::vector<Codec>* negotiated_codecs,
bool keep_offer_order,
const webrtc::FieldTrialsView* field_trials) {
for (const Codec& ours : local_codecs) {
absl::optional<Codec> theirs =
FindMatchingCodec(local_codecs, offered_codecs, ours, field_trials);
// Note that we intentionally only find one matching codec for each of our
// local codecs, in case the remote offer contains duplicate codecs.
if (theirs) {
Codec negotiated = ours;
NegotiatePacketization(ours, *theirs, &negotiated);
negotiated.IntersectFeedbackParams(*theirs);
if (IsRtxCodec(negotiated)) {
const auto apt_it =
theirs->params.find(kCodecParamAssociatedPayloadType);
// FindMatchingCodec shouldn't return something with no apt value.
RTC_DCHECK(apt_it != theirs->params.end());
negotiated.SetParam(kCodecParamAssociatedPayloadType, apt_it->second);
// We support parsing the declarative rtx-time parameter.
const auto rtx_time_it = theirs->params.find(kCodecParamRtxTime);
if (rtx_time_it != theirs->params.end()) {
negotiated.SetParam(kCodecParamRtxTime, rtx_time_it->second);
}
} else if (IsRedCodec(negotiated)) {
const auto red_it =
theirs->params.find(kCodecParamNotInNameValueFormat);
if (red_it != theirs->params.end()) {
negotiated.SetParam(kCodecParamNotInNameValueFormat, red_it->second);
}
}
if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
webrtc::H264GenerateProfileLevelIdForAnswer(ours.params, theirs->params,
&negotiated.params);
}
negotiated.id = theirs->id;
negotiated.name = theirs->name;
negotiated_codecs->push_back(std::move(negotiated));
}
}
if (keep_offer_order) {
// RFC3264: Although the answerer MAY list the formats in their desired
// order of preference, it is RECOMMENDED that unless there is a
// specific reason, the answerer list formats in the same relative order
// they were present in the offer.
// This can be skipped when the transceiver has any codec preferences.
std::unordered_map<int, int> payload_type_preferences;
int preference = static_cast<int>(offered_codecs.size() + 1);
for (const Codec& codec : offered_codecs) {
payload_type_preferences[codec.id] = preference--;
}
absl::c_sort(*negotiated_codecs, [&payload_type_preferences](
const Codec& a, const Codec& b) {
return payload_type_preferences[a.id] > payload_type_preferences[b.id];
});
}
}
// Find the codec in `codec_list` that `rtx_codec` is associated with.
const Codec* GetAssociatedCodecForRtx(const std::vector<Codec>& codec_list,
const Codec& rtx_codec) {
std::string associated_pt_str;
if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_pt_str)) {
RTC_LOG(LS_WARNING) << "RTX codec " << rtx_codec.id
<< " is missing an associated payload type.";
return nullptr;
}
int associated_pt;
if (!rtc::FromString(associated_pt_str, &associated_pt)) {
RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
<< " of RTX codec " << rtx_codec.id
<< " to an integer.";
return nullptr;
}
// Find the associated codec for the RTX codec.
const Codec* associated_codec = FindCodecById(codec_list, associated_pt);
if (!associated_codec) {
RTC_LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
<< associated_pt << " for RTX codec " << rtx_codec.id
<< ".";
}
return associated_codec;
}
// Find the codec in `codec_list` that `red_codec` is associated with.
const Codec* GetAssociatedCodecForRed(const std::vector<Codec>& codec_list,
const Codec& red_codec) {
std::string fmtp;
if (!red_codec.GetParam(kCodecParamNotInNameValueFormat, &fmtp)) {
// Don't log for video/RED where this is normal.
if (red_codec.type == Codec::Type::kAudio) {
RTC_LOG(LS_WARNING) << "RED codec " << red_codec.id
<< " is missing an associated payload type.";
}
return nullptr;
}
std::vector<absl::string_view> redundant_payloads = rtc::split(fmtp, '/');
if (redundant_payloads.size() < 2) {
return nullptr;
}
absl::string_view associated_pt_str = redundant_payloads[0];
int associated_pt;
if (!rtc::FromString(associated_pt_str, &associated_pt)) {
RTC_LOG(LS_WARNING) << "Couldn't convert first payload type "
<< associated_pt_str << " of RED codec " << red_codec.id
<< " to an integer.";
return nullptr;
}
// Find the associated codec for the RED codec.
const Codec* associated_codec = FindCodecById(codec_list, associated_pt);
if (!associated_codec) {
RTC_LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
<< associated_pt << " for RED codec " << red_codec.id
<< ".";
}
return associated_codec;
}
// Adds all codecs from `reference_codecs` to `offered_codecs` that don't
// already exist in `offered_codecs` and ensure the payload types don't
// collide.
void MergeCodecs(const std::vector<Codec>& reference_codecs,
std::vector<Codec>* offered_codecs,
UsedPayloadTypes* used_pltypes,
const webrtc::FieldTrialsView* field_trials) {
// Add all new codecs that are not RTX/RED codecs.
// The two-pass splitting of the loops means preferring payload types
// of actual codecs with respect to collisions.
for (const Codec& reference_codec : reference_codecs) {
if (!IsRtxCodec(reference_codec) && !IsRedCodec(reference_codec) &&
!FindMatchingCodec(reference_codecs, *offered_codecs, reference_codec,
field_trials)) {
Codec codec = reference_codec;
used_pltypes->FindAndSetIdUsed(&codec);
offered_codecs->push_back(codec);
}
}
// Add all new RTX or RED codecs.
for (const Codec& reference_codec : reference_codecs) {
if (IsRtxCodec(reference_codec) &&
!FindMatchingCodec(reference_codecs, *offered_codecs, reference_codec,
field_trials)) {
Codec rtx_codec = reference_codec;
const Codec* associated_codec =
GetAssociatedCodecForRtx(reference_codecs, rtx_codec);
if (!associated_codec) {
continue;
}
// Find a codec in the offered list that matches the reference codec.
// Its payload type may be different than the reference codec.
absl::optional<Codec> matching_codec = FindMatchingCodec(
reference_codecs, *offered_codecs, *associated_codec, field_trials);
if (!matching_codec) {
RTC_LOG(LS_WARNING)
<< "Couldn't find matching " << associated_codec->name << " codec.";
continue;
}
rtx_codec.params[kCodecParamAssociatedPayloadType] =
rtc::ToString(matching_codec->id);
used_pltypes->FindAndSetIdUsed(&rtx_codec);
offered_codecs->push_back(rtx_codec);
} else if (IsRedCodec(reference_codec) &&
!FindMatchingCodec(reference_codecs, *offered_codecs,
reference_codec, field_trials)) {
Codec red_codec = reference_codec;
const Codec* associated_codec =
GetAssociatedCodecForRed(reference_codecs, red_codec);
if (associated_codec) {
absl::optional<Codec> matching_codec = FindMatchingCodec(
reference_codecs, *offered_codecs, *associated_codec, field_trials);
if (!matching_codec) {
RTC_LOG(LS_WARNING) << "Couldn't find matching "
<< associated_codec->name << " codec.";
continue;
}
red_codec.params[kCodecParamNotInNameValueFormat] =
rtc::ToString(matching_codec->id) + "/" +
rtc::ToString(matching_codec->id);
}
used_pltypes->FindAndSetIdUsed(&red_codec);
offered_codecs->push_back(red_codec);
}
}
}
// `codecs` is a full list of codecs with correct payload type mappings, which
// don't conflict with mappings of the other media type; `supported_codecs` is
// a list filtered for the media section`s direction but with default payload
// types.
std::vector<Codec> MatchCodecPreference(
const std::vector<webrtc::RtpCodecCapability>& codec_preferences,
const std::vector<Codec>& codecs,
const std::vector<Codec>& supported_codecs,
const webrtc::FieldTrialsView* field_trials) {
std::vector<Codec> filtered_codecs;
bool want_rtx = false;
bool want_red = false;
for (const auto& codec_preference : codec_preferences) {
if (IsRtxCodec(codec_preference)) {
want_rtx = true;
} else if (IsRedCodec(codec_preference)) {
want_red = true;
}
}
bool red_was_added = false;
for (const auto& codec_preference : codec_preferences) {
auto found_codec = absl::c_find_if(
supported_codecs, [&codec_preference](const Codec& codec) {
webrtc::RtpCodecParameters codec_parameters =
codec.ToCodecParameters();
return codec_parameters.name == codec_preference.name &&
codec_parameters.kind == codec_preference.kind &&
codec_parameters.num_channels ==
codec_preference.num_channels &&
codec_parameters.clock_rate == codec_preference.clock_rate &&
codec_parameters.parameters == codec_preference.parameters;
});
if (found_codec != supported_codecs.end()) {
absl::optional<Codec> found_codec_with_correct_pt = FindMatchingCodec(
supported_codecs, codecs, *found_codec, field_trials);
if (found_codec_with_correct_pt) {
// RED may already have been added if its primary codec is before RED
// in the codec list.
bool is_red_codec = IsRedCodec(*found_codec_with_correct_pt);
if (!is_red_codec || !red_was_added) {
filtered_codecs.push_back(*found_codec_with_correct_pt);
red_was_added = is_red_codec ? true : red_was_added;
}
std::string id = rtc::ToString(found_codec_with_correct_pt->id);
// Search for the matching rtx or red codec.
if (want_red || want_rtx) {
for (const auto& codec : codecs) {
if (IsRtxCodec(codec)) {
const auto apt =
codec.params.find(cricket::kCodecParamAssociatedPayloadType);
if (apt != codec.params.end() && apt->second == id) {
filtered_codecs.push_back(codec);
break;
}
} else if (IsRedCodec(codec)) {
// For RED, do not insert the codec again if it was already
// inserted. audio/red for opus gets enabled by having RED before
// the primary codec.
const auto fmtp =
codec.params.find(cricket::kCodecParamNotInNameValueFormat);
if (fmtp != codec.params.end()) {
std::vector<absl::string_view> redundant_payloads =
rtc::split(fmtp->second, '/');
if (!redundant_payloads.empty() &&
redundant_payloads[0] == id) {
if (!red_was_added) {
filtered_codecs.push_back(codec);
red_was_added = true;
}
break;
}
}
}
}
}
}
}
}
return filtered_codecs;
}
// Compute the union of `codecs1` and `codecs2`.
std::vector<Codec> ComputeCodecsUnion(
const std::vector<Codec>& codecs1,
const std::vector<Codec>& codecs2,
const webrtc::FieldTrialsView* field_trials) {
std::vector<Codec> all_codecs;
UsedPayloadTypes used_payload_types;
for (const Codec& codec : codecs1) {
Codec codec_mutable = codec;
used_payload_types.FindAndSetIdUsed(&codec_mutable);
all_codecs.push_back(codec_mutable);
}
// Use MergeCodecs to merge the second half of our list as it already checks
// and fixes problems with duplicate payload types.
MergeCodecs(codecs2, &all_codecs, &used_payload_types, field_trials);
return all_codecs;
}
// Adds all extensions from `reference_extensions` to `offered_extensions` that
// don't already exist in `offered_extensions` and ensure the IDs don't
// collide. If an extension is added, it's also added to `regular_extensions` or
// `encrypted_extensions`, and if the extension is in `regular_extensions` or
// `encrypted_extensions`, its ID is marked as used in `used_ids`.
// `offered_extensions` is for either audio or video while `regular_extensions`
// and `encrypted_extensions` are used for both audio and video. There could be
// overlap between audio extensions and video extensions.
void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
RtpHeaderExtensions* offered_extensions,
RtpHeaderExtensions* regular_extensions,
RtpHeaderExtensions* encrypted_extensions,
UsedRtpHeaderExtensionIds* used_ids) {
for (auto reference_extension : reference_extensions) {
if (!webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
*offered_extensions, reference_extension.uri,
reference_extension.encrypt)) {
if (reference_extension.encrypt) {
const webrtc::RtpExtension* existing =
webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
*encrypted_extensions, reference_extension.uri,
reference_extension.encrypt);
if (existing) {
offered_extensions->push_back(*existing);
} else {
used_ids->FindAndSetIdUsed(&reference_extension);
encrypted_extensions->push_back(reference_extension);
offered_extensions->push_back(reference_extension);
}
} else {
const webrtc::RtpExtension* existing =
webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
*regular_extensions, reference_extension.uri,
reference_extension.encrypt);
if (existing) {
offered_extensions->push_back(*existing);
} else {
used_ids->FindAndSetIdUsed(&reference_extension);
regular_extensions->push_back(reference_extension);
offered_extensions->push_back(reference_extension);
}
}
}
}
}
void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* offered_extensions,
RtpHeaderExtensions* encrypted_extensions,
UsedRtpHeaderExtensionIds* used_ids) {
RtpHeaderExtensions encrypted_extensions_to_add;
for (const auto& extension : *offered_extensions) {
// Skip existing encrypted offered extension
if (extension.encrypt) {
continue;
}
// Skip if we cannot encrypt the extension
if (!webrtc::RtpExtension::IsEncryptionSupported(extension.uri)) {
continue;
}
// Skip if an encrypted extension with that URI already exists in the
// offered extensions.
const bool have_encrypted_extension =
webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
*offered_extensions, extension.uri, true);
if (have_encrypted_extension) {
continue;
}
// Determine if a shared encrypted extension with that URI already exists.
const webrtc::RtpExtension* shared_encrypted_extension =
webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
*encrypted_extensions, extension.uri, true);
if (shared_encrypted_extension) {
// Re-use the shared encrypted extension
encrypted_extensions_to_add.push_back(*shared_encrypted_extension);
continue;
}
// None exists. Create a new shared encrypted extension from the
// non-encrypted one.
webrtc::RtpExtension new_encrypted_extension(extension);
new_encrypted_extension.encrypt = true;
used_ids->FindAndSetIdUsed(&new_encrypted_extension);
encrypted_extensions->push_back(new_encrypted_extension);
encrypted_extensions_to_add.push_back(new_encrypted_extension);
}
// Append the additional encrypted extensions to be offered
offered_extensions->insert(offered_extensions->end(),
encrypted_extensions_to_add.begin(),
encrypted_extensions_to_add.end());
}
// Mostly identical to RtpExtension::FindHeaderExtensionByUri but discards any
// encrypted extensions that this implementation cannot encrypt.
const webrtc::RtpExtension* FindHeaderExtensionByUriDiscardUnsupported(
const std::vector<webrtc::RtpExtension>& extensions,
absl::string_view uri,
webrtc::RtpExtension::Filter filter) {
// Note: While it's technically possible to decrypt extensions that we don't
// encrypt, the symmetric API of libsrtp does not allow us to supply
// different IDs for encryption/decryption of header extensions depending on
// whether the packet is inbound or outbound. Thereby, we are limited to
// what we can send in encrypted form.
if (!webrtc::RtpExtension::IsEncryptionSupported(uri)) {
// If there's no encryption support and we only want encrypted extensions,
// there's no point in continuing the search here.
if (filter == webrtc::RtpExtension::kRequireEncryptedExtension) {
return nullptr;
}
// Instruct to only return non-encrypted extensions
filter = webrtc::RtpExtension::Filter::kDiscardEncryptedExtension;
}
return webrtc::RtpExtension::FindHeaderExtensionByUri(extensions, uri,
filter);
}
void NegotiateRtpHeaderExtensions(const RtpHeaderExtensions& local_extensions,
const RtpHeaderExtensions& offered_extensions,
webrtc::RtpExtension::Filter filter,
RtpHeaderExtensions* negotiated_extensions) {
// TransportSequenceNumberV2 is not offered by default. The special logic for
// the TransportSequenceNumber extensions works as follows:
// Offer Answer
// V1 V1 if in local_extensions.
// V1 and V2 V2 regardless of local_extensions.
// V2 V2 regardless of local_extensions.
const webrtc::RtpExtension* transport_sequence_number_v2_offer =
FindHeaderExtensionByUriDiscardUnsupported(
offered_extensions,
webrtc::RtpExtension::kTransportSequenceNumberV2Uri, filter);
bool frame_descriptor_in_local = false;
bool dependency_descriptor_in_local = false;
bool abs_capture_time_in_local = false;
for (const webrtc::RtpExtension& ours : local_extensions) {
if (ours.uri == webrtc::RtpExtension::kGenericFrameDescriptorUri00)
frame_descriptor_in_local = true;
else if (ours.uri == webrtc::RtpExtension::kDependencyDescriptorUri)
dependency_descriptor_in_local = true;
else if (ours.uri == webrtc::RtpExtension::kAbsoluteCaptureTimeUri)
abs_capture_time_in_local = true;
const webrtc::RtpExtension* theirs =
FindHeaderExtensionByUriDiscardUnsupported(offered_extensions, ours.uri,
filter);
if (theirs) {
if (transport_sequence_number_v2_offer &&
ours.uri == webrtc::RtpExtension::kTransportSequenceNumberUri) {
// Don't respond to
// http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
// if we get an offer including
// http://www.webrtc.org/experiments/rtp-hdrext/transport-wide-cc-02
continue;
} else {
// We respond with their RTP header extension id.
negotiated_extensions->push_back(*theirs);
}
}
}
if (transport_sequence_number_v2_offer) {
// Respond that we support kTransportSequenceNumberV2Uri.
negotiated_extensions->push_back(*transport_sequence_number_v2_offer);
}
// Frame descriptors support. If the extension is not present locally, but is
// in the offer, we add it to the list.
if (!dependency_descriptor_in_local) {
const webrtc::RtpExtension* theirs =
FindHeaderExtensionByUriDiscardUnsupported(
offered_extensions, webrtc::RtpExtension::kDependencyDescriptorUri,
filter);
if (theirs) {
negotiated_extensions->push_back(*theirs);
}
}
if (!frame_descriptor_in_local) {
const webrtc::RtpExtension* theirs =
FindHeaderExtensionByUriDiscardUnsupported(
offered_extensions,
webrtc::RtpExtension::kGenericFrameDescriptorUri00, filter);
if (theirs) {
negotiated_extensions->push_back(*theirs);
}
}
// Absolute capture time support. If the extension is not present locally, but
// is in the offer, we add it to the list.
if (!abs_capture_time_in_local) {
const webrtc::RtpExtension* theirs =
FindHeaderExtensionByUriDiscardUnsupported(
offered_extensions, webrtc::RtpExtension::kAbsoluteCaptureTimeUri,
filter);
if (theirs) {
negotiated_extensions->push_back(*theirs);
}
}
}
void StripCNCodecs(AudioCodecs* audio_codecs) {
audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
[](const AudioCodec& codec) {
return IsComfortNoiseCodec(codec);
}),
audio_codecs->end());
}
bool SetCodecsInAnswer(const MediaContentDescription* offer,
const std::vector<Codec>& local_codecs,
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
UniqueRandomIdGenerator* ssrc_generator,
StreamParamsVec* current_streams,
MediaContentDescription* answer,
const webrtc::FieldTrialsView& field_trials) {
RTC_DCHECK(offer->type() == MEDIA_TYPE_AUDIO ||
offer->type() == MEDIA_TYPE_VIDEO);
std::vector<Codec> negotiated_codecs;
NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs,
media_description_options.codec_preferences.empty(),
&field_trials);
answer->AddCodecs(negotiated_codecs);
answer->set_protocol(offer->protocol());
if (!AddStreamParams(media_description_options.sender_options,
session_options.rtcp_cname, ssrc_generator,
current_streams, answer, field_trials)) {
return false; // Something went seriously wrong.
}
return true;
}
// Create a media content to be answered for the given `sender_options`
// according to the given session_options.rtcp_mux, session_options.streams,
// codecs, crypto, and current_streams. If we don't currently have crypto (in
// current_cryptos) and it is enabled (in secure_policy), crypto is created
// (according to crypto_suites). The codecs, rtcp_mux, and crypto are all
// negotiated with the offer. If the negotiation fails, this method returns
// false. The created content is added to the offer.
bool CreateMediaContentAnswer(
const MediaContentDescription* offer,
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const SecurePolicy& sdes_policy,
const CryptoParamsVec* current_cryptos,
const RtpHeaderExtensions& local_rtp_extensions,
UniqueRandomIdGenerator* ssrc_generator,
bool enable_encrypted_rtp_header_extensions,
StreamParamsVec* current_streams,
bool bundle_enabled,
MediaContentDescription* answer) {
answer->set_extmap_allow_mixed_enum(offer->extmap_allow_mixed_enum());
const webrtc::RtpExtension::Filter extensions_filter =
enable_encrypted_rtp_header_extensions
? webrtc::RtpExtension::Filter::kPreferEncryptedExtension
: webrtc::RtpExtension::Filter::kDiscardEncryptedExtension;
// Filter local extensions by capabilities and direction.
RtpHeaderExtensions local_rtp_extensions_to_reply_with;
for (auto extension_with_id : local_rtp_extensions) {
for (const auto& extension : media_description_options.header_extensions) {
if (extension_with_id.uri == extension.uri) {
// TODO(crbug.com/1051821): Configure the extension direction from
// the information in the media_description_options extension
// capability. For now, do not include stopped extensions.
// See also crbug.com/webrtc/7477 about the general lack of direction.
if (extension.direction != RtpTransceiverDirection::kStopped) {
local_rtp_extensions_to_reply_with.push_back(extension_with_id);
}
}
}
}
RtpHeaderExtensions negotiated_rtp_extensions;
NegotiateRtpHeaderExtensions(local_rtp_extensions_to_reply_with,
offer->rtp_header_extensions(),
extensions_filter, &negotiated_rtp_extensions);
answer->set_rtp_header_extensions(negotiated_rtp_extensions);
answer->set_rtcp_mux(session_options.rtcp_mux_enabled && offer->rtcp_mux());
if (answer->type() == cricket::MEDIA_TYPE_VIDEO) {
answer->set_rtcp_reduced_size(offer->rtcp_reduced_size());
}
answer->set_remote_estimate(offer->remote_estimate());
if (sdes_policy != SEC_DISABLED) {
CryptoParams crypto;
if (SelectCrypto(offer, bundle_enabled, session_options.crypto_options,
&crypto)) {
if (current_cryptos) {
FindMatchingCrypto(*current_cryptos, crypto, &crypto);
}
answer->AddCrypto(crypto);
}
}
if (answer->cryptos().empty() && sdes_policy == SEC_REQUIRED) {
return false;
}
AddSimulcastToMediaDescription(media_description_options, answer);
answer->set_direction(NegotiateRtpTransceiverDirection(
offer->direction(), media_description_options.direction));
return true;
}
bool IsMediaProtocolSupported(MediaType type,
const std::string& protocol,
bool secure_transport) {
// Since not all applications serialize and deserialize the media protocol,
// we will have to accept `protocol` to be empty.
if (protocol.empty()) {
return true;
}
if (type == MEDIA_TYPE_DATA) {
// Check for SCTP
if (secure_transport) {
// Most likely scenarios first.
return IsDtlsSctp(protocol);
} else {
return IsPlainSctp(protocol);
}
}
// Allow for non-DTLS RTP protocol even when using DTLS because that's what
// JSEP specifies.
if (secure_transport) {
// Most likely scenarios first.
return IsDtlsRtp(protocol) || IsPlainRtp(protocol);
} else {
return IsPlainRtp(protocol);
}
}
void SetMediaProtocol(bool secure_transport, MediaContentDescription* desc) {
if (!desc->cryptos().empty())
desc->set_protocol(kMediaProtocolSavpf);
else if (secure_transport)
desc->set_protocol(kMediaProtocolDtlsSavpf);
else
desc->set_protocol(kMediaProtocolAvpf);
}
// Gets the TransportInfo of the given `content_name` from the
// `current_description`. If doesn't exist, returns a new one.
const TransportDescription* GetTransportDescription(
const std::string& content_name,
const SessionDescription* current_description) {
const TransportDescription* desc = NULL;
if (current_description) {
const TransportInfo* info =
current_description->GetTransportInfoByName(content_name);
if (info) {
desc = &info->description;
}
}
return desc;
}
// Gets the current DTLS state from the transport description.
bool IsDtlsActive(const ContentInfo* content,
const SessionDescription* current_description) {
if (!content) {
return false;
}
size_t msection_index = content - &current_description->contents()[0];
if (current_description->transport_infos().size() <= msection_index) {
return false;
}
return current_description->transport_infos()[msection_index]
.description.secure();
}
} // namespace
void MediaDescriptionOptions::AddAudioSender(
const std::string& track_id,
const std::vector<std::string>& stream_ids) {
RTC_DCHECK(type == MEDIA_TYPE_AUDIO);
AddSenderInternal(track_id, stream_ids, {}, SimulcastLayerList(), 1);
}
void MediaDescriptionOptions::AddVideoSender(
const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<RidDescription>& rids,
const SimulcastLayerList& simulcast_layers,
int num_sim_layers) {
RTC_DCHECK(type == MEDIA_TYPE_VIDEO);
RTC_DCHECK(rids.empty() || num_sim_layers == 0)
<< "RIDs are the compliant way to indicate simulcast.";
RTC_DCHECK(ValidateSimulcastLayers(rids, simulcast_layers));
AddSenderInternal(track_id, stream_ids, rids, simulcast_layers,
num_sim_layers);
}
void MediaDescriptionOptions::AddSenderInternal(
const std::string& track_id,
const std::vector<std::string>& stream_ids,
const std::vector<RidDescription>& rids,
const SimulcastLayerList& simulcast_layers,
int num_sim_layers) {
// TODO(steveanton): Support any number of stream ids.
RTC_CHECK(stream_ids.size() == 1U);
SenderOptions options;
options.track_id = track_id;
options.stream_ids = stream_ids;
options.simulcast_layers = simulcast_layers;
options.rids = rids;
options.num_sim_layers = num_sim_layers;
sender_options.push_back(options);
}
bool MediaSessionOptions::HasMediaDescription(MediaType type) const {
return absl::c_any_of(
media_description_options,
[type](const MediaDescriptionOptions& t) { return t.type == type; });
}
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
cricket::MediaEngineInterface* media_engine,
bool rtx_enabled,
rtc::UniqueRandomIdGenerator* ssrc_generator,
const TransportDescriptionFactory* transport_desc_factory)
: ssrc_generator_(ssrc_generator),
transport_desc_factory_(transport_desc_factory) {
RTC_CHECK(transport_desc_factory_);
if (media_engine) {
audio_send_codecs_ = media_engine->voice().send_codecs();
audio_recv_codecs_ = media_engine->voice().recv_codecs();
video_send_codecs_ = media_engine->video().send_codecs(rtx_enabled);
video_recv_codecs_ = media_engine->video().recv_codecs(rtx_enabled);
}
ComputeAudioCodecsIntersectionAndUnion();
ComputeVideoCodecsIntersectionAndUnion();
}
const AudioCodecs& MediaSessionDescriptionFactory::audio_sendrecv_codecs()
const {
return audio_sendrecv_codecs_;
}
const AudioCodecs& MediaSessionDescriptionFactory::audio_send_codecs() const {
return audio_send_codecs_;
}
const AudioCodecs& MediaSessionDescriptionFactory::audio_recv_codecs() const {
return audio_recv_codecs_;
}
void MediaSessionDescriptionFactory::set_audio_codecs(
const AudioCodecs& send_codecs,
const AudioCodecs& recv_codecs) {
audio_send_codecs_ = send_codecs;
audio_recv_codecs_ = recv_codecs;
ComputeAudioCodecsIntersectionAndUnion();
}
const VideoCodecs& MediaSessionDescriptionFactory::video_sendrecv_codecs()
const {
return video_sendrecv_codecs_;
}
const VideoCodecs& MediaSessionDescriptionFactory::video_send_codecs() const {
return video_send_codecs_;
}
const VideoCodecs& MediaSessionDescriptionFactory::video_recv_codecs() const {
return video_recv_codecs_;
}
void MediaSessionDescriptionFactory::set_video_codecs(
const VideoCodecs& send_codecs,
const VideoCodecs& recv_codecs) {
video_send_codecs_ = send_codecs;
video_recv_codecs_ = recv_codecs;
ComputeVideoCodecsIntersectionAndUnion();
}
RtpHeaderExtensions
MediaSessionDescriptionFactory::filtered_rtp_header_extensions(
RtpHeaderExtensions extensions) const {
if (!is_unified_plan_) {
// Remove extensions only supported with unified-plan.
extensions.erase(
std::remove_if(
extensions.begin(), extensions.end(),
[](const webrtc::RtpExtension& extension) {
return extension.uri == webrtc::RtpExtension::kMidUri ||
extension.uri == webrtc::RtpExtension::kRidUri ||
extension.uri == webrtc::RtpExtension::kRepairedRidUri;
}),
extensions.end());
}
return extensions;
}
webrtc::RTCErrorOr<std::unique_ptr<SessionDescription>>
MediaSessionDescriptionFactory::CreateOfferOrError(
const MediaSessionOptions& session_options,
const SessionDescription* current_description) const {
// Must have options for each existing section.
if (current_description) {
RTC_DCHECK_LE(current_description->contents().size(),
session_options.media_description_options.size());
}
IceCredentialsIterator ice_credentials(
session_options.pooled_ice_credentials);
std::vector<const ContentInfo*> current_active_contents;
if (current_description) {
current_active_contents =
GetActiveContents(*current_description, session_options);
}
StreamParamsVec current_streams =
GetCurrentStreamParams(current_active_contents);
AudioCodecs offer_audio_codecs;
VideoCodecs offer_video_codecs;
GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
&offer_video_codecs);
AudioVideoRtpHeaderExtensions extensions_with_ids =
GetOfferedRtpHeaderExtensionsWithIds(
current_active_contents, session_options.offer_extmap_allow_mixed,
session_options.media_description_options);
auto offer = std::make_unique<SessionDescription>();
// Iterate through the media description options, matching with existing media
// descriptions in `current_description`.
size_t msection_index = 0;
for (const MediaDescriptionOptions& media_description_options :
session_options.media_description_options) {
const ContentInfo* current_content = nullptr;
if (current_description &&
msection_index < current_description->contents().size()) {
current_content = &current_description->contents()[msection_index];
// Media type must match unless this media section is being recycled.
}
RTCError error;
switch (media_description_options.type) {
case MEDIA_TYPE_AUDIO:
error = AddAudioContentForOffer(
media_description_options, session_options, current_content,
current_description, extensions_with_ids.audio, offer_audio_codecs,
&current_streams, offer.get(), &ice_credentials);
break;
case MEDIA_TYPE_VIDEO:
error = AddVideoContentForOffer(
media_description_options, session_options, current_content,
current_description, extensions_with_ids.video, offer_video_codecs,
&current_streams, offer.get(), &ice_credentials);
break;
case MEDIA_TYPE_DATA:
error = AddDataContentForOffer(media_description_options,
session_options, current_content,
current_description, &current_streams,
offer.get(), &ice_credentials);
break;
case MEDIA_TYPE_UNSUPPORTED:
error = AddUnsupportedContentForOffer(
media_description_options, session_options, current_content,
current_description, offer.get(), &ice_credentials);
break;
default:
RTC_DCHECK_NOTREACHED();
}
if (!error.ok()) {
return error;
}
++msection_index;
}
// Bundle the contents together, if we've been asked to do so, and update any
// parameters that need to be tweaked for BUNDLE.
if (session_options.bundle_enabled) {
ContentGroup offer_bundle(GROUP_TYPE_BUNDLE);
for (const ContentInfo& content : offer->contents()) {
if (content.rejected) {
continue;
}
// TODO(deadbeef): There are conditions that make bundling two media
// descriptions together illegal. For example, they use the same payload
// type to represent different codecs, or same IDs for different header
// extensions. We need to detect this and not try to bundle those media
// descriptions together.
offer_bundle.AddContentName(content.name);
}
if (!offer_bundle.content_names().empty()) {
offer->AddGroup(offer_bundle);
if (!UpdateTransportInfoForBundle(offer_bundle, offer.get())) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"CreateOffer failed to UpdateTransportInfoForBundle");
}
if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"CreateOffer failed to UpdateCryptoParamsForBundle.");
}
}
}
// The following determines how to signal MSIDs to ensure compatibility with
// older endpoints (in particular, older Plan B endpoints).
if (is_unified_plan_) {
// Be conservative and signal using both a=msid and a=ssrc lines. Unified
// Plan answerers will look at a=msid and Plan B answerers will look at the
// a=ssrc MSID line.
offer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute);
} else {
// Plan B always signals MSID using a=ssrc lines.
offer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
}
offer->set_extmap_allow_mixed(session_options.offer_extmap_allow_mixed);
return offer;
}
webrtc::RTCErrorOr<std::unique_ptr<SessionDescription>>
MediaSessionDescriptionFactory::CreateAnswerOrError(
const SessionDescription* offer,
const MediaSessionOptions& session_options,
const SessionDescription* current_description) const {
if (!offer) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, "Called without offer.");
}
// Must have options for exactly as many sections as in the offer.
RTC_DCHECK_EQ(offer->contents().size(),
session_options.media_description_options.size());
IceCredentialsIterator ice_credentials(
session_options.pooled_ice_credentials);
std::vector<const ContentInfo*> current_active_contents;
if (current_description) {
current_active_contents =
GetActiveContents(*current_description, session_options);
}
StreamParamsVec current_streams =
GetCurrentStreamParams(current_active_contents);
// Get list of all possible codecs that respects existing payload type
// mappings and uses a single payload type space.
//
// Note that these lists may be further filtered for each m= section; this
// step is done just to establish the payload type mappings shared by all
// sections.
AudioCodecs answer_audio_codecs;
VideoCodecs answer_video_codecs;
GetCodecsForAnswer(current_active_contents, *offer, &answer_audio_codecs,
&answer_video_codecs);
auto answer = std::make_unique<SessionDescription>();
// If the offer supports BUNDLE, and we want to use it too, create a BUNDLE
// group in the answer with the appropriate content names.
std::vector<const ContentGroup*> offer_bundles =
offer->GetGroupsByName(GROUP_TYPE_BUNDLE);
// There are as many answer BUNDLE groups as offer BUNDLE groups (even if
// rejected, we respond with an empty group). `offer_bundles`,
// `answer_bundles` and `bundle_transports` share the same size and indices.
std::vector<ContentGroup> answer_bundles;
std::vector<std::unique_ptr<TransportInfo>> bundle_transports;
answer_bundles.reserve(offer_bundles.size());
bundle_transports.reserve(offer_bundles.size());
for (size_t i = 0; i < offer_bundles.size(); ++i) {
answer_bundles.emplace_back(GROUP_TYPE_BUNDLE);
bundle_transports.emplace_back(nullptr);
}
answer->set_extmap_allow_mixed(offer->extmap_allow_mixed());
// Iterate through the media description options, matching with existing
// media descriptions in `current_description`.
size_t msection_index = 0;
for (const MediaDescriptionOptions& media_description_options :
session_options.media_description_options) {
const ContentInfo* offer_content = &offer->contents()[msection_index];
// Media types and MIDs must match between the remote offer and the
// MediaDescriptionOptions.
RTC_DCHECK(
IsMediaContentOfType(offer_content, media_description_options.type));
RTC_DCHECK(media_description_options.mid == offer_content->name);
// Get the index of the BUNDLE group that this MID belongs to, if any.
absl::optional<size_t> bundle_index;
for (size_t i = 0; i < offer_bundles.size(); ++i) {
if (offer_bundles[i]->HasContentName(media_description_options.mid)) {
bundle_index = i;
break;
}
}
TransportInfo* bundle_transport =
bundle_index.has_value() ? bundle_transports[bundle_index.value()].get()
: nullptr;
const ContentInfo* current_content = nullptr;
if (current_description &&
msection_index < current_description->contents().size()) {
current_content = &current_description->contents()[msection_index];
}
RtpHeaderExtensions header_extensions = RtpHeaderExtensionsFromCapabilities(
UnstoppedRtpHeaderExtensionCapabilities(
media_description_options.header_extensions));
RTCError error;
switch (media_description_options.type) {
case MEDIA_TYPE_AUDIO:
error = AddAudioContentForAnswer(
media_description_options, session_options, offer_content, offer,
current_content, current_description, bundle_transport,
answer_audio_codecs, header_extensions, &current_streams,
answer.get(), &ice_credentials);
break;
case MEDIA_TYPE_VIDEO:
error = AddVideoContentForAnswer(
media_description_options, session_options, offer_content, offer,
current_content, current_description, bundle_transport,
answer_video_codecs, header_extensions, &current_streams,
answer.get(), &ice_credentials);
break;
case MEDIA_TYPE_DATA:
error = AddDataContentForAnswer(
media_description_options, session_options, offer_content, offer,
current_content, current_description, bundle_transport,
&current_streams, answer.get(), &ice_credentials);
break;
case MEDIA_TYPE_UNSUPPORTED:
error = AddUnsupportedContentForAnswer(
media_description_options, session_options, offer_content, offer,
current_content, current_description, bundle_transport,
answer.get(), &ice_credentials);
break;
default:
RTC_DCHECK_NOTREACHED();
}
if (!error.ok()) {
return error;
}
++msection_index;
// See if we can add the newly generated m= section to the BUNDLE group in
// the answer.
ContentInfo& added = answer->contents().back();
if (!added.rejected && session_options.bundle_enabled &&
bundle_index.has_value()) {
// The `bundle_index` is for `media_description_options.mid`.
RTC_DCHECK_EQ(media_description_options.mid, added.name);
answer_bundles[bundle_index.value()].AddContentName(added.name);
bundle_transports[bundle_index.value()].reset(
new TransportInfo(*answer->GetTransportInfoByName(added.name)));
}
}
// If BUNDLE group(s) were offered, put the same number of BUNDLE groups in
// the answer even if they're empty. RFC5888 says:
//
// A SIP entity that receives an offer that contains an "a=group" line
// with semantics that are understood MUST return an answer that
// contains an "a=group" line with the same semantics.
if (!offer_bundles.empty()) {
for (const ContentGroup& answer_bundle : answer_bundles) {
answer->AddGroup(answer_bundle);
if (answer_bundle.FirstContentName()) {
// Share the same ICE credentials and crypto params across all contents,
// as BUNDLE requires.
if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"CreateAnswer failed to UpdateTransportInfoForBundle.");
}
if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"CreateAnswer failed to UpdateCryptoParamsForBundle.");
}
}
}
}
// The following determines how to signal MSIDs to ensure compatibility with
// older endpoints (in particular, older Plan B endpoints).
if (is_unified_plan_) {
// Unified Plan needs to look at what the offer included to find the most
// compatible answer.
if (offer->msid_signaling() == 0) {
// We end up here in one of three cases:
// 1. An empty offer. We'll reply with an empty answer so it doesn't
// matter what we pick here.
// 2. A data channel only offer. We won't add any MSIDs to the answer so
// it also doesn't matter what we pick here.
// 3. Media that's either sendonly or inactive from the remote endpoint.
// We don't have any information to say whether the endpoint is Plan B
// or Unified Plan, so be conservative and send both.
answer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute);
} else if (offer->msid_signaling() ==
(cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute)) {
// If both a=msid and a=ssrc MSID signaling methods were used, we're
// probably talking to a Unified Plan endpoint so respond with just
// a=msid.
answer->set_msid_signaling(cricket::kMsidSignalingMediaSection);
} else {
// Otherwise, it's clear which method the offerer is using so repeat that
// back to them.
answer->set_msid_signaling(offer->msid_signaling());
}
} else {
// Plan B always signals MSID using a=ssrc lines.
answer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
}
return answer;
}
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer(
const RtpTransceiverDirection& direction) const {
switch (direction) {
// If stream is inactive - generate list as if sendrecv.
case RtpTransceiverDirection::kSendRecv:
case RtpTransceiverDirection::kStopped:
case RtpTransceiverDirection::kInactive:
return audio_sendrecv_codecs_;
case RtpTransceiverDirection::kSendOnly:
return audio_send_codecs_;
case RtpTransceiverDirection::kRecvOnly:
return audio_recv_codecs_;
}
RTC_CHECK_NOTREACHED();
}
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForAnswer(
const RtpTransceiverDirection& offer,
const RtpTransceiverDirection& answer) const {
switch (answer) {
// For inactive and sendrecv answers, generate lists as if we were to accept
// the offer's direction. See RFC 3264 Section 6.1.
case RtpTransceiverDirection::kSendRecv:
case RtpTransceiverDirection::kStopped:
case RtpTransceiverDirection::kInactive:
return GetAudioCodecsForOffer(
webrtc::RtpTransceiverDirectionReversed(offer));
case RtpTransceiverDirection::kSendOnly:
return audio_send_codecs_;
case RtpTransceiverDirection::kRecvOnly:
return audio_recv_codecs_;
}
RTC_CHECK_NOTREACHED();
}
const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForOffer(
const RtpTransceiverDirection& direction) const {
switch (direction) {
// If stream is inactive - generate list as if sendrecv.
case RtpTransceiverDirection::kSendRecv:
case RtpTransceiverDirection::kStopped:
case RtpTransceiverDirection::kInactive:
return video_sendrecv_codecs_;
case RtpTransceiverDirection::kSendOnly:
return video_send_codecs_;
case RtpTransceiverDirection::kRecvOnly:
return video_recv_codecs_;
}
RTC_CHECK_NOTREACHED();
}
const VideoCodecs& MediaSessionDescriptionFactory::GetVideoCodecsForAnswer(
const RtpTransceiverDirection& offer,
const RtpTransceiverDirection& answer) const {
switch (answer) {
// For inactive and sendrecv answers, generate lists as if we were to accept
// the offer's direction. See RFC 3264 Section 6.1.
case RtpTransceiverDirection::kSendRecv:
case RtpTransceiverDirection::kStopped:
case RtpTransceiverDirection::kInactive:
return GetVideoCodecsForOffer(
webrtc::RtpTransceiverDirectionReversed(offer));
case RtpTransceiverDirection::kSendOnly:
return video_send_codecs_;
case RtpTransceiverDirection::kRecvOnly:
return video_recv_codecs_;
}
RTC_CHECK_NOTREACHED();
}
void MergeCodecsFromDescription(
const std::vector<const ContentInfo*>& current_active_contents,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs,
UsedPayloadTypes* used_pltypes,
const webrtc::FieldTrialsView* field_trials) {
for (const ContentInfo* content : current_active_contents) {
if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
MergeCodecs(content->media_description()->codecs(), audio_codecs,
used_pltypes, field_trials);
} else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
MergeCodecs(content->media_description()->codecs(), video_codecs,
used_pltypes, field_trials);
}
}
}
// Getting codecs for an offer involves these steps:
//
// 1. Construct payload type -> codec mappings for current description.
// 2. Add any reference codecs that weren't already present
// 3. For each individual media description (m= section), filter codecs based
// on the directional attribute (happens in another method).
void MediaSessionDescriptionFactory::GetCodecsForOffer(
const std::vector<const ContentInfo*>& current_active_contents,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
// First - get all codecs from the current description if the media type
// is used. Add them to `used_pltypes` so the payload type is not reused if a
// new media type is added.
UsedPayloadTypes used_pltypes;
MergeCodecsFromDescription(current_active_contents, audio_codecs,
video_codecs, &used_pltypes, field_trials);
// Add our codecs that are not in the current description.
MergeCodecs(all_audio_codecs_, audio_codecs, &used_pltypes, field_trials);
MergeCodecs(all_video_codecs_, video_codecs, &used_pltypes, field_trials);
}
// Getting codecs for an answer involves these steps:
//
// 1. Construct payload type -> codec mappings for current description.
// 2. Add any codecs from the offer that weren't already present.
// 3. Add any remaining codecs that weren't already present.
// 4. For each individual media description (m= section), filter codecs based
// on the directional attribute (happens in another method).
void MediaSessionDescriptionFactory::GetCodecsForAnswer(
const std::vector<const ContentInfo*>& current_active_contents,
const SessionDescription& remote_offer,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
// First - get all codecs from the current description if the media type
// is used. Add them to `used_pltypes` so the payload type is not reused if a
// new media type is added.
UsedPayloadTypes used_pltypes;
MergeCodecsFromDescription(current_active_contents, audio_codecs,
video_codecs, &used_pltypes, field_trials);
// Second - filter out codecs that we don't support at all and should ignore.
AudioCodecs filtered_offered_audio_codecs;
VideoCodecs filtered_offered_video_codecs;
for (const ContentInfo& content : remote_offer.contents()) {
if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
std::vector<Codec> offered_codecs = content.media_description()->codecs();
for (const Codec& offered_audio_codec : offered_codecs) {
if (!FindMatchingCodec(offered_codecs, filtered_offered_audio_codecs,
offered_audio_codec, field_trials) &&
FindMatchingCodec(offered_codecs, all_audio_codecs_,
offered_audio_codec, field_trials)) {
filtered_offered_audio_codecs.push_back(offered_audio_codec);
}
}
} else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) {
std::vector<Codec> offered_codecs = content.media_description()->codecs();
for (const Codec& offered_video_codec : offered_codecs) {
if (!FindMatchingCodec(offered_codecs, filtered_offered_video_codecs,
offered_video_codec, field_trials) &&
FindMatchingCodec(offered_codecs, all_video_codecs_,
offered_video_codec, field_trials)) {
filtered_offered_video_codecs.push_back(offered_video_codec);
}
}
}
}
// Add codecs that are not in the current description but were in
// `remote_offer`.
MergeCodecs(filtered_offered_audio_codecs, audio_codecs, &used_pltypes,
field_trials);
MergeCodecs(filtered_offered_video_codecs, video_codecs, &used_pltypes,
field_trials);
}
MediaSessionDescriptionFactory::AudioVideoRtpHeaderExtensions
MediaSessionDescriptionFactory::GetOfferedRtpHeaderExtensionsWithIds(
const std::vector<const ContentInfo*>& current_active_contents,
bool extmap_allow_mixed,
const std::vector<MediaDescriptionOptions>& media_description_options)
const {
// All header extensions allocated from the same range to avoid potential
// issues when using BUNDLE.
// Strictly speaking the SDP attribute extmap_allow_mixed signals that the
// receiver supports an RTP stream where one- and two-byte RTP header
// extensions are mixed. For backwards compatibility reasons it's used in
// WebRTC to signal that two-byte RTP header extensions are supported.
UsedRtpHeaderExtensionIds used_ids(
extmap_allow_mixed ? UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed
: UsedRtpHeaderExtensionIds::IdDomain::kOneByteOnly);
RtpHeaderExtensions all_regular_extensions;
RtpHeaderExtensions all_encrypted_extensions;
AudioVideoRtpHeaderExtensions offered_extensions;
// First - get all extensions from the current description if the media type
// is used.
// Add them to `used_ids` so the local ids are not reused if a new media
// type is added.
for (const ContentInfo* content : current_active_contents) {
if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
MergeRtpHdrExts(content->media_description()->rtp_header_extensions(),
&offered_extensions.audio, &all_regular_extensions,
&all_encrypted_extensions, &used_ids);
} else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
MergeRtpHdrExts(content->media_description()->rtp_header_extensions(),
&offered_extensions.video, &all_regular_extensions,
&all_encrypted_extensions, &used_ids);
}
}
// Add all encountered header extensions in the media description options that
// are not in the current description.
for (const auto& entry : media_description_options) {
RtpHeaderExtensions filtered_extensions =
filtered_rtp_header_extensions(UnstoppedOrPresentRtpHeaderExtensions(
entry.header_extensions, all_regular_extensions,
all_encrypted_extensions));
if (entry.type == MEDIA_TYPE_AUDIO)
MergeRtpHdrExts(filtered_extensions, &offered_extensions.audio,
&all_regular_extensions, &all_encrypted_extensions,
&used_ids);
else if (entry.type == MEDIA_TYPE_VIDEO)
MergeRtpHdrExts(filtered_extensions, &offered_extensions.video,
&all_regular_extensions, &all_encrypted_extensions,
&used_ids);
}
// TODO(jbauch): Support adding encrypted header extensions to existing
// sessions.
if (enable_encrypted_rtp_header_extensions_ &&
current_active_contents.empty()) {
AddEncryptedVersionsOfHdrExts(&offered_extensions.audio,
&all_encrypted_extensions, &used_ids);
AddEncryptedVersionsOfHdrExts(&offered_extensions.video,
&all_encrypted_extensions, &used_ids);
}
return offered_extensions;
}
RTCError MediaSessionDescriptionFactory::AddTransportOffer(
const std::string& content_name,
const TransportOptions& transport_options,
const SessionDescription* current_desc,
SessionDescription* offer_desc,
IceCredentialsIterator* ice_credentials) const {
const TransportDescription* current_tdesc =
GetTransportDescription(content_name, current_desc);
std::unique_ptr<TransportDescription> new_tdesc(
transport_desc_factory_->CreateOffer(transport_options, current_tdesc,
ice_credentials));
if (!new_tdesc) {
RTC_LOG(LS_ERROR) << "Failed to AddTransportOffer, content name="
<< content_name;
}
offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc));
return RTCError::OK();
}
std::unique_ptr<TransportDescription>
MediaSessionDescriptionFactory::CreateTransportAnswer(
const std::string& content_name,
const SessionDescription* offer_desc,
const TransportOptions& transport_options,
const SessionDescription* current_desc,
bool require_transport_attributes,
IceCredentialsIterator* ice_credentials) const {
const TransportDescription* offer_tdesc =
GetTransportDescription(content_name, offer_desc);
const TransportDescription* current_tdesc =
GetTransportDescription(content_name, current_desc);
return transport_desc_factory_->CreateAnswer(offer_tdesc, transport_options,
require_transport_attributes,
current_tdesc, ice_credentials);
}
RTCError MediaSessionDescriptionFactory::AddTransportAnswer(
const std::string& content_name,
const TransportDescription& transport_desc,
SessionDescription* answer_desc) const {
answer_desc->AddTransportInfo(TransportInfo(content_name, transport_desc));
return RTCError::OK();
}
// `audio_codecs` = set of all possible codecs that can be used, with correct
// payload type mappings
//
// `supported_audio_codecs` = set of codecs that are supported for the direction
// of this m= section
//
// mcd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
RTCError MediaSessionDescriptionFactory::AddAudioContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
const RtpHeaderExtensions& audio_rtp_extensions,
const AudioCodecs& audio_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
// Filter audio_codecs (which includes all codecs, with correctly remapped
// payload types) based on transceiver direction.
const AudioCodecs& supported_audio_codecs =
GetAudioCodecsForOffer(media_description_options.direction);
AudioCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
// Add the codecs from the current transceiver's codec preferences.
// They override any existing codecs from previous negotiations.
filtered_codecs = MatchCodecPreference(
media_description_options.codec_preferences, audio_codecs,
supported_audio_codecs, field_trials);
} else {
// Add the codecs from current content if it exists and is not rejected nor
// recycled.
if (current_content && !current_content->rejected &&
current_content->name == media_description_options.mid) {
if (!IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)) {
// Can happen if the remote side re-uses a MID while recycling.
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Media type for content with mid='" +
current_content->name +
"' does not match previous type.");
}
const MediaContentDescription* mcd = current_content->media_description();
for (const Codec& codec : mcd->codecs()) {
if (FindMatchingCodec(mcd->codecs(), audio_codecs, codec,
field_trials)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported audio codecs.
for (const Codec& codec : supported_audio_codecs) {
absl::optional<Codec> found_codec = FindMatchingCodec(
supported_audio_codecs, audio_codecs, codec, field_trials);
if (found_codec &&
!FindMatchingCodec(supported_audio_codecs, filtered_codecs, codec,
field_trials)) {
// Use the `found_codec` from `audio_codecs` because it has the
// correctly mapped payload type.
filtered_codecs.push_back(*found_codec);
}
}
}
if (!session_options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(&filtered_codecs);
}
cricket::SecurePolicy sdes_policy =
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
: secure();
auto audio = std::make_unique<AudioContentDescription>();
std::vector<std::string> crypto_suites;
GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
&crypto_suites);
auto error = CreateMediaContentOffer(
media_description_options, session_options, filtered_codecs, sdes_policy,
GetCryptos(current_content), crypto_suites, audio_rtp_extensions,
ssrc_generator(), current_streams, audio.get(),
transport_desc_factory_->trials());
if (!error.ok()) {
return error;
}
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, audio.get());
audio->set_direction(media_description_options.direction);
desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
media_description_options.stopped, std::move(audio));
error = AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials);
if (!error.ok()) {
return error;
}
return RTCError::OK();
}
// TODO(kron): This function is very similar to AddAudioContentForOffer.
// Refactor to reuse shared code.
RTCError MediaSessionDescriptionFactory::AddVideoContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
const RtpHeaderExtensions& video_rtp_extensions,
const VideoCodecs& video_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
// Filter video_codecs (which includes all codecs, with correctly remapped
// payload types) based on transceiver direction.
const VideoCodecs& supported_video_codecs =
GetVideoCodecsForOffer(media_description_options.direction);
VideoCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
// Add the codecs from the current transceiver's codec preferences.
// They override any existing codecs from previous negotiations.
filtered_codecs = MatchCodecPreference(
media_description_options.codec_preferences, video_codecs,
supported_video_codecs, field_trials);
} else {
// Add the codecs from current content if it exists and is not rejected nor
// recycled.
if (current_content && !current_content->rejected &&
current_content->name == media_description_options.mid) {
if (!IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)) {
// Can happen if the remote side re-uses a MID while recycling.
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Media type for content with mid='" +
current_content->name +
"' does not match previous type.");
}
const MediaContentDescription* mcd = current_content->media_description();
for (const Codec& codec : mcd->codecs()) {
if (FindMatchingCodec(mcd->codecs(), video_codecs, codec,
field_trials)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported video codecs.
for (const Codec& codec : supported_video_codecs) {
absl::optional<Codec> found_codec = FindMatchingCodec(
supported_video_codecs, video_codecs, codec, field_trials);
if (found_codec &&
!FindMatchingCodec(supported_video_codecs, filtered_codecs, codec,
field_trials)) {
// Use the `found_codec` from `video_codecs` because it has the
// correctly mapped payload type.
if (IsRtxCodec(codec)) {
// For RTX we might need to adjust the apt parameter if we got a
// remote offer without RTX for a codec for which we support RTX.
auto referenced_codec =
GetAssociatedCodecForRtx(supported_video_codecs, codec);
RTC_DCHECK(referenced_codec);
// Find the codec we should be referencing and point to it.
absl::optional<Codec> changed_referenced_codec =
FindMatchingCodec(supported_video_codecs, filtered_codecs,
*referenced_codec, field_trials);
if (changed_referenced_codec) {
found_codec->SetParam(kCodecParamAssociatedPayloadType,
changed_referenced_codec->id);
}
}
filtered_codecs.push_back(*found_codec);
}
}
}
if (session_options.raw_packetization_for_video) {
for (Codec& codec : filtered_codecs) {
if (codec.IsMediaCodec()) {
codec.packetization = kPacketizationParamRaw;
}
}
}
cricket::SecurePolicy sdes_policy =
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
: secure();
auto video = std::make_unique<VideoContentDescription>();
std::vector<std::string> crypto_suites;
GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
&crypto_suites);
auto error = CreateMediaContentOffer(
media_description_options, session_options, filtered_codecs, sdes_policy,
GetCryptos(current_content), crypto_suites, video_rtp_extensions,
ssrc_generator(), current_streams, video.get(),
transport_desc_factory_->trials());
if (!error.ok()) {
return error;
}
video->set_bandwidth(kAutoBandwidth);
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
SetMediaProtocol(secure_transport, video.get());
video->set_direction(media_description_options.direction);
desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
media_description_options.stopped, std::move(video));
return AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials);
}
RTCError MediaSessionDescriptionFactory::AddDataContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
auto data = std::make_unique<SctpDataContentDescription>();
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
cricket::SecurePolicy sdes_policy =
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
: secure();
std::vector<std::string> crypto_suites;
// SDES doesn't make sense for SCTP, so we disable it, and we only
// get SDES crypto suites for RTP-based data channels.
sdes_policy = cricket::SEC_DISABLED;
// Unlike SetMediaProtocol below, we need to set the protocol
// before we call CreateMediaContentOffer. Otherwise,
// CreateMediaContentOffer won't know this is SCTP and will
// generate SSRCs rather than SIDs.
data->set_protocol(secure_transport ? kMediaProtocolUdpDtlsSctp
: kMediaProtocolSctp);
data->set_use_sctpmap(session_options.use_obsolete_sctp_sdp);
data->set_max_message_size(kSctpSendBufferSize);
auto error = CreateContentOffer(
media_description_options, session_options, sdes_policy,
GetCryptos(current_content), crypto_suites, RtpHeaderExtensions(),
ssrc_generator(), current_streams, data.get());
if (!error.ok()) {
return error;
}
desc->AddContent(media_description_options.mid, MediaProtocolType::kSctp,
media_description_options.stopped, std::move(data));
return AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials);
}
RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_UNSUPPORTED));
const UnsupportedContentDescription* current_unsupported_description =
current_content->media_description()->as_unsupported();
auto unsupported = std::make_unique<UnsupportedContentDescription>(
current_unsupported_description->media_type());
unsupported->set_protocol(current_content->media_description()->protocol());
desc->AddContent(media_description_options.mid, MediaProtocolType::kOther,
/*rejected=*/true, std::move(unsupported));
return AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials);
}
// `audio_codecs` = set of all possible codecs that can be used, with correct
// payload type mappings
//
// `supported_audio_codecs` = set of codecs that are supported for the direction
// of this m= section
//
// mcd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from mcd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
RTCError MediaSessionDescriptionFactory::AddAudioContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* offer_content,
const SessionDescription* offer_description,
const ContentInfo* current_content,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
const AudioCodecs& audio_codecs,
const RtpHeaderExtensions& rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO));
const AudioContentDescription* offer_audio_description =
offer_content->media_description()->as_audio();
std::unique_ptr<TransportDescription> audio_transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport != nullptr, ice_credentials);
if (!audio_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, audio transport is missing");
}
// Pick codecs based on the requested communications direction in the offer
// and the selected direction in the answer.
// Note these will be filtered one final time in CreateMediaContentAnswer.
auto wants_rtd = media_description_options.direction;
auto offer_rtd = offer_audio_description->direction();
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
AudioCodecs supported_audio_codecs =
GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
AudioCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
filtered_codecs = MatchCodecPreference(
media_description_options.codec_preferences, audio_codecs,
supported_audio_codecs, field_trials);
} else {
// Add the codecs from current content if it exists and is not rejected nor
// recycled.
if (current_content && !current_content->rejected &&
current_content->name == media_description_options.mid) {
if (!IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO)) {
// Can happen if the remote side re-uses a MID while recycling.
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Media type for content with mid='" +
current_content->name +
"' does not match previous type.");
}
const MediaContentDescription* mcd = current_content->media_description();
for (const Codec& codec : mcd->codecs()) {
if (FindMatchingCodec(mcd->codecs(), audio_codecs, codec,
field_trials)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported audio codecs.
for (const Codec& codec : supported_audio_codecs) {
if (FindMatchingCodec(supported_audio_codecs, audio_codecs, codec,
field_trials) &&
!FindMatchingCodec(supported_audio_codecs, filtered_codecs, codec,
field_trials)) {
// We should use the local codec with local parameters and the codec id
// would be correctly mapped in `NegotiateCodecs`.
filtered_codecs.push_back(codec);
}
}
}
if (!session_options.vad_enabled) {
// If application doesn't want CN codecs in answer.
StripCNCodecs(&filtered_codecs);
}
// Determine if we have media codecs in common.
bool has_common_media_codecs =
std::find_if(filtered_codecs.begin(), filtered_codecs.end(),
[](const AudioCodec& c) {
return !(IsRedCodec(c) || IsComfortNoiseCodec(c));
}) != filtered_codecs.end();
bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;
auto audio_answer = std::make_unique<AudioContentDescription>();
// Do not require or create SDES cryptos if DTLS is used.
cricket::SecurePolicy sdes_policy =
audio_transport->secure() ? cricket::SEC_DISABLED : secure();
if (!SetCodecsInAnswer(offer_audio_description, filtered_codecs,
media_description_options, session_options,
ssrc_generator(), current_streams, audio_answer.get(),
transport_desc_factory_->trials())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to set codecs in answer");
}
if (!CreateMediaContentAnswer(
offer_audio_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
filtered_rtp_header_extensions(rtp_header_extensions),
ssrc_generator(), enable_encrypted_rtp_header_extensions_,
current_streams, bundle_enabled, audio_answer.get())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create answer");
}
bool secure = bundle_transport ? bundle_transport->description.secure()
: audio_transport->secure();
bool rejected = media_description_options.stopped ||
offer_content->rejected || !has_common_media_codecs ||
!IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
audio_answer->protocol(), secure);
auto error = AddTransportAnswer(media_description_options.mid,
*(audio_transport.get()), answer);
if (!error.ok()) {
return error;
}
if (rejected) {
RTC_LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid
<< "' being rejected in answer.";
}
answer->AddContent(media_description_options.mid, offer_content->type,
rejected, std::move(audio_answer));
return RTCError::OK();
}
// TODO(kron): This function is very similar to AddAudioContentForAnswer.
// Refactor to reuse shared code.
RTCError MediaSessionDescriptionFactory::AddVideoContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* offer_content,
const SessionDescription* offer_description,
const ContentInfo* current_content,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
const VideoCodecs& video_codecs,
const RtpHeaderExtensions& default_video_rtp_header_extensions,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_VIDEO));
const VideoContentDescription* offer_video_description =
offer_content->media_description()->as_video();
std::unique_ptr<TransportDescription> video_transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport != nullptr, ice_credentials);
if (!video_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, video transport is missing");
}
// Pick codecs based on the requested communications direction in the offer
// and the selected direction in the answer.
// Note these will be filtered one final time in CreateMediaContentAnswer.
auto wants_rtd = media_description_options.direction;
auto offer_rtd = offer_video_description->direction();
auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
VideoCodecs supported_video_codecs =
GetVideoCodecsForAnswer(offer_rtd, answer_rtd);
VideoCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
filtered_codecs = MatchCodecPreference(
media_description_options.codec_preferences, video_codecs,
supported_video_codecs, field_trials);
} else {
// Add the codecs from current content if it exists and is not rejected nor
// recycled.
if (current_content && !current_content->rejected &&
current_content->name == media_description_options.mid) {
if (!IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO)) {
// Can happen if the remote side re-uses a MID while recycling.
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Media type for content with mid='" +
current_content->name +
"' does not match previous type.");
}
const MediaContentDescription* mcd = current_content->media_description();
for (const Codec& codec : mcd->codecs()) {
if (FindMatchingCodec(mcd->codecs(), video_codecs, codec,
field_trials)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported video codecs.
VideoCodecs other_video_codecs;
for (const Codec& codec : supported_video_codecs) {
if (FindMatchingCodec(supported_video_codecs, video_codecs, codec,
field_trials) &&
!FindMatchingCodec(supported_video_codecs, filtered_codecs, codec,
field_trials)) {
// We should use the local codec with local parameters and the codec id
// would be correctly mapped in `NegotiateCodecs`.
other_video_codecs.push_back(codec);
}
}
// Use ComputeCodecsUnion to avoid having duplicate payload IDs
filtered_codecs =
ComputeCodecsUnion(filtered_codecs, other_video_codecs, field_trials);
}
// Determine if we have media codecs in common.
bool has_common_media_codecs =
std::find_if(
filtered_codecs.begin(), filtered_codecs.end(), [](const Codec& c) {
return !(IsRedCodec(c) || IsUlpfecCodec(c) || IsFlexfecCodec(c));
}) != filtered_codecs.end();
if (session_options.raw_packetization_for_video) {
for (Codec& codec : filtered_codecs) {
if (codec.IsMediaCodec()) {
codec.packetization = kPacketizationParamRaw;
}
}
}
bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;
auto video_answer = std::make_unique<VideoContentDescription>();
// Do not require or create SDES cryptos if DTLS is used.
cricket::SecurePolicy sdes_policy =
video_transport->secure() ? cricket::SEC_DISABLED : secure();
if (!SetCodecsInAnswer(offer_video_description, filtered_codecs,
media_description_options, session_options,
ssrc_generator(), current_streams, video_answer.get(),
transport_desc_factory_->trials())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to set codecs in answer");
}
if (!CreateMediaContentAnswer(
offer_video_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
filtered_rtp_header_extensions(default_video_rtp_header_extensions),
ssrc_generator(), enable_encrypted_rtp_header_extensions_,
current_streams, bundle_enabled, video_answer.get())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create answer");
}
bool secure = bundle_transport ? bundle_transport->description.secure()
: video_transport->secure();
bool rejected = media_description_options.stopped ||
offer_content->rejected || !has_common_media_codecs ||
!IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
video_answer->protocol(), secure);
auto error = AddTransportAnswer(media_description_options.mid,
*(video_transport.get()), answer);
if (!error.ok()) {
return error;
}
if (!rejected) {
video_answer->set_bandwidth(kAutoBandwidth);
} else {
RTC_LOG(LS_INFO) << "Video m= section '" << media_description_options.mid
<< "' being rejected in answer.";
}
answer->AddContent(media_description_options.mid, offer_content->type,
rejected, std::move(video_answer));
return RTCError::OK();
}
RTCError MediaSessionDescriptionFactory::AddDataContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* offer_content,
const SessionDescription* offer_description,
const ContentInfo* current_content,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
std::unique_ptr<TransportDescription> data_transport = CreateTransportAnswer(
media_description_options.mid, offer_description,
media_description_options.transport_options, current_description,
bundle_transport != nullptr, ice_credentials);
if (!data_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, data transport is missing");
}
// Do not require or create SDES cryptos if DTLS is used.
cricket::SecurePolicy sdes_policy =
data_transport->secure() ? cricket::SEC_DISABLED : secure();
bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_DATA));
std::unique_ptr<MediaContentDescription> data_answer;
if (offer_content->media_description()->as_sctp()) {
// SCTP data content
data_answer = std::make_unique<SctpDataContentDescription>();
const SctpDataContentDescription* offer_data_description =
offer_content->media_description()->as_sctp();
// Respond with the offerer's proto, whatever it is.
data_answer->as_sctp()->set_protocol(offer_data_description->protocol());
// Respond with our max message size or the remote max messsage size,
// whichever is smaller.
// 0 is treated specially - it means "I can accept any size". Since
// we do not implement infinite size messages, reply with
// kSctpSendBufferSize.
if (offer_data_description->max_message_size() == 0) {
data_answer->as_sctp()->set_max_message_size(kSctpSendBufferSize);
} else {
data_answer->as_sctp()->set_max_message_size(std::min(
offer_data_description->max_message_size(), kSctpSendBufferSize));
}
if (!CreateMediaContentAnswer(
offer_data_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content), RtpHeaderExtensions(),
ssrc_generator(), enable_encrypted_rtp_header_extensions_,
current_streams, bundle_enabled, data_answer.get())) {
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
"Failed to create answer");
}
// Respond with sctpmap if the offer uses sctpmap.
bool offer_uses_sctpmap = offer_data_description->use_sctpmap();
data_answer->as_sctp()->set_use_sctpmap(offer_uses_sctpmap);
} else {
RTC_DCHECK_NOTREACHED() << "Non-SCTP data content found";
}
bool secure = bundle_transport ? bundle_transport->description.secure()
: data_transport->secure();
bool rejected = media_description_options.stopped ||
offer_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_DATA,
data_answer->protocol(), secure);
auto error = AddTransportAnswer(media_description_options.mid,
*(data_transport.get()), answer);
if (!error.ok()) {
return error;
}
answer->AddContent(media_description_options.mid, offer_content->type,
rejected, std::move(data_answer));
return RTCError::OK();
}
RTCError MediaSessionDescriptionFactory::AddUnsupportedContentForAnswer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* offer_content,
const SessionDescription* offer_description,
const ContentInfo* current_content,
const SessionDescription* current_description,
const TransportInfo* bundle_transport,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
std::unique_ptr<TransportDescription> unsupported_transport =
CreateTransportAnswer(media_description_options.mid, offer_description,
media_description_options.transport_options,
current_description, bundle_transport != nullptr,
ice_credentials);
if (!unsupported_transport) {
LOG_AND_RETURN_ERROR(
RTCErrorType::INTERNAL_ERROR,
"Failed to create transport answer, unsupported transport is missing");
}
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_UNSUPPORTED));
const UnsupportedContentDescription* offer_unsupported_description =
offer_content->media_description()->as_unsupported();
std::unique_ptr<MediaContentDescription> unsupported_answer =
std::make_unique<UnsupportedContentDescription>(
offer_unsupported_description->media_type());
unsupported_answer->set_protocol(offer_unsupported_description->protocol());
auto error = AddTransportAnswer(media_description_options.mid,
*(unsupported_transport.get()), answer);
if (!error.ok()) {
return error;
}
answer->AddContent(media_description_options.mid, offer_content->type,
/*rejected=*/true, std::move(unsupported_answer));
return RTCError::OK();
}
void MediaSessionDescriptionFactory::ComputeAudioCodecsIntersectionAndUnion() {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
audio_sendrecv_codecs_.clear();
all_audio_codecs_.clear();
// Compute the audio codecs union.
for (const Codec& send : audio_send_codecs_) {
all_audio_codecs_.push_back(send);
if (!FindMatchingCodec(audio_send_codecs_, audio_recv_codecs_, send,
field_trials)) {
// It doesn't make sense to have an RTX codec we support sending but not
// receiving.
RTC_DCHECK(!IsRtxCodec(send));
}
}
for (const Codec& recv : audio_recv_codecs_) {
if (!FindMatchingCodec(audio_recv_codecs_, audio_send_codecs_, recv,
field_trials)) {
all_audio_codecs_.push_back(recv);
}
}
// Use NegotiateCodecs to merge our codec lists, since the operation is
// essentially the same. Put send_codecs as the offered_codecs, which is the
// order we'd like to follow. The reasoning is that encoding is usually more
// expensive than decoding, and prioritizing a codec in the send list probably
// means it's a codec we can handle efficiently.
NegotiateCodecs(audio_recv_codecs_, audio_send_codecs_,
&audio_sendrecv_codecs_, true, field_trials);
}
void MediaSessionDescriptionFactory::ComputeVideoCodecsIntersectionAndUnion() {
const webrtc::FieldTrialsView* field_trials =
&transport_desc_factory_->trials();
video_sendrecv_codecs_.clear();
// Use ComputeCodecsUnion to avoid having duplicate payload IDs
all_video_codecs_ =
ComputeCodecsUnion(video_recv_codecs_, video_send_codecs_, field_trials);
// Use NegotiateCodecs to merge our codec lists, since the operation is
// essentially the same. Put send_codecs as the offered_codecs, which is the
// order we'd like to follow. The reasoning is that encoding is usually more
// expensive than decoding, and prioritizing a codec in the send list probably
// means it's a codec we can handle efficiently.
NegotiateCodecs(video_recv_codecs_, video_send_codecs_,
&video_sendrecv_codecs_, true, field_trials);
}
bool IsMediaContent(const ContentInfo* content) {
return (content && (content->type == MediaProtocolType::kRtp ||
content->type == MediaProtocolType::kSctp));
}
bool IsAudioContent(const ContentInfo* content) {
return IsMediaContentOfType(content, MEDIA_TYPE_AUDIO);
}
bool IsVideoContent(const ContentInfo* content) {
return IsMediaContentOfType(content, MEDIA_TYPE_VIDEO);
}
bool IsDataContent(const ContentInfo* content) {
return IsMediaContentOfType(content, MEDIA_TYPE_DATA);
}
bool IsUnsupportedContent(const ContentInfo* content) {
return IsMediaContentOfType(content, MEDIA_TYPE_UNSUPPORTED);
}
const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
MediaType media_type) {
for (const ContentInfo& content : contents) {
if (IsMediaContentOfType(&content, media_type)) {
return &content;
}
}
return nullptr;
}
const ContentInfo* GetFirstAudioContent(const ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
}
const ContentInfo* GetFirstVideoContent(const ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
}
const ContentInfo* GetFirstDataContent(const ContentInfos& contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
}
const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
MediaType media_type) {
if (sdesc == nullptr) {
return nullptr;
}
return GetFirstMediaContent(sdesc->contents(), media_type);
}
const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
}
const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
}
const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
}
const MediaContentDescription* GetFirstMediaContentDescription(
const SessionDescription* sdesc,
MediaType media_type) {
const ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
return (content ? content->media_description() : nullptr);
}
const AudioContentDescription* GetFirstAudioContentDescription(
const SessionDescription* sdesc) {
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO);
return desc ? desc->as_audio() : nullptr;
}
const VideoContentDescription* GetFirstVideoContentDescription(
const SessionDescription* sdesc) {
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO);
return desc ? desc->as_video() : nullptr;
}
const SctpDataContentDescription* GetFirstSctpDataContentDescription(
const SessionDescription* sdesc) {
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
return desc ? desc->as_sctp() : nullptr;
}
//
// Non-const versions of the above functions.
//
ContentInfo* GetFirstMediaContent(ContentInfos* contents,
MediaType media_type) {
for (ContentInfo& content : *contents) {
if (IsMediaContentOfType(&content, media_type)) {
return &content;
}
}
return nullptr;
}
ContentInfo* GetFirstAudioContent(ContentInfos* contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_AUDIO);
}
ContentInfo* GetFirstVideoContent(ContentInfos* contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_VIDEO);
}
ContentInfo* GetFirstDataContent(ContentInfos* contents) {
return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
}
ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
MediaType media_type) {
if (sdesc == nullptr) {
return nullptr;
}
return GetFirstMediaContent(&sdesc->contents(), media_type);
}
ContentInfo* GetFirstAudioContent(SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_AUDIO);
}
ContentInfo* GetFirstVideoContent(SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_VIDEO);
}
ContentInfo* GetFirstDataContent(SessionDescription* sdesc) {
return GetFirstMediaContent(sdesc, MEDIA_TYPE_DATA);
}
MediaContentDescription* GetFirstMediaContentDescription(
SessionDescription* sdesc,
MediaType media_type) {
ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
return (content ? content->media_description() : nullptr);
}
AudioContentDescription* GetFirstAudioContentDescription(
SessionDescription* sdesc) {
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_AUDIO);
return desc ? desc->as_audio() : nullptr;
}
VideoContentDescription* GetFirstVideoContentDescription(
SessionDescription* sdesc) {
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_VIDEO);
return desc ? desc->as_video() : nullptr;
}
SctpDataContentDescription* GetFirstSctpDataContentDescription(
SessionDescription* sdesc) {
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
return desc ? desc->as_sctp() : nullptr;
}
} // namespace cricket