blob: e229ed67589ac7936f80bc37a6aab384ddb33c1c [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 <algorithm>
#include <functional>
#include <map>
#include <memory>
#include <set>
#include <unordered_map>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "absl/types/optional.h"
#include "api/crypto_params.h"
#include "media/base/h264_profile_level_id.h"
#include "media/base/media_constants.h"
#include "media/sctp/sctp_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "pc/channel_manager.h"
#include "pc/media_protocol_names.h"
#include "pc/rtp_media_utils.h"
#include "pc/srtp_filter.h"
#include "pc/used_ids.h"
#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/third_party/base64/base64.h"
#include "rtc_base/unique_id_generator.h"
namespace {
using rtc::UniqueRandomIdGenerator;
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));
}
}
} // namespace
namespace cricket {
// RTP Profile names
// http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml
// RFC4585
const char kMediaProtocolAvpf[] = "RTP/AVPF";
// RFC5124
const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF";
// We always generate offers with "UDP/TLS/RTP/SAVPF" when using DTLS-SRTP,
// but we tolerate "RTP/SAVPF" in offers we receive, for compatibility.
const char kMediaProtocolSavpf[] = "RTP/SAVPF";
// Note that the below functions support some protocol strings purely for
// legacy compatibility, as required by JSEP in Section 5.1.2, Profile Names
// and Interoperability.
static bool IsDtlsRtp(const std::string& protocol) {
// Most-likely values first.
return protocol == "UDP/TLS/RTP/SAVPF" || protocol == "TCP/TLS/RTP/SAVPF" ||
protocol == "UDP/TLS/RTP/SAVP" || protocol == "TCP/TLS/RTP/SAVP";
}
static bool IsPlainRtp(const std::string& protocol) {
// Most-likely values first.
return protocol == "RTP/SAVPF" || protocol == "RTP/AVPF" ||
protocol == "RTP/SAVP" || protocol == "RTP/AVP";
}
static 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);
}
static bool IsMediaContentOfType(const ContentInfo* content,
MediaType media_type) {
if (!content || !content->media_description()) {
return false;
}
return content->media_description()->type() == media_type;
}
static 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->cipher_suite = cipher;
crypto_out->key_params = kInline;
crypto_out->key_params += key;
return true;
}
static bool AddCryptoParams(const std::string& cipher_suite,
CryptoParamsVec* cryptos_out) {
int size = static_cast<int>(cryptos_out->size());
cryptos_out->resize(size + 1);
return CreateCryptoParams(size, cipher_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_gcm_crypto_suites) {
crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
}
if (crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher) {
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
}
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
}
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) {
if (crypto_options.srtp.enable_gcm_crypto_suites) {
crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
}
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
}
void GetSupportedVideoSdesCryptoSuiteNames(
const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites,
crypto_options, crypto_suite_names);
}
void GetSupportedDataSdesCryptoSuites(
const webrtc::CryptoOptions& crypto_options,
std::vector<int>* crypto_suites) {
if (crypto_options.srtp.enable_gcm_crypto_suites) {
crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
}
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
}
void GetSupportedDataSdesCryptoSuiteNames(
const webrtc::CryptoOptions& crypto_options,
std::vector<std::string>* crypto_suite_names) {
GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites,
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.
static 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.cipher_suite)) ||
rtc::CS_AES_CM_128_HMAC_SHA1_80 == crypto.cipher_suite ||
(rtc::CS_AES_CM_128_HMAC_SHA1_32 == crypto.cipher_suite && audio &&
!bundle && crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher)) {
return CreateCryptoParams(crypto.tag, crypto.cipher_suite, crypto_out);
}
}
return false;
}
// Finds all StreamParams of all media types and attach them to stream_params.
static 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;
}
// Filters the data codecs for the data channel type.
void FilterDataCodecs(std::vector<DataCodec>* codecs, bool sctp) {
// Filter RTP codec for SCTP and vice versa.
const char* codec_name =
sctp ? kGoogleRtpDataCodecName : kGoogleSctpDataCodecName;
codecs->erase(std::remove_if(codecs->begin(), codecs->end(),
[&codec_name](const DataCodec& codec) {
return absl::EqualsIgnoreCase(codec.name,
codec_name);
}),
codecs->end());
}
static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
const SenderOptions& sender,
const std::string& rtcp_cname,
bool include_rtx_streams,
bool include_flexfec_stream,
UniqueRandomIdGenerator* ssrc_generator) {
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.";
}
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;
}
static 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;
});
});
}
static 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.
static 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.
template <class C>
static bool AddStreamParams(
const std::vector<SenderOptions>& sender_options,
const std::string& rtcp_cname,
UniqueRandomIdGenerator* ssrc_generator,
StreamParamsVec* current_streams,
MediaContentDescriptionImpl<C>* content_description) {
// 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) {
// groupid is empty for StreamParams generated using
// MediaSessionDescriptionFactory.
StreamParams* param =
GetStreamByIds(*current_streams, "" /*group_id*/, 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)
:
// 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|.
static 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;
const absl::optional<OpaqueTransportParameters>& selected_opaque_parameters =
selected_transport_info->description.opaque_parameters;
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;
transport_info.description.opaque_parameters = selected_opaque_parameters;
}
}
return true;
}
// Gets the CryptoParamsVec of the given |content_name| from |sdesc|, and
// sets it to |cryptos|.
static 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 (cipher_suite)
// which are not available in |filter|.
static 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 cipher_suite is not
// found in |filter|.
[&filter](const CryptoParams& crypto) {
for (const CryptoParams& entry : filter) {
if (entry.cipher_suite == crypto.cipher_suite)
return false;
}
return true;
}),
target_cryptos->end());
}
static 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.
static 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;
}
static 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;
}
template <class C>
static bool ContainsRtxCodec(const std::vector<C>& codecs) {
for (const auto& codec : codecs) {
if (IsRtxCodec(codec)) {
return true;
}
}
return false;
}
template <class C>
static bool IsRtxCodec(const C& codec) {
return absl::EqualsIgnoreCase(codec.name, kRtxCodecName);
}
template <class C>
static bool ContainsFlexfecCodec(const std::vector<C>& codecs) {
for (const auto& codec : codecs) {
if (IsFlexfecCodec(codec)) {
return true;
}
}
return false;
}
template <class C>
static bool IsFlexfecCodec(const C& codec) {
return absl::EqualsIgnoreCase(codec.name, kFlexfecCodecName);
}
// 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.
static bool 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);
}
offer->set_rtp_header_extensions(rtp_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)) {
return false;
}
}
}
if (secure_policy == SEC_REQUIRED && offer->cryptos().empty()) {
return false;
}
return true;
}
template <class C>
static bool CreateMediaContentOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const std::vector<C>& 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,
MediaContentDescriptionImpl<C>* offer) {
offer->AddCodecs(codecs);
if (!AddStreamParams(media_description_options.sender_options,
session_options.rtcp_cname, ssrc_generator,
current_streams, offer)) {
return false;
}
return CreateContentOffer(media_description_options, session_options,
secure_policy, current_cryptos, crypto_suites,
rtp_extensions, ssrc_generator, current_streams,
offer);
}
template <class C>
static bool ReferencedCodecsMatch(const std::vector<C>& codecs1,
const int codec1_id,
const std::vector<C>& codecs2,
const int codec2_id) {
const C* codec1 = FindCodecById(codecs1, codec1_id);
const C* codec2 = FindCodecById(codecs2, codec2_id);
return codec1 != nullptr && codec2 != nullptr && codec1->Matches(*codec2);
}
template <class C>
static void NegotiatePacketization(const C& local_codec,
const C& remote_codec,
C* negotiated_codec) {}
template <>
void NegotiatePacketization(const VideoCodec& local_codec,
const VideoCodec& remote_codec,
VideoCodec* negotiated_codec) {
negotiated_codec->packetization =
VideoCodec::IntersectPacketization(local_codec, remote_codec);
}
template <class C>
static void NegotiateCodecs(const std::vector<C>& local_codecs,
const std::vector<C>& offered_codecs,
std::vector<C>* negotiated_codecs,
bool keep_offer_order) {
for (const C& ours : local_codecs) {
C theirs;
// Note that we intentionally only find one matching codec for each of our
// local codecs, in case the remote offer contains duplicate codecs.
if (FindMatchingCodec(local_codecs, offered_codecs, ours, &theirs)) {
C 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);
}
if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
webrtc::H264::GenerateProfileLevelIdForAnswer(
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 C& codec : offered_codecs) {
payload_type_preferences[codec.id] = preference--;
}
absl::c_sort(*negotiated_codecs, [&payload_type_preferences](const C& a,
const C& b) {
return payload_type_preferences[a.id] > payload_type_preferences[b.id];
});
}
}
// Finds a codec in |codecs2| that matches |codec_to_match|, which is
// a member of |codecs1|. If |codec_to_match| is an RTX codec, both
// the codecs themselves and their associated codecs must match.
template <class C>
static bool FindMatchingCodec(const std::vector<C>& codecs1,
const std::vector<C>& codecs2,
const C& codec_to_match,
C* found_codec) {
// |codec_to_match| should be a member of |codecs1|, in order to look up RTX
// codecs' associated codecs correctly. If not, that's a programming error.
RTC_DCHECK(absl::c_any_of(codecs1, [&codec_to_match](const C& codec) {
return &codec == &codec_to_match;
}));
for (const C& potential_match : codecs2) {
if (potential_match.Matches(codec_to_match)) {
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)) {
continue;
}
}
if (found_codec) {
*found_codec = potential_match;
}
return true;
}
}
return false;
}
// Find the codec in |codec_list| that |rtx_codec| is associated with.
template <class C>
static const C* GetAssociatedCodec(const std::vector<C>& codec_list,
const C& rtx_codec) {
std::string associated_pt_str;
if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
&associated_pt_str)) {
RTC_LOG(LS_WARNING) << "RTX codec " << rtx_codec.name
<< " 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.name
<< " to an integer.";
return nullptr;
}
// Find the associated reference codec for the reference RTX codec.
const C* 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.name
<< ".";
}
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.
template <class C>
static void MergeCodecs(const std::vector<C>& reference_codecs,
std::vector<C>* offered_codecs,
UsedPayloadTypes* used_pltypes) {
// Add all new codecs that are not RTX codecs.
for (const C& reference_codec : reference_codecs) {
if (!IsRtxCodec(reference_codec) &&
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
reference_codec, nullptr)) {
C codec = reference_codec;
used_pltypes->FindAndSetIdUsed(&codec);
offered_codecs->push_back(codec);
}
}
// Add all new RTX codecs.
for (const C& reference_codec : reference_codecs) {
if (IsRtxCodec(reference_codec) &&
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
reference_codec, nullptr)) {
C rtx_codec = reference_codec;
const C* associated_codec =
GetAssociatedCodec(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.
C matching_codec;
if (!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
*associated_codec, &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);
}
}
}
template <typename Codecs>
static Codecs MatchCodecPreference(
const std::vector<webrtc::RtpCodecCapability>& codec_preferences,
const Codecs& codecs) {
Codecs filtered_codecs;
std::set<std::string> kept_codecs_ids;
bool want_rtx = false;
for (const auto& codec_preference : codec_preferences) {
auto found_codec = absl::c_find_if(
codecs, [&codec_preference](const typename Codecs::value_type& 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 != codecs.end()) {
filtered_codecs.push_back(*found_codec);
kept_codecs_ids.insert(std::to_string(found_codec->id));
} else if (IsRtxCodec(codec_preference)) {
want_rtx = true;
}
}
if (want_rtx) {
for (const auto& codec : codecs) {
if (IsRtxCodec(codec)) {
const auto apt =
codec.params.find(cricket::kCodecParamAssociatedPayloadType);
if (apt != codec.params.end() &&
kept_codecs_ids.count(apt->second) > 0) {
filtered_codecs.push_back(codec);
}
}
}
}
return filtered_codecs;
}
static bool FindByUriAndEncryption(const RtpHeaderExtensions& extensions,
const webrtc::RtpExtension& ext_to_match,
webrtc::RtpExtension* found_extension) {
auto it = absl::c_find_if(
extensions, [&ext_to_match](const webrtc::RtpExtension& extension) {
// We assume that all URIs are given in a canonical
// format.
return extension.uri == ext_to_match.uri &&
extension.encrypt == ext_to_match.encrypt;
});
if (it == extensions.end()) {
return false;
}
if (found_extension) {
*found_extension = *it;
}
return true;
}
static bool FindByUri(const RtpHeaderExtensions& extensions,
const webrtc::RtpExtension& ext_to_match,
webrtc::RtpExtension* found_extension) {
// We assume that all URIs are given in a canonical format.
const webrtc::RtpExtension* found =
webrtc::RtpExtension::FindHeaderExtensionByUri(extensions,
ext_to_match.uri);
if (!found) {
return false;
}
if (found_extension) {
*found_extension = *found;
}
return true;
}
static bool FindByUriWithEncryptionPreference(
const RtpHeaderExtensions& extensions,
const webrtc::RtpExtension& ext_to_match,
bool encryption_preference,
webrtc::RtpExtension* found_extension) {
const webrtc::RtpExtension* unencrypted_extension = nullptr;
for (const webrtc::RtpExtension& extension : extensions) {
// We assume that all URIs are given in a canonical format.
if (extension.uri == ext_to_match.uri) {
if (!encryption_preference || extension.encrypt) {
if (found_extension) {
*found_extension = extension;
}
return true;
}
unencrypted_extension = &extension;
}
}
if (unencrypted_extension) {
if (found_extension) {
*found_extension = *unencrypted_extension;
}
return true;
}
return false;
}
// 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.
static 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 (!FindByUriAndEncryption(*offered_extensions, reference_extension,
nullptr)) {
webrtc::RtpExtension existing;
if (reference_extension.encrypt) {
if (FindByUriAndEncryption(*encrypted_extensions, reference_extension,
&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 {
if (FindByUriAndEncryption(*regular_extensions, reference_extension,
&existing)) {
offered_extensions->push_back(existing);
} else {
used_ids->FindAndSetIdUsed(&reference_extension);
regular_extensions->push_back(reference_extension);
offered_extensions->push_back(reference_extension);
}
}
}
}
}
static void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* extensions,
RtpHeaderExtensions* all_extensions,
UsedRtpHeaderExtensionIds* used_ids) {
RtpHeaderExtensions encrypted_extensions;
for (const webrtc::RtpExtension& extension : *extensions) {
webrtc::RtpExtension existing;
// Don't add encrypted extensions again that were already included in a
// previous offer or regular extensions that are also included as encrypted
// extensions.
if (extension.encrypt ||
!webrtc::RtpExtension::IsEncryptionSupported(extension.uri) ||
(FindByUriWithEncryptionPreference(*extensions, extension, true,
&existing) &&
existing.encrypt)) {
continue;
}
if (FindByUri(*all_extensions, extension, &existing)) {
encrypted_extensions.push_back(existing);
} else {
webrtc::RtpExtension encrypted(extension);
encrypted.encrypt = true;
used_ids->FindAndSetIdUsed(&encrypted);
all_extensions->push_back(encrypted);
encrypted_extensions.push_back(encrypted);
}
}
extensions->insert(extensions->end(), encrypted_extensions.begin(),
encrypted_extensions.end());
}
static void NegotiateRtpHeaderExtensions(
const RtpHeaderExtensions& local_extensions,
const RtpHeaderExtensions& offered_extensions,
bool enable_encrypted_rtp_header_extensions,
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 =
webrtc::RtpExtension::FindHeaderExtensionByUri(
offered_extensions,
webrtc::RtpExtension::kTransportSequenceNumberV2Uri);
for (const webrtc::RtpExtension& ours : local_extensions) {
webrtc::RtpExtension theirs;
if (FindByUriWithEncryptionPreference(
offered_extensions, ours, enable_encrypted_rtp_header_extensions,
&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);
}
}
static void StripCNCodecs(AudioCodecs* audio_codecs) {
audio_codecs->erase(std::remove_if(audio_codecs->begin(), audio_codecs->end(),
[](const AudioCodec& codec) {
return absl::EqualsIgnoreCase(
codec.name, kComfortNoiseCodecName);
}),
audio_codecs->end());
}
template <class C>
static bool SetCodecsInAnswer(
const MediaContentDescriptionImpl<C>* offer,
const std::vector<C>& local_codecs,
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
UniqueRandomIdGenerator* ssrc_generator,
StreamParamsVec* current_streams,
MediaContentDescriptionImpl<C>* answer) {
std::vector<C> negotiated_codecs;
NegotiateCodecs(local_codecs, offer->codecs(), &negotiated_codecs,
media_description_options.codec_preferences.empty());
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)) {
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.
static 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_extenstions,
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());
RtpHeaderExtensions negotiated_rtp_extensions;
NegotiateRtpHeaderExtensions(
local_rtp_extenstions, offer->rtp_header_extensions(),
enable_encrypted_rtp_header_extensions, &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;
}
static 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, but also for RTP for RTP-based data channels.
// TODO(pthatcher): Remove RTP once RTP-based data channels are gone.
if (secure_transport) {
// Most likely scenarios first.
return IsDtlsSctp(protocol) || IsDtlsRtp(protocol) ||
IsPlainRtp(protocol);
} else {
return IsPlainSctp(protocol) || IsPlainRtp(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);
}
}
static 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.
static 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.
static 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();
}
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::AddRtpDataChannel(const std::string& track_id,
const std::string& stream_id) {
RTC_DCHECK(type == MEDIA_TYPE_DATA);
// TODO(steveanton): Is it the case that RtpDataChannel will never have more
// than one stream?
AddSenderInternal(track_id, {stream_id}, {}, SimulcastLayerList(), 1);
}
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(
const TransportDescriptionFactory* transport_desc_factory,
rtc::UniqueRandomIdGenerator* ssrc_generator)
: ssrc_generator_(ssrc_generator),
transport_desc_factory_(transport_desc_factory) {
RTC_DCHECK(ssrc_generator_);
}
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
ChannelManager* channel_manager,
const TransportDescriptionFactory* transport_desc_factory,
rtc::UniqueRandomIdGenerator* ssrc_generator)
: MediaSessionDescriptionFactory(transport_desc_factory, ssrc_generator) {
channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
channel_manager->GetSupportedAudioRtpHeaderExtensions(&audio_rtp_extensions_);
channel_manager->GetSupportedVideoCodecs(&video_codecs_);
channel_manager->GetSupportedVideoRtpHeaderExtensions(&video_rtp_extensions_);
channel_manager->GetSupportedDataCodecs(&rtp_data_codecs_);
ComputeAudioCodecsIntersectionAndUnion();
}
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();
}
static void AddUnifiedPlanExtensions(RtpHeaderExtensions* extensions) {
RTC_DCHECK(extensions);
rtc::UniqueNumberGenerator<int> unique_id_generator;
unique_id_generator.AddKnownId(0); // The first valid RTP extension ID is 1.
for (const webrtc::RtpExtension& extension : *extensions) {
const bool collision_free = unique_id_generator.AddKnownId(extension.id);
RTC_DCHECK(collision_free);
}
// Unified Plan also offers the MID and RID header extensions.
extensions->push_back(webrtc::RtpExtension(webrtc::RtpExtension::kMidUri,
unique_id_generator()));
extensions->push_back(webrtc::RtpExtension(webrtc::RtpExtension::kRidUri,
unique_id_generator()));
extensions->push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kRepairedRidUri, unique_id_generator()));
}
RtpHeaderExtensions
MediaSessionDescriptionFactory::audio_rtp_header_extensions() const {
RtpHeaderExtensions extensions = audio_rtp_extensions_;
if (is_unified_plan_) {
AddUnifiedPlanExtensions(&extensions);
}
return extensions;
}
RtpHeaderExtensions
MediaSessionDescriptionFactory::video_rtp_header_extensions() const {
RtpHeaderExtensions extensions = video_rtp_extensions_;
if (is_unified_plan_) {
AddUnifiedPlanExtensions(&extensions);
}
return extensions;
}
std::unique_ptr<SessionDescription> MediaSessionDescriptionFactory::CreateOffer(
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;
RtpDataCodecs offer_rtp_data_codecs;
GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
&offer_video_codecs, &offer_rtp_data_codecs);
if (!session_options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(&offer_audio_codecs);
}
FilterDataCodecs(&offer_rtp_data_codecs,
session_options.data_channel_type == DCT_SCTP);
RtpHeaderExtensions audio_rtp_extensions;
RtpHeaderExtensions video_rtp_extensions;
GetRtpHdrExtsToOffer(current_active_contents,
session_options.offer_extmap_allow_mixed,
&audio_rtp_extensions, &video_rtp_extensions);
auto offer = absl::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.
RTC_DCHECK(current_content->name != media_description_options.mid ||
IsMediaContentOfType(current_content,
media_description_options.type));
}
switch (media_description_options.type) {
case MEDIA_TYPE_AUDIO:
if (!AddAudioContentForOffer(
media_description_options, session_options, current_content,
current_description, audio_rtp_extensions, offer_audio_codecs,
&current_streams, offer.get(), &ice_credentials)) {
return nullptr;
}
break;
case MEDIA_TYPE_VIDEO:
if (!AddVideoContentForOffer(
media_description_options, session_options, current_content,
current_description, video_rtp_extensions, offer_video_codecs,
&current_streams, offer.get(), &ice_credentials)) {
return nullptr;
}
break;
case MEDIA_TYPE_DATA:
if (!AddDataContentForOffer(media_description_options, session_options,
current_content, current_description,
offer_rtp_data_codecs, &current_streams,
offer.get(), &ice_credentials)) {
return nullptr;
}
break;
default:
RTC_NOTREACHED();
}
++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())) {
RTC_LOG(LS_ERROR)
<< "CreateOffer failed to UpdateTransportInfoForBundle.";
return nullptr;
}
if (!UpdateCryptoParamsForBundle(offer_bundle, offer.get())) {
RTC_LOG(LS_ERROR)
<< "CreateOffer failed to UpdateCryptoParamsForBundle.";
return nullptr;
}
}
}
// 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);
if (session_options.media_transport_settings.has_value()) {
offer->AddMediaTransportSetting(
session_options.media_transport_settings->transport_name,
session_options.media_transport_settings->transport_setting);
}
return offer;
}
std::unique_ptr<SessionDescription>
MediaSessionDescriptionFactory::CreateAnswer(
const SessionDescription* offer,
const MediaSessionOptions& session_options,
const SessionDescription* current_description) const {
if (!offer) {
return nullptr;
}
// 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;
RtpDataCodecs answer_rtp_data_codecs;
GetCodecsForAnswer(current_active_contents, *offer, &answer_audio_codecs,
&answer_video_codecs, &answer_rtp_data_codecs);
if (!session_options.vad_enabled) {
// If application doesn't want CN codecs in answer.
StripCNCodecs(&answer_audio_codecs);
}
FilterDataCodecs(&answer_rtp_data_codecs,
session_options.data_channel_type == DCT_SCTP);
auto answer = absl::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.
const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE);
ContentGroup answer_bundle(GROUP_TYPE_BUNDLE);
// Transport info shared by the bundle group.
std::unique_ptr<TransportInfo> bundle_transport;
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);
const ContentInfo* current_content = nullptr;
if (current_description &&
msection_index < current_description->contents().size()) {
current_content = &current_description->contents()[msection_index];
}
switch (media_description_options.type) {
case MEDIA_TYPE_AUDIO:
if (!AddAudioContentForAnswer(
media_description_options, session_options, offer_content,
offer, current_content, current_description,
bundle_transport.get(), answer_audio_codecs, &current_streams,
answer.get(), &ice_credentials)) {
return nullptr;
}
break;
case MEDIA_TYPE_VIDEO:
if (!AddVideoContentForAnswer(
media_description_options, session_options, offer_content,
offer, current_content, current_description,
bundle_transport.get(), answer_video_codecs, &current_streams,
answer.get(), &ice_credentials)) {
return nullptr;
}
break;
case MEDIA_TYPE_DATA:
if (!AddDataContentForAnswer(
media_description_options, session_options, offer_content,
offer, current_content, current_description,
bundle_transport.get(), answer_rtp_data_codecs,
&current_streams, answer.get(), &ice_credentials)) {
return nullptr;
}
break;
default:
RTC_NOTREACHED();
}
++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 && offer_bundle &&
offer_bundle->HasContentName(added.name)) {
answer_bundle.AddContentName(added.name);
bundle_transport.reset(
new TransportInfo(*answer->GetTransportInfoByName(added.name)));
}
}
// If a BUNDLE group was offered, put a BUNDLE group in the answer even if
// it's 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_bundle) {
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())) {
RTC_LOG(LS_ERROR)
<< "CreateAnswer failed to UpdateTransportInfoForBundle.";
return NULL;
}
if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
RTC_LOG(LS_ERROR)
<< "CreateAnswer failed to UpdateCryptoParamsForBundle.";
return NULL;
}
}
// 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::kInactive:
return audio_sendrecv_codecs_;
case RtpTransceiverDirection::kSendOnly:
return audio_send_codecs_;
case RtpTransceiverDirection::kRecvOnly:
return audio_recv_codecs_;
}
RTC_NOTREACHED();
return audio_sendrecv_codecs_;
}
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::kInactive:
return GetAudioCodecsForOffer(
webrtc::RtpTransceiverDirectionReversed(offer));
case RtpTransceiverDirection::kSendOnly:
return audio_send_codecs_;
case RtpTransceiverDirection::kRecvOnly:
return audio_recv_codecs_;
}
RTC_NOTREACHED();
return audio_sendrecv_codecs_;
}
void MergeCodecsFromDescription(
const std::vector<const ContentInfo*>& current_active_contents,
AudioCodecs* audio_codecs,
VideoCodecs* video_codecs,
RtpDataCodecs* rtp_data_codecs,
UsedPayloadTypes* used_pltypes) {
for (const ContentInfo* content : current_active_contents) {
if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
const AudioContentDescription* audio =
content->media_description()->as_audio();
MergeCodecs<AudioCodec>(audio->codecs(), audio_codecs, used_pltypes);
} else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
const VideoContentDescription* video =
content->media_description()->as_video();
MergeCodecs<VideoCodec>(video->codecs(), video_codecs, used_pltypes);
} else if (IsMediaContentOfType(content, MEDIA_TYPE_DATA)) {
const RtpDataContentDescription* data =
content->media_description()->as_rtp_data();
if (data) {
// Only relevant for RTP datachannels
MergeCodecs<RtpDataCodec>(data->codecs(), rtp_data_codecs,
used_pltypes);
}
}
}
}
// 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,
RtpDataCodecs* rtp_data_codecs) const {
// 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, rtp_data_codecs, &used_pltypes);
// Add our codecs that are not in the current description.
MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes);
MergeCodecs<VideoCodec>(video_codecs_, video_codecs, &used_pltypes);
MergeCodecs<DataCodec>(rtp_data_codecs_, rtp_data_codecs, &used_pltypes);
}
// 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,
RtpDataCodecs* rtp_data_codecs) const {
// 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, rtp_data_codecs, &used_pltypes);
// Second - filter out codecs that we don't support at all and should ignore.
AudioCodecs filtered_offered_audio_codecs;
VideoCodecs filtered_offered_video_codecs;
RtpDataCodecs filtered_offered_rtp_data_codecs;
for (const ContentInfo& content : remote_offer.contents()) {
if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
const AudioContentDescription* audio =
content.media_description()->as_audio();
for (const AudioCodec& offered_audio_codec : audio->codecs()) {
if (!FindMatchingCodec<AudioCodec>(audio->codecs(),
filtered_offered_audio_codecs,
offered_audio_codec, nullptr) &&
FindMatchingCodec<AudioCodec>(audio->codecs(), all_audio_codecs_,
offered_audio_codec, nullptr)) {
filtered_offered_audio_codecs.push_back(offered_audio_codec);
}
}
} else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) {
const VideoContentDescription* video =
content.media_description()->as_video();
for (const VideoCodec& offered_video_codec : video->codecs()) {
if (!FindMatchingCodec<VideoCodec>(video->codecs(),
filtered_offered_video_codecs,
offered_video_codec, nullptr) &&
FindMatchingCodec<VideoCodec>(video->codecs(), video_codecs_,
offered_video_codec, nullptr)) {
filtered_offered_video_codecs.push_back(offered_video_codec);
}
}
} else if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) {
const RtpDataContentDescription* data =
content.media_description()->as_rtp_data();
if (data) {
// RTP data. This part is inactive for SCTP data.
for (const RtpDataCodec& offered_rtp_data_codec : data->codecs()) {
if (!FindMatchingCodec<RtpDataCodec>(
data->codecs(), filtered_offered_rtp_data_codecs,
offered_rtp_data_codec, nullptr) &&
FindMatchingCodec<RtpDataCodec>(data->codecs(), rtp_data_codecs_,
offered_rtp_data_codec,
nullptr)) {
filtered_offered_rtp_data_codecs.push_back(offered_rtp_data_codec);
}
}
}
}
}
// Add codecs that are not in the current description but were in
// |remote_offer|.
MergeCodecs<AudioCodec>(filtered_offered_audio_codecs, audio_codecs,
&used_pltypes);
MergeCodecs<VideoCodec>(filtered_offered_video_codecs, video_codecs,
&used_pltypes);
MergeCodecs<DataCodec>(filtered_offered_rtp_data_codecs, rtp_data_codecs,
&used_pltypes);
}
void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
const std::vector<const ContentInfo*>& current_active_contents,
bool extmap_allow_mixed,
RtpHeaderExtensions* offer_audio_extensions,
RtpHeaderExtensions* offer_video_extensions) 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;
// 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)) {
const AudioContentDescription* audio =
content->media_description()->as_audio();
MergeRtpHdrExts(audio->rtp_header_extensions(), offer_audio_extensions,
&all_regular_extensions, &all_encrypted_extensions,
&used_ids);
} else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
const VideoContentDescription* video =
content->media_description()->as_video();
MergeRtpHdrExts(video->rtp_header_extensions(), offer_video_extensions,
&all_regular_extensions, &all_encrypted_extensions,
&used_ids);
}
}
// Add our default RTP header extensions that are not in the current
// description.
MergeRtpHdrExts(audio_rtp_header_extensions(), offer_audio_extensions,
&all_regular_extensions, &all_encrypted_extensions,
&used_ids);
MergeRtpHdrExts(video_rtp_header_extensions(), offer_video_extensions,
&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(offer_audio_extensions,
&all_encrypted_extensions, &used_ids);
AddEncryptedVersionsOfHdrExts(offer_video_extensions,
&all_encrypted_extensions, &used_ids);
}
}
bool MediaSessionDescriptionFactory::AddTransportOffer(
const std::string& content_name,
const TransportOptions& transport_options,
const SessionDescription* current_desc,
SessionDescription* offer_desc,
IceCredentialsIterator* ice_credentials) const {
if (!transport_desc_factory_)
return false;
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 true;
}
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 {
if (!transport_desc_factory_)
return NULL;
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);
}
bool MediaSessionDescriptionFactory::AddTransportAnswer(
const std::string& content_name,
const TransportDescription& transport_desc,
SessionDescription* answer_desc) const {
answer_desc->AddTransportInfo(TransportInfo(content_name, transport_desc));
return true;
}
// |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
//
// acd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from acd->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.
bool 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 {
// 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, supported_audio_codecs);
} 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) {
RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO));
const AudioContentDescription* acd =
current_content->media_description()->as_audio();
for (const AudioCodec& codec : acd->codecs()) {
if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec,
nullptr)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported audio codecs.
AudioCodec found_codec;
for (const AudioCodec& codec : supported_audio_codecs) {
if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs,
codec, &found_codec) &&
!FindMatchingCodec<AudioCodec>(supported_audio_codecs,
filtered_codecs, codec, nullptr)) {
// Use the |found_codec| from |audio_codecs| because it has the
// correctly mapped payload type.
filtered_codecs.push_back(found_codec);
}
}
}
cricket::SecurePolicy sdes_policy =
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
: secure();
std::unique_ptr<AudioContentDescription> audio(new AudioContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedAudioSdesCryptoSuiteNames(session_options.crypto_options,
&crypto_suites);
if (!CreateMediaContentOffer(media_description_options, session_options,
filtered_codecs, sdes_policy,
GetCryptos(current_content), crypto_suites,
audio_rtp_extensions, ssrc_generator_,
current_streams, audio.get())) {
return false;
}
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));
if (!AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials)) {
return false;
}
return true;
}
bool 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 {
cricket::SecurePolicy sdes_policy =
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
: secure();
std::unique_ptr<VideoContentDescription> video(new VideoContentDescription());
std::vector<std::string> crypto_suites;
GetSupportedVideoSdesCryptoSuiteNames(session_options.crypto_options,
&crypto_suites);
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_);
} 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) {
RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO));
const VideoContentDescription* vcd =
current_content->media_description()->as_video();
for (const VideoCodec& codec : vcd->codecs()) {
if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec,
nullptr)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported video codecs.
VideoCodec found_codec;
for (const VideoCodec& codec : video_codecs_) {
if (FindMatchingCodec<VideoCodec>(video_codecs_, video_codecs, codec,
&found_codec) &&
!FindMatchingCodec<VideoCodec>(video_codecs_, filtered_codecs, codec,
nullptr)) {
// Use the |found_codec| from |video_codecs| because it has the
// correctly mapped payload type.
filtered_codecs.push_back(found_codec);
}
}
}
if (session_options.raw_packetization_for_video) {
for (VideoCodec& codec : filtered_codecs) {
if (codec.GetCodecType() == VideoCodec::CODEC_VIDEO) {
codec.packetization = kPacketizationParamRaw;
}
}
}
if (!CreateMediaContentOffer(media_description_options, session_options,
filtered_codecs, sdes_policy,
GetCryptos(current_content), crypto_suites,
video_rtp_extensions, ssrc_generator_,
current_streams, video.get())) {
return false;
}
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));
if (!AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddSctpDataContentForOffer(
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 {
std::unique_ptr<SctpDataContentDescription> data(
new 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);
if (!CreateContentOffer(media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
crypto_suites, RtpHeaderExtensions(), ssrc_generator_,
current_streams, data.get())) {
return false;
}
desc->AddContent(media_description_options.mid, MediaProtocolType::kSctp,
std::move(data));
if (!AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddRtpDataContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
const RtpDataCodecs& rtp_data_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
std::unique_ptr<RtpDataContentDescription> data(
new RtpDataContentDescription());
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;
GetSupportedDataSdesCryptoSuiteNames(session_options.crypto_options,
&crypto_suites);
if (!CreateMediaContentOffer(media_description_options, session_options,
rtp_data_codecs, sdes_policy,
GetCryptos(current_content), crypto_suites,
RtpHeaderExtensions(), ssrc_generator_,
current_streams, data.get())) {
return false;
}
data->set_bandwidth(kDataMaxBandwidth);
SetMediaProtocol(secure_transport, data.get());
desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
media_description_options.stopped, std::move(data));
if (!AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials)) {
return false;
}
return true;
}
bool MediaSessionDescriptionFactory::AddDataContentForOffer(
const MediaDescriptionOptions& media_description_options,
const MediaSessionOptions& session_options,
const ContentInfo* current_content,
const SessionDescription* current_description,
const RtpDataCodecs& rtp_data_codecs,
StreamParamsVec* current_streams,
SessionDescription* desc,
IceCredentialsIterator* ice_credentials) const {
bool is_sctp =
(session_options.data_channel_type == DCT_SCTP ||
session_options.data_channel_type == DCT_DATA_CHANNEL_TRANSPORT_SCTP);
// If the DataChannel type is not specified, use the DataChannel type in
// the current description.
if (session_options.data_channel_type == DCT_NONE && current_content) {
RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_DATA));
is_sctp = (current_content->media_description()->protocol() ==
kMediaProtocolSctp);
}
if (is_sctp) {
return AddSctpDataContentForOffer(
media_description_options, session_options, current_content,
current_description, current_streams, desc, ice_credentials);
} else {
return AddRtpDataContentForOffer(media_description_options, session_options,
current_content, current_description,
rtp_data_codecs, current_streams, 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
//
// acd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from acd->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.
bool 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,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
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) {
return false;
}
// 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, supported_audio_codecs);
} 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) {
RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO));
const AudioContentDescription* acd =
current_content->media_description()->as_audio();
for (const AudioCodec& codec : acd->codecs()) {
if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec,
nullptr)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported audio codecs.
for (const AudioCodec& codec : supported_audio_codecs) {
if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs,
codec, nullptr) &&
!FindMatchingCodec<AudioCodec>(supported_audio_codecs,
filtered_codecs, codec, nullptr)) {
// We should use the local codec with local parameters and the codec id
// would be correctly mapped in |NegotiateCodecs|.
filtered_codecs.push_back(codec);
}
}
}
bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;
std::unique_ptr<AudioContentDescription> audio_answer(
new 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())) {
return false;
}
if (!CreateMediaContentAnswer(
offer_audio_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
audio_rtp_header_extensions(), ssrc_generator_,
enable_encrypted_rtp_header_extensions_, current_streams,
bundle_enabled, audio_answer.get())) {
return false; // Fails the session setup.
}
bool secure = bundle_transport ? bundle_transport->description.secure()
: audio_transport->secure();
bool rejected = media_description_options.stopped ||
offer_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
audio_answer->protocol(), secure);
if (!AddTransportAnswer(media_description_options.mid,
*(audio_transport.get()), answer)) {
return false;
}
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 true;
}
bool 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,
StreamParamsVec* current_streams,
SessionDescription* answer,
IceCredentialsIterator* ice_credentials) const {
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) {
return false;
}
VideoCodecs filtered_codecs;
if (!media_description_options.codec_preferences.empty()) {
filtered_codecs = MatchCodecPreference(
media_description_options.codec_preferences, video_codecs_);
} 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) {
RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_VIDEO));
const VideoContentDescription* vcd =
current_content->media_description()->as_video();
for (const VideoCodec& codec : vcd->codecs()) {
if (FindMatchingCodec<VideoCodec>(vcd->codecs(), video_codecs, codec,
nullptr)) {
filtered_codecs.push_back(codec);
}
}
}
// Add other supported video codecs.
for (const VideoCodec& codec : video_codecs_) {
if (FindMatchingCodec<VideoCodec>(video_codecs_, video_codecs, codec,
nullptr) &&
!FindMatchingCodec<VideoCodec>(video_codecs_, filtered_codecs, codec,
nullptr)) {
// 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.raw_packetization_for_video) {
for (VideoCodec& codec : filtered_codecs) {
if (codec.GetCodecType() == VideoCodec::CODEC_VIDEO) {
codec.packetization = kPacketizationParamRaw;
}
}
}
bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
session_options.bundle_enabled;
std::unique_ptr<VideoContentDescription> video_answer(
new 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())) {
return false;
}
if (!CreateMediaContentAnswer(
offer_video_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
video_rtp_header_extensions(), ssrc_generator_,
enable_encrypted_rtp_header_extensions_, current_streams,
bundle_enabled, video_answer.get())) {
return false; // Failed the sessin setup.
}
bool secure = bundle_transport ? bundle_transport->description.secure()
: video_transport->secure();
bool rejected = media_description_options.stopped ||
offer_content->rejected ||
!IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
video_answer->protocol(), secure);
if (!AddTransportAnswer(media_description_options.mid,
*(video_transport.get()), answer)) {
return false;
}
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 true;
}
bool 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,
const RtpDataCodecs& rtp_data_codecs,
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) {
return false;
}
// 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 = absl::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())) {
return false; // Fails the session setup.
}
// 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 {
// RTP offer
data_answer = absl::make_unique<RtpDataContentDescription>();
const RtpDataContentDescription* offer_data_description =
offer_content->media_description()->as_rtp_data();
RTC_CHECK(offer_data_description);
if (!SetCodecsInAnswer(offer_data_description, rtp_data_codecs,
media_description_options, session_options,
ssrc_generator_, current_streams,
data_answer->as_rtp_data())) {
return false;
}
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())) {
return false; // Fails the session setup.
}
}
bool secure