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webrtc/pc/media_session.cc
Henrik Boström f8187e0a82 [Unified Plan] Support multiple BUNDLE groups. 
In this CL, JsepTransportController and MediaSessionDescriptionFactory
are updated not to assume that there only exists at most a single BUNDLE
group but a list of N groups. This makes it possible to create multiple
BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP.

This makes it possible to have some m= sections in one group and some
other m= sections in another group. For example, you could group all
audio m= sections in one group and all video m= sections in another
group. This enables "send all audio tracks on one transport and all
video tracks on another transport" in Unified Plan. This is something
that was possible in Plan B because all ssrcs in the same m= section
were implicitly bundled together forming a group of audio m= section and
video m= section (even without use of the BUNDLE tag).

PeerConnection will never create multiple BUNDLE groups by default, but
upon setting SDP with multiple BUNDLE groups the PeerConnection will
accept them if configured to accept BUNDLE. This makes it possible to
accept an SFU's BUNDLE offer without having to SDP munge the answer.

C++ unit tests are added. This fix has also been verified manually on:
https://jsfiddle.net/henbos/to89L6ce/43/

Without fix: 0+2 get bundled, 1+3 don't get bundled.
With fix: 0+2 get bundled in first group, 1+3 get bundled in second
group.

Bug: webrtc:10208
Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33838}
2021-04-27 05:53:37 +00:00

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/*
* 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 <memory>
#include <set>
#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 "api/video_codecs/h264_profile_level_id.h"
#include "media/base/codec.h"
#include "media/base/media_constants.h"
#include "media/base/sdp_video_format_utils.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/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"
#include "system_wrappers/include/field_trial.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));
}
}
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 {
// 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_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);
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);
}
}
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::SRTP_AES128_CM_SHA1_80);
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);
}
}
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) {
crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
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);
}
}
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;
}
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.";
}
if (include_flexfec_stream &&
!webrtc::field_trial::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;
}
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;
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|.
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);
}
// 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.
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)) {
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);
// 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);
}
}
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 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,
absl::string_view uri_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 == uri_to_match) {
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.uri, 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);
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;
webrtc::RtpExtension theirs;
if (FindByUriWithEncryptionPreference(
offered_extensions, ours.uri,
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);
}
// Frame descriptors support. If the extension is not present locally, but is
// in the offer, we add it to the list.
webrtc::RtpExtension theirs;
if (!dependency_descriptor_in_local &&
FindByUriWithEncryptionPreference(
offered_extensions, webrtc::RtpExtension::kDependencyDescriptorUri,
enable_encrypted_rtp_header_extensions, &theirs)) {
negotiated_extensions->push_back(theirs);
}
if (!frame_descriptor_in_local &&
FindByUriWithEncryptionPreference(
offered_extensions,
webrtc::RtpExtension::kGenericFrameDescriptorUri00,
enable_encrypted_rtp_header_extensions, &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 &&
FindByUriWithEncryptionPreference(
offered_extensions, webrtc::RtpExtension::kAbsoluteCaptureTimeUri,
enable_encrypted_rtp_header_extensions, &theirs)) {
negotiated_extensions->push_back(theirs);
}
}
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_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());
RtpHeaderExtensions negotiated_rtp_extensions;
NegotiateRtpHeaderExtensions(
local_rtp_extensions, 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::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->GetSupportedVideoSendCodecs(&video_send_codecs_);
channel_manager->GetSupportedVideoReceiveCodecs(&video_recv_codecs_);
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();
}
static void RemoveUnifiedPlanExtensions(RtpHeaderExtensions* extensions) {
RTC_DCHECK(extensions);
extensions->erase(
std::remove_if(extensions->begin(), extensions->end(),
[](auto extension) {
return extension.uri == webrtc::RtpExtension::kMidUri ||
extension.uri == webrtc::RtpExtension::kRidUri ||
extension.uri ==
webrtc::RtpExtension::kRepairedRidUri;
}),
extensions->end());
}
RtpHeaderExtensions
MediaSessionDescriptionFactory::filtered_rtp_header_extensions(
RtpHeaderExtensions extensions) const {
if (!is_unified_plan_) {
RemoveUnifiedPlanExtensions(&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;
GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
&offer_video_codecs);
if (!session_options.vad_enabled) {
// If application doesn't want CN codecs in offer.
StripCNCodecs(&offer_audio_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.
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,
extensions_with_ids.audio,
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,
extensions_with_ids.video,
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,
&current_streams, offer.get(),
&ice_credentials)) {
return nullptr;
}
break;
case MEDIA_TYPE_UNSUPPORTED:
if (!AddUnsupportedContentForOffer(
media_description_options, session_options, current_content,
current_description, 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);
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;
GetCodecsForAnswer(current_active_contents, *offer, &answer_audio_codecs,
&answer_video_codecs);
if (!session_options.vad_enabled) {
// If application doesn't want CN codecs in answer.
StripCNCodecs(&answer_audio_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));
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,
answer_audio_codecs, header_extensions, &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,
answer_video_codecs, header_extensions, &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,
&current_streams, answer.get(), &ice_credentials)) {
return nullptr;
}
break;
case MEDIA_TYPE_UNSUPPORTED:
if (!AddUnsupportedContentForAnswer(
media_description_options, session_options, offer_content,
offer, current_content, current_description, bundle_transport,
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 &&
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())) {
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::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) {
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);
}
}
}
// 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 {
// 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);
// Add our codecs that are not in the current description.
MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes);
MergeCodecs<VideoCodec>(all_video_codecs_, video_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) 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, &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;
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(), all_video_codecs_,
offered_video_codec, nullptr)) {
filtered_offered_video_codecs.push_back(offered_video_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);
}
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)) {
const AudioContentDescription* audio =
content->media_description()->as_audio();
MergeRtpHdrExts(audio->rtp_header_extensions(), &offered_extensions.audio,
&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(), &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;
}
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();
auto audio = std::make_unique<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;
}
// TODO(kron): This function is very similar to AddAudioContentForOffer.
// Refactor to reuse shared code.
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 {
// 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, supported_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 : supported_video_codecs) {
if (FindMatchingCodec<VideoCodec>(supported_video_codecs, video_codecs,
codec, &found_codec) &&
!FindMatchingCodec<VideoCodec>(supported_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;
}
}
}
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);
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::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);
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,
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::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));
if (!AddTransportOffer(media_description_options.mid,
media_description_options.transport_options,
current_description, desc, ice_credentials)) {
return false;
}
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::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& default_audio_rtp_header_extensions,
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;
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())) {
return false;
}
if (!CreateMediaContentAnswer(
offer_audio_description, media_description_options, session_options,
sdes_policy, GetCryptos(current_content),
filtered_rtp_header_extensions(default_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;
}
// TODO(kron): This function is very similar to AddAudioContentForAnswer.
// Refactor to reuse shared code.
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,
const RtpHeaderExtensions& default_video_rtp_header_extensions,
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;
}
// 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, supported_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 : supported_video_codecs) {
if (FindMatchingCodec<VideoCodec>(supported_video_codecs, video_codecs,
codec, nullptr) &&
!FindMatchingCodec<VideoCodec>(supported_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;
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())) {
return false;
}
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())) {
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,
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 = 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())) {
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 {
RTC_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);
if (!AddTransportAnswer(media_description_options.mid,
*(data_transport.get()), answer)) {
return false;
}
answer->AddContent(media_description_options.mid, offer_content->type,
rejected, std::move(data_answer));
return true;
}
bool 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) {
return false;
}
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());
if (!AddTransportAnswer(media_description_options.mid,
*(unsupported_transport.get()), answer)) {
return false;
}
answer->AddContent(media_description_options.mid, offer_content->type,
/*rejected=*/true, std::move(unsupported_answer));
return true;
}
void MediaSessionDescriptionFactory::ComputeAudioCodecsIntersectionAndUnion() {
audio_sendrecv_codecs_.clear();
all_audio_codecs_.clear();
// Compute the audio codecs union.
for (const AudioCodec& send : audio_send_codecs_) {
all_audio_codecs_.push_back(send);
if (!FindMatchingCodec<AudioCodec>(audio_send_codecs_, audio_recv_codecs_,
send, nullptr)) {
// It doesn't make sense to have an RTX codec we support sending but not
// receiving.
RTC_DCHECK(!IsRtxCodec(send));
}
}
for (const AudioCodec& recv : audio_recv_codecs_) {
if (!FindMatchingCodec<AudioCodec>(audio_recv_codecs_, audio_send_codecs_,
recv, nullptr)) {
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);
}
void MediaSessionDescriptionFactory::ComputeVideoCodecsIntersectionAndUnion() {
video_sendrecv_codecs_.clear();
all_video_codecs_.clear();
// Compute the video codecs union.
for (const VideoCodec& send : video_send_codecs_) {
all_video_codecs_.push_back(send);
if (!FindMatchingCodec<VideoCodec>(video_send_codecs_, video_recv_codecs_,
send, nullptr)) {
// TODO(kron): This check is violated by the unit test:
// MediaSessionDescriptionFactoryTest.RtxWithoutApt
// Remove either the test or the check.
// It doesn't make sense to have an RTX codec we support sending but not
// receiving.
// RTC_DCHECK(!IsRtxCodec(send));
}
}
for (const VideoCodec& recv : video_recv_codecs_) {
if (!FindMatchingCodec<VideoCodec>(video_recv_codecs_, video_send_codecs_,
recv, nullptr)) {
all_video_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(video_recv_codecs_, video_send_codecs_,
&video_sendrecv_codecs_, true);
}
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
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