3069 lines
121 KiB
C++
3069 lines
121 KiB
C++
/*
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* Copyright 2004 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/media_session.h"
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#include <algorithm>
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#include <functional>
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#include <map>
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#include <memory>
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#include <set>
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#include <unordered_map>
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#include <utility>
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#include "absl/algorithm/container.h"
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#include "absl/strings/match.h"
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#include "absl/types/optional.h"
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#include "api/crypto_params.h"
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#include "media/base/h264_profile_level_id.h"
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#include "media/base/media_constants.h"
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#include "media/sctp/sctp_transport_internal.h"
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#include "p2p/base/p2p_constants.h"
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#include "pc/channel_manager.h"
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#include "pc/media_protocol_names.h"
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#include "pc/rtp_media_utils.h"
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#include "pc/srtp_filter.h"
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#include "pc/used_ids.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/helpers.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/third_party/base64/base64.h"
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#include "rtc_base/unique_id_generator.h"
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#include "system_wrappers/include/field_trial.h"
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namespace {
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using rtc::UniqueRandomIdGenerator;
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using webrtc::RtpTransceiverDirection;
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const char kInline[] = "inline:";
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void GetSupportedSdesCryptoSuiteNames(
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void (*func)(const webrtc::CryptoOptions&, std::vector<int>*),
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* names) {
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std::vector<int> crypto_suites;
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func(crypto_options, &crypto_suites);
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for (const auto crypto : crypto_suites) {
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names->push_back(rtc::SrtpCryptoSuiteToName(crypto));
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}
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}
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webrtc::RtpExtension RtpExtensionFromCapability(
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const webrtc::RtpHeaderExtensionCapability& capability) {
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return webrtc::RtpExtension(capability.uri,
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capability.preferred_id.value_or(1));
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}
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cricket::RtpHeaderExtensions RtpHeaderExtensionsFromCapabilities(
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const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities) {
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cricket::RtpHeaderExtensions exts;
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for (const auto& capability : capabilities) {
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exts.push_back(RtpExtensionFromCapability(capability));
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}
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return exts;
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}
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std::vector<webrtc::RtpHeaderExtensionCapability>
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UnstoppedRtpHeaderExtensionCapabilities(
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std::vector<webrtc::RtpHeaderExtensionCapability> capabilities) {
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capabilities.erase(
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std::remove_if(
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capabilities.begin(), capabilities.end(),
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[](const webrtc::RtpHeaderExtensionCapability& capability) {
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return capability.direction == RtpTransceiverDirection::kStopped;
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}),
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capabilities.end());
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return capabilities;
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}
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bool IsCapabilityPresent(const webrtc::RtpHeaderExtensionCapability& capability,
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const cricket::RtpHeaderExtensions& extensions) {
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return std::find_if(extensions.begin(), extensions.end(),
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[&capability](const webrtc::RtpExtension& extension) {
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return capability.uri == extension.uri;
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}) != extensions.end();
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}
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cricket::RtpHeaderExtensions UnstoppedOrPresentRtpHeaderExtensions(
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const std::vector<webrtc::RtpHeaderExtensionCapability>& capabilities,
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const cricket::RtpHeaderExtensions& unencrypted,
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const cricket::RtpHeaderExtensions& encrypted) {
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cricket::RtpHeaderExtensions extensions;
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for (const auto& capability : capabilities) {
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if (capability.direction != RtpTransceiverDirection::kStopped ||
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IsCapabilityPresent(capability, unencrypted) ||
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IsCapabilityPresent(capability, encrypted)) {
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extensions.push_back(RtpExtensionFromCapability(capability));
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}
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}
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return extensions;
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}
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} // namespace
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namespace cricket {
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// RTP Profile names
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// http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xml
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// RFC4585
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const char kMediaProtocolAvpf[] = "RTP/AVPF";
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// RFC5124
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const char kMediaProtocolDtlsSavpf[] = "UDP/TLS/RTP/SAVPF";
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// We always generate offers with "UDP/TLS/RTP/SAVPF" when using DTLS-SRTP,
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// but we tolerate "RTP/SAVPF" in offers we receive, for compatibility.
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const char kMediaProtocolSavpf[] = "RTP/SAVPF";
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// Note that the below functions support some protocol strings purely for
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// legacy compatibility, as required by JSEP in Section 5.1.2, Profile Names
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// and Interoperability.
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static bool IsDtlsRtp(const std::string& protocol) {
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// Most-likely values first.
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return protocol == "UDP/TLS/RTP/SAVPF" || protocol == "TCP/TLS/RTP/SAVPF" ||
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protocol == "UDP/TLS/RTP/SAVP" || protocol == "TCP/TLS/RTP/SAVP";
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}
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static bool IsPlainRtp(const std::string& protocol) {
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// Most-likely values first.
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return protocol == "RTP/SAVPF" || protocol == "RTP/AVPF" ||
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protocol == "RTP/SAVP" || protocol == "RTP/AVP";
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}
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static RtpTransceiverDirection NegotiateRtpTransceiverDirection(
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RtpTransceiverDirection offer,
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RtpTransceiverDirection wants) {
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bool offer_send = webrtc::RtpTransceiverDirectionHasSend(offer);
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bool offer_recv = webrtc::RtpTransceiverDirectionHasRecv(offer);
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bool wants_send = webrtc::RtpTransceiverDirectionHasSend(wants);
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bool wants_recv = webrtc::RtpTransceiverDirectionHasRecv(wants);
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return webrtc::RtpTransceiverDirectionFromSendRecv(offer_recv && wants_send,
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offer_send && wants_recv);
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}
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static bool IsMediaContentOfType(const ContentInfo* content,
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MediaType media_type) {
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if (!content || !content->media_description()) {
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return false;
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}
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return content->media_description()->type() == media_type;
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}
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static bool CreateCryptoParams(int tag,
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const std::string& cipher,
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CryptoParams* crypto_out) {
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int key_len;
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int salt_len;
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if (!rtc::GetSrtpKeyAndSaltLengths(rtc::SrtpCryptoSuiteFromName(cipher),
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&key_len, &salt_len)) {
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return false;
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}
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int master_key_len = key_len + salt_len;
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std::string master_key;
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if (!rtc::CreateRandomData(master_key_len, &master_key)) {
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return false;
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}
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RTC_CHECK_EQ(master_key_len, master_key.size());
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std::string key = rtc::Base64::Encode(master_key);
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crypto_out->tag = tag;
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crypto_out->cipher_suite = cipher;
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crypto_out->key_params = kInline;
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crypto_out->key_params += key;
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return true;
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}
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static bool AddCryptoParams(const std::string& cipher_suite,
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CryptoParamsVec* cryptos_out) {
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int size = static_cast<int>(cryptos_out->size());
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cryptos_out->resize(size + 1);
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return CreateCryptoParams(size, cipher_suite, &cryptos_out->at(size));
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}
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void AddMediaCryptos(const CryptoParamsVec& cryptos,
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MediaContentDescription* media) {
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for (const CryptoParams& crypto : cryptos) {
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media->AddCrypto(crypto);
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}
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}
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bool CreateMediaCryptos(const std::vector<std::string>& crypto_suites,
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MediaContentDescription* media) {
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CryptoParamsVec cryptos;
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for (const std::string& crypto_suite : crypto_suites) {
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if (!AddCryptoParams(crypto_suite, &cryptos)) {
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return false;
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}
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}
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AddMediaCryptos(cryptos, media);
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return true;
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}
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const CryptoParamsVec* GetCryptos(const ContentInfo* content) {
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if (!content || !content->media_description()) {
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return nullptr;
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}
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return &content->media_description()->cryptos();
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}
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bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
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const CryptoParams& crypto,
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CryptoParams* crypto_out) {
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auto it = absl::c_find_if(
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cryptos, [&crypto](const CryptoParams& c) { return crypto.Matches(c); });
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if (it == cryptos.end()) {
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return false;
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}
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*crypto_out = *it;
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return true;
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}
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// For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the
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// low overhead.
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void GetSupportedAudioSdesCryptoSuites(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<int>* crypto_suites) {
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if (crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher) {
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crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
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}
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crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
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if (crypto_options.srtp.enable_gcm_crypto_suites) {
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crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
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crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
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}
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}
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void GetSupportedAudioSdesCryptoSuiteNames(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedSdesCryptoSuiteNames(GetSupportedAudioSdesCryptoSuites,
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crypto_options, crypto_suite_names);
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}
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void GetSupportedVideoSdesCryptoSuites(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<int>* crypto_suites) {
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crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
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if (crypto_options.srtp.enable_gcm_crypto_suites) {
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crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
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crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
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}
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}
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void GetSupportedVideoSdesCryptoSuiteNames(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedSdesCryptoSuiteNames(GetSupportedVideoSdesCryptoSuites,
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crypto_options, crypto_suite_names);
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}
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void GetSupportedDataSdesCryptoSuites(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<int>* crypto_suites) {
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crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
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if (crypto_options.srtp.enable_gcm_crypto_suites) {
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crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
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crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
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}
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}
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void GetSupportedDataSdesCryptoSuiteNames(
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const webrtc::CryptoOptions& crypto_options,
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std::vector<std::string>* crypto_suite_names) {
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GetSupportedSdesCryptoSuiteNames(GetSupportedDataSdesCryptoSuites,
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crypto_options, crypto_suite_names);
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}
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// Support any GCM cipher (if enabled through options). For video support only
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// 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated (if enabled) unless
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// bundle is enabled because it is low overhead.
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// Pick the crypto in the list that is supported.
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static bool SelectCrypto(const MediaContentDescription* offer,
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bool bundle,
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const webrtc::CryptoOptions& crypto_options,
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CryptoParams* crypto_out) {
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bool audio = offer->type() == MEDIA_TYPE_AUDIO;
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const CryptoParamsVec& cryptos = offer->cryptos();
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for (const CryptoParams& crypto : cryptos) {
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if ((crypto_options.srtp.enable_gcm_crypto_suites &&
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rtc::IsGcmCryptoSuiteName(crypto.cipher_suite)) ||
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rtc::CS_AES_CM_128_HMAC_SHA1_80 == crypto.cipher_suite ||
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(rtc::CS_AES_CM_128_HMAC_SHA1_32 == crypto.cipher_suite && audio &&
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!bundle && crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher)) {
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return CreateCryptoParams(crypto.tag, crypto.cipher_suite, crypto_out);
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}
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}
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return false;
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}
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// Finds all StreamParams of all media types and attach them to stream_params.
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static StreamParamsVec GetCurrentStreamParams(
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const std::vector<const ContentInfo*>& active_local_contents) {
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StreamParamsVec stream_params;
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for (const ContentInfo* content : active_local_contents) {
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for (const StreamParams& params : content->media_description()->streams()) {
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stream_params.push_back(params);
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}
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}
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return stream_params;
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}
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static StreamParams CreateStreamParamsForNewSenderWithSsrcs(
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const SenderOptions& sender,
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const std::string& rtcp_cname,
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bool include_rtx_streams,
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bool include_flexfec_stream,
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UniqueRandomIdGenerator* ssrc_generator) {
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StreamParams result;
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result.id = sender.track_id;
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// TODO(brandtr): Update when we support multistream protection.
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if (include_flexfec_stream && sender.num_sim_layers > 1) {
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include_flexfec_stream = false;
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RTC_LOG(LS_WARNING)
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<< "Our FlexFEC implementation only supports protecting "
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"a single media streams. This session has multiple "
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"media streams however, so no FlexFEC SSRC will be generated.";
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}
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if (include_flexfec_stream &&
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!webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03")) {
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include_flexfec_stream = false;
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RTC_LOG(LS_WARNING)
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<< "WebRTC-FlexFEC trial is not enabled, not sending FlexFEC";
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}
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result.GenerateSsrcs(sender.num_sim_layers, include_rtx_streams,
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include_flexfec_stream, ssrc_generator);
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result.cname = rtcp_cname;
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result.set_stream_ids(sender.stream_ids);
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return result;
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}
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static bool ValidateSimulcastLayers(
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const std::vector<RidDescription>& rids,
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const SimulcastLayerList& simulcast_layers) {
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return absl::c_all_of(
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simulcast_layers.GetAllLayers(), [&rids](const SimulcastLayer& layer) {
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return absl::c_any_of(rids, [&layer](const RidDescription& rid) {
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return rid.rid == layer.rid;
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});
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});
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}
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static StreamParams CreateStreamParamsForNewSenderWithRids(
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const SenderOptions& sender,
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const std::string& rtcp_cname) {
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RTC_DCHECK(!sender.rids.empty());
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RTC_DCHECK_EQ(sender.num_sim_layers, 0)
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<< "RIDs are the compliant way to indicate simulcast.";
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RTC_DCHECK(ValidateSimulcastLayers(sender.rids, sender.simulcast_layers));
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StreamParams result;
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result.id = sender.track_id;
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result.cname = rtcp_cname;
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result.set_stream_ids(sender.stream_ids);
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// More than one rid should be signaled.
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if (sender.rids.size() > 1) {
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result.set_rids(sender.rids);
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}
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return result;
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}
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// Adds SimulcastDescription if indicated by the media description options.
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// MediaContentDescription should already be set up with the send rids.
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static void AddSimulcastToMediaDescription(
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const MediaDescriptionOptions& media_description_options,
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MediaContentDescription* description) {
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RTC_DCHECK(description);
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// Check if we are using RIDs in this scenario.
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if (absl::c_all_of(description->streams(), [](const StreamParams& params) {
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return !params.has_rids();
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})) {
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return;
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}
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RTC_DCHECK_EQ(1, description->streams().size())
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<< "RIDs are only supported in Unified Plan semantics.";
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RTC_DCHECK_EQ(1, media_description_options.sender_options.size());
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RTC_DCHECK(description->type() == MediaType::MEDIA_TYPE_AUDIO ||
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description->type() == MediaType::MEDIA_TYPE_VIDEO);
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// One RID or less indicates that simulcast is not needed.
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if (description->streams()[0].rids().size() <= 1) {
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return;
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}
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// Only negotiate the send layers.
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SimulcastDescription simulcast;
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simulcast.send_layers() =
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media_description_options.sender_options[0].simulcast_layers;
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description->set_simulcast_description(simulcast);
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}
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// Adds a StreamParams for each SenderOptions in |sender_options| to
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// content_description.
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// |current_params| - All currently known StreamParams of any media type.
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template <class C>
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static bool AddStreamParams(
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const std::vector<SenderOptions>& sender_options,
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const std::string& rtcp_cname,
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UniqueRandomIdGenerator* ssrc_generator,
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StreamParamsVec* current_streams,
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MediaContentDescriptionImpl<C>* content_description) {
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// SCTP streams are not negotiated using SDP/ContentDescriptions.
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if (IsSctpProtocol(content_description->protocol())) {
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return true;
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}
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const bool include_rtx_streams =
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ContainsRtxCodec(content_description->codecs());
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const bool include_flexfec_stream =
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ContainsFlexfecCodec(content_description->codecs());
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for (const SenderOptions& sender : sender_options) {
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// groupid is empty for StreamParams generated using
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// MediaSessionDescriptionFactory.
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StreamParams* param =
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GetStreamByIds(*current_streams, "" /*group_id*/, sender.track_id);
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if (!param) {
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// This is a new sender.
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StreamParams stream_param =
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sender.rids.empty()
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?
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// Signal SSRCs and legacy simulcast (if requested).
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CreateStreamParamsForNewSenderWithSsrcs(
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sender, rtcp_cname, include_rtx_streams,
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include_flexfec_stream, ssrc_generator)
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:
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// Signal RIDs and spec-compliant simulcast (if requested).
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CreateStreamParamsForNewSenderWithRids(sender, rtcp_cname);
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content_description->AddStream(stream_param);
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// Store the new StreamParams in current_streams.
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// This is necessary so that we can use the CNAME for other media types.
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current_streams->push_back(stream_param);
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} else {
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// Use existing generated SSRCs/groups, but update the sync_label if
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// necessary. This may be needed if a MediaStreamTrack was moved from one
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// MediaStream to another.
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param->set_stream_ids(sender.stream_ids);
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content_description->AddStream(*param);
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}
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}
|
|
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);
|
|
}
|
|
if (absl::EqualsIgnoreCase(ours.name, kH264CodecName)) {
|
|
webrtc::H264::GenerateProfileLevelIdForAnswer(
|
|
ours.params, theirs.params, &negotiated.params);
|
|
}
|
|
negotiated.id = theirs.id;
|
|
negotiated.name = theirs.name;
|
|
negotiated_codecs->push_back(std::move(negotiated));
|
|
}
|
|
}
|
|
if (keep_offer_order) {
|
|
// RFC3264: Although the answerer MAY list the formats in their desired
|
|
// order of preference, it is RECOMMENDED that unless there is a
|
|
// specific reason, the answerer list formats in the same relative order
|
|
// they were present in the offer.
|
|
// This can be skipped when the transceiver has any codec preferences.
|
|
std::unordered_map<int, int> payload_type_preferences;
|
|
int preference = static_cast<int>(offered_codecs.size() + 1);
|
|
for (const C& codec : offered_codecs) {
|
|
payload_type_preferences[codec.id] = preference--;
|
|
}
|
|
absl::c_sort(*negotiated_codecs, [&payload_type_preferences](const C& a,
|
|
const C& b) {
|
|
return payload_type_preferences[a.id] > payload_type_preferences[b.id];
|
|
});
|
|
}
|
|
}
|
|
|
|
// Finds a codec in |codecs2| that matches |codec_to_match|, which is
|
|
// a member of |codecs1|. If |codec_to_match| is an RTX codec, both
|
|
// the codecs themselves and their associated codecs must match.
|
|
template <class C>
|
|
static bool FindMatchingCodec(const std::vector<C>& codecs1,
|
|
const std::vector<C>& codecs2,
|
|
const C& codec_to_match,
|
|
C* found_codec) {
|
|
// |codec_to_match| should be a member of |codecs1|, in order to look up RTX
|
|
// codecs' associated codecs correctly. If not, that's a programming error.
|
|
RTC_DCHECK(absl::c_any_of(codecs1, [&codec_to_match](const C& codec) {
|
|
return &codec == &codec_to_match;
|
|
}));
|
|
for (const C& potential_match : codecs2) {
|
|
if (potential_match.Matches(codec_to_match)) {
|
|
if (IsRtxCodec(codec_to_match)) {
|
|
int apt_value_1 = 0;
|
|
int apt_value_2 = 0;
|
|
if (!codec_to_match.GetParam(kCodecParamAssociatedPayloadType,
|
|
&apt_value_1) ||
|
|
!potential_match.GetParam(kCodecParamAssociatedPayloadType,
|
|
&apt_value_2)) {
|
|
RTC_LOG(LS_WARNING) << "RTX missing associated payload type.";
|
|
continue;
|
|
}
|
|
if (!ReferencedCodecsMatch(codecs1, apt_value_1, codecs2,
|
|
apt_value_2)) {
|
|
continue;
|
|
}
|
|
}
|
|
if (found_codec) {
|
|
*found_codec = potential_match;
|
|
}
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
// Find the codec in |codec_list| that |rtx_codec| is associated with.
|
|
template <class C>
|
|
static const C* GetAssociatedCodec(const std::vector<C>& codec_list,
|
|
const C& rtx_codec) {
|
|
std::string associated_pt_str;
|
|
if (!rtx_codec.GetParam(kCodecParamAssociatedPayloadType,
|
|
&associated_pt_str)) {
|
|
RTC_LOG(LS_WARNING) << "RTX codec " << rtx_codec.name
|
|
<< " is missing an associated payload type.";
|
|
return nullptr;
|
|
}
|
|
|
|
int associated_pt;
|
|
if (!rtc::FromString(associated_pt_str, &associated_pt)) {
|
|
RTC_LOG(LS_WARNING) << "Couldn't convert payload type " << associated_pt_str
|
|
<< " of RTX codec " << rtx_codec.name
|
|
<< " to an integer.";
|
|
return nullptr;
|
|
}
|
|
|
|
// Find the associated reference codec for the reference RTX codec.
|
|
const C* associated_codec = FindCodecById(codec_list, associated_pt);
|
|
if (!associated_codec) {
|
|
RTC_LOG(LS_WARNING) << "Couldn't find associated codec with payload type "
|
|
<< associated_pt << " for RTX codec " << rtx_codec.name
|
|
<< ".";
|
|
}
|
|
return associated_codec;
|
|
}
|
|
|
|
// Adds all codecs from |reference_codecs| to |offered_codecs| that don't
|
|
// already exist in |offered_codecs| and ensure the payload types don't
|
|
// collide.
|
|
template <class C>
|
|
static void MergeCodecs(const std::vector<C>& reference_codecs,
|
|
std::vector<C>* offered_codecs,
|
|
UsedPayloadTypes* used_pltypes) {
|
|
// Add all new codecs that are not RTX codecs.
|
|
for (const C& reference_codec : reference_codecs) {
|
|
if (!IsRtxCodec(reference_codec) &&
|
|
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
|
|
reference_codec, nullptr)) {
|
|
C codec = reference_codec;
|
|
used_pltypes->FindAndSetIdUsed(&codec);
|
|
offered_codecs->push_back(codec);
|
|
}
|
|
}
|
|
|
|
// Add all new RTX codecs.
|
|
for (const C& reference_codec : reference_codecs) {
|
|
if (IsRtxCodec(reference_codec) &&
|
|
!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
|
|
reference_codec, nullptr)) {
|
|
C rtx_codec = reference_codec;
|
|
const C* associated_codec =
|
|
GetAssociatedCodec(reference_codecs, rtx_codec);
|
|
if (!associated_codec) {
|
|
continue;
|
|
}
|
|
// Find a codec in the offered list that matches the reference codec.
|
|
// Its payload type may be different than the reference codec.
|
|
C matching_codec;
|
|
if (!FindMatchingCodec<C>(reference_codecs, *offered_codecs,
|
|
*associated_codec, &matching_codec)) {
|
|
RTC_LOG(LS_WARNING)
|
|
<< "Couldn't find matching " << associated_codec->name << " codec.";
|
|
continue;
|
|
}
|
|
|
|
rtx_codec.params[kCodecParamAssociatedPayloadType] =
|
|
rtc::ToString(matching_codec.id);
|
|
used_pltypes->FindAndSetIdUsed(&rtx_codec);
|
|
offered_codecs->push_back(rtx_codec);
|
|
}
|
|
}
|
|
}
|
|
|
|
template <typename Codecs>
|
|
static Codecs MatchCodecPreference(
|
|
const std::vector<webrtc::RtpCodecCapability>& codec_preferences,
|
|
const Codecs& codecs) {
|
|
Codecs filtered_codecs;
|
|
std::set<std::string> kept_codecs_ids;
|
|
bool want_rtx = false;
|
|
|
|
for (const auto& codec_preference : codec_preferences) {
|
|
auto found_codec = absl::c_find_if(
|
|
codecs, [&codec_preference](const typename Codecs::value_type& codec) {
|
|
webrtc::RtpCodecParameters codec_parameters =
|
|
codec.ToCodecParameters();
|
|
return codec_parameters.name == codec_preference.name &&
|
|
codec_parameters.kind == codec_preference.kind &&
|
|
codec_parameters.num_channels ==
|
|
codec_preference.num_channels &&
|
|
codec_parameters.clock_rate == codec_preference.clock_rate &&
|
|
codec_parameters.parameters == codec_preference.parameters;
|
|
});
|
|
|
|
if (found_codec != codecs.end()) {
|
|
filtered_codecs.push_back(*found_codec);
|
|
kept_codecs_ids.insert(std::to_string(found_codec->id));
|
|
} else if (IsRtxCodec(codec_preference)) {
|
|
want_rtx = true;
|
|
}
|
|
}
|
|
|
|
if (want_rtx) {
|
|
for (const auto& codec : codecs) {
|
|
if (IsRtxCodec(codec)) {
|
|
const auto apt =
|
|
codec.params.find(cricket::kCodecParamAssociatedPayloadType);
|
|
if (apt != codec.params.end() &&
|
|
kept_codecs_ids.count(apt->second) > 0) {
|
|
filtered_codecs.push_back(codec);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
return filtered_codecs;
|
|
}
|
|
|
|
static bool FindByUriAndEncryption(const RtpHeaderExtensions& extensions,
|
|
const webrtc::RtpExtension& ext_to_match,
|
|
webrtc::RtpExtension* found_extension) {
|
|
auto it = absl::c_find_if(
|
|
extensions, [&ext_to_match](const webrtc::RtpExtension& extension) {
|
|
// We assume that all URIs are given in a canonical
|
|
// format.
|
|
return extension.uri == ext_to_match.uri &&
|
|
extension.encrypt == ext_to_match.encrypt;
|
|
});
|
|
if (it == extensions.end()) {
|
|
return false;
|
|
}
|
|
if (found_extension) {
|
|
*found_extension = *it;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
static bool FindByUri(const RtpHeaderExtensions& extensions,
|
|
const webrtc::RtpExtension& ext_to_match,
|
|
webrtc::RtpExtension* found_extension) {
|
|
// We assume that all URIs are given in a canonical format.
|
|
const webrtc::RtpExtension* found =
|
|
webrtc::RtpExtension::FindHeaderExtensionByUri(extensions,
|
|
ext_to_match.uri);
|
|
if (!found) {
|
|
return false;
|
|
}
|
|
if (found_extension) {
|
|
*found_extension = *found;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
static bool FindByUriWithEncryptionPreference(
|
|
const RtpHeaderExtensions& extensions,
|
|
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 - ¤t_description->contents()[0];
|
|
|
|
if (current_description->transport_infos().size() <= msection_index) {
|
|
return false;
|
|
}
|
|
|
|
return current_description->transport_infos()[msection_index]
|
|
.description.secure();
|
|
}
|
|
|
|
void MediaDescriptionOptions::AddAudioSender(
|
|
const std::string& track_id,
|
|
const std::vector<std::string>& stream_ids) {
|
|
RTC_DCHECK(type == MEDIA_TYPE_AUDIO);
|
|
AddSenderInternal(track_id, stream_ids, {}, SimulcastLayerList(), 1);
|
|
}
|
|
|
|
void MediaDescriptionOptions::AddVideoSender(
|
|
const std::string& track_id,
|
|
const std::vector<std::string>& stream_ids,
|
|
const std::vector<RidDescription>& rids,
|
|
const SimulcastLayerList& simulcast_layers,
|
|
int num_sim_layers) {
|
|
RTC_DCHECK(type == MEDIA_TYPE_VIDEO);
|
|
RTC_DCHECK(rids.empty() || num_sim_layers == 0)
|
|
<< "RIDs are the compliant way to indicate simulcast.";
|
|
RTC_DCHECK(ValidateSimulcastLayers(rids, simulcast_layers));
|
|
AddSenderInternal(track_id, stream_ids, rids, simulcast_layers,
|
|
num_sim_layers);
|
|
}
|
|
|
|
void MediaDescriptionOptions::AddRtpDataChannel(const std::string& track_id,
|
|
const std::string& stream_id) {
|
|
RTC_DCHECK(type == MEDIA_TYPE_DATA);
|
|
// TODO(steveanton): Is it the case that RtpDataChannel will never have more
|
|
// than one stream?
|
|
AddSenderInternal(track_id, {stream_id}, {}, SimulcastLayerList(), 1);
|
|
}
|
|
|
|
void MediaDescriptionOptions::AddSenderInternal(
|
|
const std::string& track_id,
|
|
const std::vector<std::string>& stream_ids,
|
|
const std::vector<RidDescription>& rids,
|
|
const SimulcastLayerList& simulcast_layers,
|
|
int num_sim_layers) {
|
|
// TODO(steveanton): Support any number of stream ids.
|
|
RTC_CHECK(stream_ids.size() == 1U);
|
|
SenderOptions options;
|
|
options.track_id = track_id;
|
|
options.stream_ids = stream_ids;
|
|
options.simulcast_layers = simulcast_layers;
|
|
options.rids = rids;
|
|
options.num_sim_layers = num_sim_layers;
|
|
sender_options.push_back(options);
|
|
}
|
|
|
|
bool MediaSessionOptions::HasMediaDescription(MediaType type) const {
|
|
return absl::c_any_of(
|
|
media_description_options,
|
|
[type](const MediaDescriptionOptions& t) { return t.type == type; });
|
|
}
|
|
|
|
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
|
|
const TransportDescriptionFactory* transport_desc_factory,
|
|
rtc::UniqueRandomIdGenerator* ssrc_generator)
|
|
: ssrc_generator_(ssrc_generator),
|
|
transport_desc_factory_(transport_desc_factory) {
|
|
RTC_DCHECK(ssrc_generator_);
|
|
}
|
|
|
|
MediaSessionDescriptionFactory::MediaSessionDescriptionFactory(
|
|
ChannelManager* channel_manager,
|
|
const TransportDescriptionFactory* transport_desc_factory,
|
|
rtc::UniqueRandomIdGenerator* ssrc_generator)
|
|
: MediaSessionDescriptionFactory(transport_desc_factory, ssrc_generator) {
|
|
channel_manager->GetSupportedAudioSendCodecs(&audio_send_codecs_);
|
|
channel_manager->GetSupportedAudioReceiveCodecs(&audio_recv_codecs_);
|
|
channel_manager->GetSupportedVideoSendCodecs(&video_send_codecs_);
|
|
channel_manager->GetSupportedVideoReceiveCodecs(&video_recv_codecs_);
|
|
channel_manager->GetSupportedDataCodecs(&rtp_data_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;
|
|
RtpDataCodecs offer_rtp_data_codecs;
|
|
GetCodecsForOffer(current_active_contents, &offer_audio_codecs,
|
|
&offer_video_codecs, &offer_rtp_data_codecs);
|
|
if (!session_options.vad_enabled) {
|
|
// If application doesn't want CN codecs in offer.
|
|
StripCNCodecs(&offer_audio_codecs);
|
|
}
|
|
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 = ¤t_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, ¤t_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, ¤t_streams,
|
|
offer.get(), &ice_credentials)) {
|
|
return nullptr;
|
|
}
|
|
break;
|
|
case MEDIA_TYPE_DATA:
|
|
if (!AddDataContentForOffer(media_description_options, session_options,
|
|
current_content, current_description,
|
|
offer_rtp_data_codecs, ¤t_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;
|
|
RtpDataCodecs answer_rtp_data_codecs;
|
|
GetCodecsForAnswer(current_active_contents, *offer, &answer_audio_codecs,
|
|
&answer_video_codecs, &answer_rtp_data_codecs);
|
|
|
|
if (!session_options.vad_enabled) {
|
|
// If application doesn't want CN codecs in answer.
|
|
StripCNCodecs(&answer_audio_codecs);
|
|
}
|
|
|
|
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.
|
|
const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE);
|
|
ContentGroup answer_bundle(GROUP_TYPE_BUNDLE);
|
|
// Transport info shared by the bundle group.
|
|
std::unique_ptr<TransportInfo> bundle_transport;
|
|
|
|
answer->set_extmap_allow_mixed(offer->extmap_allow_mixed());
|
|
|
|
// Iterate through the media description options, matching with existing
|
|
// media descriptions in |current_description|.
|
|
size_t msection_index = 0;
|
|
for (const MediaDescriptionOptions& media_description_options :
|
|
session_options.media_description_options) {
|
|
const ContentInfo* offer_content = &offer->contents()[msection_index];
|
|
// Media types and MIDs must match between the remote offer and the
|
|
// MediaDescriptionOptions.
|
|
RTC_DCHECK(
|
|
IsMediaContentOfType(offer_content, media_description_options.type));
|
|
RTC_DCHECK(media_description_options.mid == offer_content->name);
|
|
const ContentInfo* current_content = nullptr;
|
|
if (current_description &&
|
|
msection_index < current_description->contents().size()) {
|
|
current_content = ¤t_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.get(), answer_audio_codecs, header_extensions,
|
|
¤t_streams, answer.get(), &ice_credentials)) {
|
|
return nullptr;
|
|
}
|
|
break;
|
|
case MEDIA_TYPE_VIDEO:
|
|
if (!AddVideoContentForAnswer(
|
|
media_description_options, session_options, offer_content,
|
|
offer, current_content, current_description,
|
|
bundle_transport.get(), answer_video_codecs, header_extensions,
|
|
¤t_streams, answer.get(), &ice_credentials)) {
|
|
return nullptr;
|
|
}
|
|
break;
|
|
case MEDIA_TYPE_DATA:
|
|
if (!AddDataContentForAnswer(
|
|
media_description_options, session_options, offer_content,
|
|
offer, current_content, current_description,
|
|
bundle_transport.get(), answer_rtp_data_codecs,
|
|
¤t_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.get(), answer.get(), &ice_credentials)) {
|
|
return nullptr;
|
|
}
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
}
|
|
++msection_index;
|
|
// See if we can add the newly generated m= section to the BUNDLE group in
|
|
// the answer.
|
|
ContentInfo& added = answer->contents().back();
|
|
if (!added.rejected && session_options.bundle_enabled && offer_bundle &&
|
|
offer_bundle->HasContentName(added.name)) {
|
|
answer_bundle.AddContentName(added.name);
|
|
bundle_transport.reset(
|
|
new TransportInfo(*answer->GetTransportInfoByName(added.name)));
|
|
}
|
|
}
|
|
|
|
// If a BUNDLE group was offered, put a BUNDLE group in the answer even if
|
|
// it's empty. RFC5888 says:
|
|
//
|
|
// A SIP entity that receives an offer that contains an "a=group" line
|
|
// with semantics that are understood MUST return an answer that
|
|
// contains an "a=group" line with the same semantics.
|
|
if (offer_bundle) {
|
|
answer->AddGroup(answer_bundle);
|
|
}
|
|
|
|
if (answer_bundle.FirstContentName()) {
|
|
// Share the same ICE credentials and crypto params across all contents,
|
|
// as BUNDLE requires.
|
|
if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "CreateAnswer failed to UpdateTransportInfoForBundle.";
|
|
return NULL;
|
|
}
|
|
|
|
if (!UpdateCryptoParamsForBundle(answer_bundle, answer.get())) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "CreateAnswer failed to UpdateCryptoParamsForBundle.";
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
// The following determines how to signal MSIDs to ensure compatibility with
|
|
// older endpoints (in particular, older Plan B endpoints).
|
|
if (is_unified_plan_) {
|
|
// Unified Plan needs to look at what the offer included to find the most
|
|
// compatible answer.
|
|
if (offer->msid_signaling() == 0) {
|
|
// We end up here in one of three cases:
|
|
// 1. An empty offer. We'll reply with an empty answer so it doesn't
|
|
// matter what we pick here.
|
|
// 2. A data channel only offer. We won't add any MSIDs to the answer so
|
|
// it also doesn't matter what we pick here.
|
|
// 3. Media that's either sendonly or inactive from the remote endpoint.
|
|
// We don't have any information to say whether the endpoint is Plan B
|
|
// or Unified Plan, so be conservative and send both.
|
|
answer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
|
|
cricket::kMsidSignalingSsrcAttribute);
|
|
} else if (offer->msid_signaling() ==
|
|
(cricket::kMsidSignalingMediaSection |
|
|
cricket::kMsidSignalingSsrcAttribute)) {
|
|
// If both a=msid and a=ssrc MSID signaling methods were used, we're
|
|
// probably talking to a Unified Plan endpoint so respond with just
|
|
// a=msid.
|
|
answer->set_msid_signaling(cricket::kMsidSignalingMediaSection);
|
|
} else {
|
|
// Otherwise, it's clear which method the offerer is using so repeat that
|
|
// back to them.
|
|
answer->set_msid_signaling(offer->msid_signaling());
|
|
}
|
|
} else {
|
|
// Plan B always signals MSID using a=ssrc lines.
|
|
answer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
|
|
}
|
|
|
|
return answer;
|
|
}
|
|
|
|
const AudioCodecs& MediaSessionDescriptionFactory::GetAudioCodecsForOffer(
|
|
const RtpTransceiverDirection& direction) const {
|
|
switch (direction) {
|
|
// If stream is inactive - generate list as if sendrecv.
|
|
case RtpTransceiverDirection::kSendRecv:
|
|
case RtpTransceiverDirection::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,
|
|
RtpDataCodecs* rtp_data_codecs,
|
|
UsedPayloadTypes* used_pltypes) {
|
|
for (const ContentInfo* content : current_active_contents) {
|
|
if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
|
|
const AudioContentDescription* audio =
|
|
content->media_description()->as_audio();
|
|
MergeCodecs<AudioCodec>(audio->codecs(), audio_codecs, used_pltypes);
|
|
} else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
|
|
const VideoContentDescription* video =
|
|
content->media_description()->as_video();
|
|
MergeCodecs<VideoCodec>(video->codecs(), video_codecs, used_pltypes);
|
|
} else if (IsMediaContentOfType(content, MEDIA_TYPE_DATA)) {
|
|
const RtpDataContentDescription* data =
|
|
content->media_description()->as_rtp_data();
|
|
if (data) {
|
|
// Only relevant for RTP datachannels
|
|
MergeCodecs<RtpDataCodec>(data->codecs(), rtp_data_codecs,
|
|
used_pltypes);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Getting codecs for an offer involves these steps:
|
|
//
|
|
// 1. Construct payload type -> codec mappings for current description.
|
|
// 2. Add any reference codecs that weren't already present
|
|
// 3. For each individual media description (m= section), filter codecs based
|
|
// on the directional attribute (happens in another method).
|
|
void MediaSessionDescriptionFactory::GetCodecsForOffer(
|
|
const std::vector<const ContentInfo*>& current_active_contents,
|
|
AudioCodecs* audio_codecs,
|
|
VideoCodecs* video_codecs,
|
|
RtpDataCodecs* rtp_data_codecs) const {
|
|
// First - get all codecs from the current description if the media type
|
|
// is used. Add them to |used_pltypes| so the payload type is not reused if a
|
|
// new media type is added.
|
|
UsedPayloadTypes used_pltypes;
|
|
MergeCodecsFromDescription(current_active_contents, audio_codecs,
|
|
video_codecs, rtp_data_codecs, &used_pltypes);
|
|
|
|
// Add our codecs that are not in the current description.
|
|
MergeCodecs<AudioCodec>(all_audio_codecs_, audio_codecs, &used_pltypes);
|
|
MergeCodecs<VideoCodec>(all_video_codecs_, video_codecs, &used_pltypes);
|
|
MergeCodecs<DataCodec>(rtp_data_codecs_, rtp_data_codecs, &used_pltypes);
|
|
}
|
|
|
|
// Getting codecs for an answer involves these steps:
|
|
//
|
|
// 1. Construct payload type -> codec mappings for current description.
|
|
// 2. Add any codecs from the offer that weren't already present.
|
|
// 3. Add any remaining codecs that weren't already present.
|
|
// 4. For each individual media description (m= section), filter codecs based
|
|
// on the directional attribute (happens in another method).
|
|
void MediaSessionDescriptionFactory::GetCodecsForAnswer(
|
|
const std::vector<const ContentInfo*>& current_active_contents,
|
|
const SessionDescription& remote_offer,
|
|
AudioCodecs* audio_codecs,
|
|
VideoCodecs* video_codecs,
|
|
RtpDataCodecs* rtp_data_codecs) const {
|
|
// First - get all codecs from the current description if the media type
|
|
// is used. Add them to |used_pltypes| so the payload type is not reused if a
|
|
// new media type is added.
|
|
UsedPayloadTypes used_pltypes;
|
|
MergeCodecsFromDescription(current_active_contents, audio_codecs,
|
|
video_codecs, rtp_data_codecs, &used_pltypes);
|
|
|
|
// Second - filter out codecs that we don't support at all and should ignore.
|
|
AudioCodecs filtered_offered_audio_codecs;
|
|
VideoCodecs filtered_offered_video_codecs;
|
|
RtpDataCodecs filtered_offered_rtp_data_codecs;
|
|
for (const ContentInfo& content : remote_offer.contents()) {
|
|
if (IsMediaContentOfType(&content, MEDIA_TYPE_AUDIO)) {
|
|
const AudioContentDescription* audio =
|
|
content.media_description()->as_audio();
|
|
for (const AudioCodec& offered_audio_codec : audio->codecs()) {
|
|
if (!FindMatchingCodec<AudioCodec>(audio->codecs(),
|
|
filtered_offered_audio_codecs,
|
|
offered_audio_codec, nullptr) &&
|
|
FindMatchingCodec<AudioCodec>(audio->codecs(), all_audio_codecs_,
|
|
offered_audio_codec, nullptr)) {
|
|
filtered_offered_audio_codecs.push_back(offered_audio_codec);
|
|
}
|
|
}
|
|
} else if (IsMediaContentOfType(&content, MEDIA_TYPE_VIDEO)) {
|
|
const VideoContentDescription* video =
|
|
content.media_description()->as_video();
|
|
for (const VideoCodec& offered_video_codec : video->codecs()) {
|
|
if (!FindMatchingCodec<VideoCodec>(video->codecs(),
|
|
filtered_offered_video_codecs,
|
|
offered_video_codec, nullptr) &&
|
|
FindMatchingCodec<VideoCodec>(video->codecs(), all_video_codecs_,
|
|
offered_video_codec, nullptr)) {
|
|
filtered_offered_video_codecs.push_back(offered_video_codec);
|
|
}
|
|
}
|
|
} else if (IsMediaContentOfType(&content, MEDIA_TYPE_DATA)) {
|
|
const RtpDataContentDescription* data =
|
|
content.media_description()->as_rtp_data();
|
|
if (data) {
|
|
// RTP data. This part is inactive for SCTP data.
|
|
for (const RtpDataCodec& offered_rtp_data_codec : data->codecs()) {
|
|
if (!FindMatchingCodec<RtpDataCodec>(
|
|
data->codecs(), filtered_offered_rtp_data_codecs,
|
|
offered_rtp_data_codec, nullptr) &&
|
|
FindMatchingCodec<RtpDataCodec>(data->codecs(), rtp_data_codecs_,
|
|
offered_rtp_data_codec,
|
|
nullptr)) {
|
|
filtered_offered_rtp_data_codecs.push_back(offered_rtp_data_codec);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Add codecs that are not in the current description but were in
|
|
// |remote_offer|.
|
|
MergeCodecs<AudioCodec>(filtered_offered_audio_codecs, audio_codecs,
|
|
&used_pltypes);
|
|
MergeCodecs<VideoCodec>(filtered_offered_video_codecs, video_codecs,
|
|
&used_pltypes);
|
|
MergeCodecs<DataCodec>(filtered_offered_rtp_data_codecs, rtp_data_codecs,
|
|
&used_pltypes);
|
|
}
|
|
|
|
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::AddSctpDataContentForOffer(
|
|
const MediaDescriptionOptions& media_description_options,
|
|
const MediaSessionOptions& session_options,
|
|
const ContentInfo* current_content,
|
|
const SessionDescription* current_description,
|
|
StreamParamsVec* current_streams,
|
|
SessionDescription* desc,
|
|
IceCredentialsIterator* ice_credentials) const {
|
|
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::AddRtpDataContentForOffer(
|
|
const MediaDescriptionOptions& media_description_options,
|
|
const MediaSessionOptions& session_options,
|
|
const ContentInfo* current_content,
|
|
const SessionDescription* current_description,
|
|
const RtpDataCodecs& rtp_data_codecs,
|
|
StreamParamsVec* current_streams,
|
|
SessionDescription* desc,
|
|
IceCredentialsIterator* ice_credentials) const {
|
|
auto data = std::make_unique<RtpDataContentDescription>();
|
|
bool secure_transport = (transport_desc_factory_->secure() != SEC_DISABLED);
|
|
|
|
cricket::SecurePolicy sdes_policy =
|
|
IsDtlsActive(current_content, current_description) ? cricket::SEC_DISABLED
|
|
: secure();
|
|
std::vector<std::string> crypto_suites;
|
|
GetSupportedDataSdesCryptoSuiteNames(session_options.crypto_options,
|
|
&crypto_suites);
|
|
if (!CreateMediaContentOffer(media_description_options, session_options,
|
|
rtp_data_codecs, sdes_policy,
|
|
GetCryptos(current_content), crypto_suites,
|
|
RtpHeaderExtensions(), ssrc_generator_,
|
|
current_streams, data.get())) {
|
|
return false;
|
|
}
|
|
|
|
data->set_bandwidth(kRtpDataMaxBandwidth);
|
|
SetMediaProtocol(secure_transport, data.get());
|
|
desc->AddContent(media_description_options.mid, MediaProtocolType::kRtp,
|
|
media_description_options.stopped, std::move(data));
|
|
if (!AddTransportOffer(media_description_options.mid,
|
|
media_description_options.transport_options,
|
|
current_description, desc, ice_credentials)) {
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool MediaSessionDescriptionFactory::AddDataContentForOffer(
|
|
const MediaDescriptionOptions& media_description_options,
|
|
const MediaSessionOptions& session_options,
|
|
const ContentInfo* current_content,
|
|
const SessionDescription* current_description,
|
|
const RtpDataCodecs& rtp_data_codecs,
|
|
StreamParamsVec* current_streams,
|
|
SessionDescription* desc,
|
|
IceCredentialsIterator* ice_credentials) const {
|
|
bool is_sctp = (session_options.data_channel_type == DCT_SCTP);
|
|
// If the DataChannel type is not specified, use the DataChannel type in
|
|
// the current description.
|
|
if (session_options.data_channel_type == DCT_NONE && current_content) {
|
|
RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_DATA));
|
|
is_sctp = (current_content->media_description()->protocol() ==
|
|
kMediaProtocolSctp);
|
|
}
|
|
if (is_sctp) {
|
|
return AddSctpDataContentForOffer(
|
|
media_description_options, session_options, current_content,
|
|
current_description, current_streams, desc, ice_credentials);
|
|
} else {
|
|
return AddRtpDataContentForOffer(media_description_options, session_options,
|
|
current_content, current_description,
|
|
rtp_data_codecs, current_streams, desc,
|
|
ice_credentials);
|
|
}
|
|
}
|
|
|
|
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,
|
|
const RtpDataCodecs& rtp_data_codecs,
|
|
StreamParamsVec* current_streams,
|
|
SessionDescription* answer,
|
|
IceCredentialsIterator* ice_credentials) const {
|
|
std::unique_ptr<TransportDescription> data_transport = CreateTransportAnswer(
|
|
media_description_options.mid, offer_description,
|
|
media_description_options.transport_options, current_description,
|
|
bundle_transport != nullptr, ice_credentials);
|
|
if (!data_transport) {
|
|
return false;
|
|
}
|
|
|
|
// Do not require or create SDES cryptos if DTLS is used.
|
|
cricket::SecurePolicy sdes_policy =
|
|
data_transport->secure() ? cricket::SEC_DISABLED : secure();
|
|
bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
|
|
session_options.bundle_enabled;
|
|
RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_DATA));
|
|
std::unique_ptr<MediaContentDescription> data_answer;
|
|
if (offer_content->media_description()->as_sctp()) {
|
|
// SCTP data content
|
|
data_answer = 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 {
|
|
// RTP offer
|
|
data_answer = std::make_unique<RtpDataContentDescription>();
|
|
|
|
const RtpDataContentDescription* offer_data_description =
|
|
offer_content->media_description()->as_rtp_data();
|
|
RTC_CHECK(offer_data_description);
|
|
if (!SetCodecsInAnswer(offer_data_description, rtp_data_codecs,
|
|
media_description_options, session_options,
|
|
ssrc_generator_, current_streams,
|
|
data_answer->as_rtp_data())) {
|
|
return false;
|
|
}
|
|
if (!CreateMediaContentAnswer(
|
|
offer_data_description, media_description_options, session_options,
|
|
sdes_policy, GetCryptos(current_content), RtpHeaderExtensions(),
|
|
ssrc_generator_, enable_encrypted_rtp_header_extensions_,
|
|
current_streams, bundle_enabled, data_answer.get())) {
|
|
return false; // Fails the session setup.
|
|
}
|
|
}
|
|
|
|
bool secure = bundle_transport ? bundle_transport->description.secure()
|
|
: data_transport->secure();
|
|
|
|
bool rejected = session_options.data_channel_type == DCT_NONE ||
|
|
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;
|
|
}
|
|
|
|
if (!rejected && session_options.data_channel_type == DCT_RTP) {
|
|
data_answer->set_bandwidth(kRtpDataMaxBandwidth);
|
|
} else {
|
|
// RFC 3264
|
|
// The answer MUST contain the same number of m-lines as the offer.
|
|
RTC_LOG(LS_INFO) << "Data is not supported in the answer.";
|
|
}
|
|
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 RtpDataContentDescription* GetFirstRtpDataContentDescription(
|
|
const SessionDescription* sdesc) {
|
|
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
|
|
return desc ? desc->as_rtp_data() : 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;
|
|
}
|
|
|
|
RtpDataContentDescription* GetFirstRtpDataContentDescription(
|
|
SessionDescription* sdesc) {
|
|
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
|
|
return desc ? desc->as_rtp_data() : nullptr;
|
|
}
|
|
|
|
SctpDataContentDescription* GetFirstSctpDataContentDescription(
|
|
SessionDescription* sdesc) {
|
|
auto desc = GetFirstMediaContentDescription(sdesc, MEDIA_TYPE_DATA);
|
|
return desc ? desc->as_sctp() : nullptr;
|
|
}
|
|
|
|
} // namespace cricket
|