Make encrypted versions of RTP extension headers be stopped by default.
By this change we aim to remove the flag enable-webrtc-srtp-encrypted-headers. Bug: chromium:40623740 Change-Id: I74692c90ff1caf2a11d7b73211c1ae4472edfb4d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362740 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Emil Vardar (xWF) <vardar@google.com> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43105}
This commit is contained in:
Emil Vardar
WebRTC LUCI CQ
parent
95e697190f
commit
f5a547aa99
@ -76,6 +76,15 @@ RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
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int preferred_id,
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RtpTransceiverDirection direction)
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: uri(uri), preferred_id(preferred_id), direction(direction) {}
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RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
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absl::string_view uri,
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int preferred_id,
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bool preferred_encrypt,
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RtpTransceiverDirection direction)
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: uri(uri),
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preferred_id(preferred_id),
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preferred_encrypt(preferred_encrypt),
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direction(direction) {}
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RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() = default;
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RtpExtension::RtpExtension() = default;
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@ -212,7 +212,6 @@ struct RTC_EXPORT RtpHeaderExtensionCapability {
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std::optional<int> preferred_id;
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// If true, it's preferred that the value in the header is encrypted.
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// TODO(deadbeef): Not implemented.
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bool preferred_encrypt = false;
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// The direction of the extension. The kStopped value is only used with
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@ -227,6 +226,10 @@ struct RTC_EXPORT RtpHeaderExtensionCapability {
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RtpHeaderExtensionCapability(absl::string_view uri,
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int preferred_id,
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RtpTransceiverDirection direction);
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RtpHeaderExtensionCapability(absl::string_view uri,
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int preferred_id,
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bool preferred_encrypt,
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RtpTransceiverDirection direction);
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~RtpHeaderExtensionCapability();
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bool operator==(const RtpHeaderExtensionCapability& o) const {
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@ -62,7 +62,8 @@ using webrtc::RtpTransceiverDirection;
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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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capability.preferred_id.value_or(1),
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capability.preferred_encrypt);
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}
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cricket::RtpHeaderExtensions RtpHeaderExtensionsFromCapabilities(
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@ -866,6 +867,7 @@ std::vector<Codec> ComputeCodecsUnion(const std::vector<Codec>& codecs1,
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// and `encrypted_extensions` are used for both audio and video. There could be
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// overlap between audio extensions and video extensions.
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void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
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bool enable_encrypted_rtp_header_extensions,
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RtpHeaderExtensions* offered_extensions,
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RtpHeaderExtensions* regular_extensions,
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RtpHeaderExtensions* encrypted_extensions,
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@ -875,6 +877,9 @@ void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
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*offered_extensions, reference_extension.uri,
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reference_extension.encrypt)) {
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if (reference_extension.encrypt) {
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if (!enable_encrypted_rtp_header_extensions) {
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continue;
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}
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const webrtc::RtpExtension* existing =
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webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
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*encrypted_extensions, reference_extension.uri,
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@ -903,55 +908,6 @@ void MergeRtpHdrExts(const RtpHeaderExtensions& reference_extensions,
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}
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}
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void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* offered_extensions,
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RtpHeaderExtensions* encrypted_extensions,
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UsedRtpHeaderExtensionIds* used_ids) {
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RtpHeaderExtensions encrypted_extensions_to_add;
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for (const auto& extension : *offered_extensions) {
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// Skip existing encrypted offered extension
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if (extension.encrypt) {
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continue;
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}
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// Skip if we cannot encrypt the extension
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if (!webrtc::RtpExtension::IsEncryptionSupported(extension.uri)) {
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continue;
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}
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// Skip if an encrypted extension with that URI already exists in the
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// offered extensions.
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const bool have_encrypted_extension =
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webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
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*offered_extensions, extension.uri, true);
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if (have_encrypted_extension) {
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continue;
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}
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// Determine if a shared encrypted extension with that URI already exists.
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const webrtc::RtpExtension* shared_encrypted_extension =
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webrtc::RtpExtension::FindHeaderExtensionByUriAndEncryption(
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*encrypted_extensions, extension.uri, true);
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if (shared_encrypted_extension) {
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// Re-use the shared encrypted extension
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encrypted_extensions_to_add.push_back(*shared_encrypted_extension);
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continue;
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}
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// None exists. Create a new shared encrypted extension from the
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// non-encrypted one.
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webrtc::RtpExtension new_encrypted_extension(extension);
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new_encrypted_extension.encrypt = true;
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used_ids->FindAndSetIdUsed(&new_encrypted_extension);
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encrypted_extensions->push_back(new_encrypted_extension);
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encrypted_extensions_to_add.push_back(new_encrypted_extension);
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}
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// Append the additional encrypted extensions to be offered
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offered_extensions->insert(offered_extensions->end(),
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encrypted_extensions_to_add.begin(),
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encrypted_extensions_to_add.end());
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}
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// Mostly identical to RtpExtension::FindHeaderExtensionByUri but discards any
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// encrypted extensions that this implementation cannot encrypt.
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const webrtc::RtpExtension* FindHeaderExtensionByUriDiscardUnsupported(
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@ -1927,6 +1883,7 @@ MediaSessionDescriptionFactory::GetOfferedRtpHeaderExtensionsWithIds(
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UsedRtpHeaderExtensionIds used_ids(
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extmap_allow_mixed ? UsedRtpHeaderExtensionIds::IdDomain::kTwoByteAllowed
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: UsedRtpHeaderExtensionIds::IdDomain::kOneByteOnly);
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RtpHeaderExtensions all_regular_extensions;
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RtpHeaderExtensions all_encrypted_extensions;
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@ -1938,10 +1895,12 @@ MediaSessionDescriptionFactory::GetOfferedRtpHeaderExtensionsWithIds(
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for (const ContentInfo* content : current_active_contents) {
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if (IsMediaContentOfType(content, MEDIA_TYPE_AUDIO)) {
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MergeRtpHdrExts(content->media_description()->rtp_header_extensions(),
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enable_encrypted_rtp_header_extensions_,
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&offered_extensions.audio, &all_regular_extensions,
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&all_encrypted_extensions, &used_ids);
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} else if (IsMediaContentOfType(content, MEDIA_TYPE_VIDEO)) {
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MergeRtpHdrExts(content->media_description()->rtp_header_extensions(),
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enable_encrypted_rtp_header_extensions_,
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&offered_extensions.video, &all_regular_extensions,
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&all_encrypted_extensions, &used_ids);
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}
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@ -1956,22 +1915,15 @@ MediaSessionDescriptionFactory::GetOfferedRtpHeaderExtensionsWithIds(
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entry.header_extensions, all_regular_extensions,
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all_encrypted_extensions));
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if (entry.type == MEDIA_TYPE_AUDIO)
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MergeRtpHdrExts(filtered_extensions, &offered_extensions.audio,
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&all_regular_extensions, &all_encrypted_extensions,
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&used_ids);
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MergeRtpHdrExts(filtered_extensions,
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enable_encrypted_rtp_header_extensions_,
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&offered_extensions.audio, &all_regular_extensions,
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&all_encrypted_extensions, &used_ids);
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else if (entry.type == MEDIA_TYPE_VIDEO)
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MergeRtpHdrExts(filtered_extensions, &offered_extensions.video,
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&all_regular_extensions, &all_encrypted_extensions,
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&used_ids);
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}
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// TODO(jbauch): Support adding encrypted header extensions to existing
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// sessions.
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if (enable_encrypted_rtp_header_extensions_ &&
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current_active_contents.empty()) {
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AddEncryptedVersionsOfHdrExts(&offered_extensions.audio,
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&all_encrypted_extensions, &used_ids);
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AddEncryptedVersionsOfHdrExts(&offered_extensions.video,
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&all_encrypted_extensions, &used_ids);
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MergeRtpHdrExts(filtered_extensions,
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enable_encrypted_rtp_header_extensions_,
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&offered_extensions.video, &all_regular_extensions,
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&all_encrypted_extensions, &used_ids);
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}
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return offered_extensions;
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}
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@ -124,8 +124,6 @@ const RtpExtension kAudioRtpExtension1[] = {
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const RtpExtension kAudioRtpExtensionEncrypted1[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
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RtpExtension("http://google.com/testing/audio_something", 10),
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RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12, true),
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RtpExtension("http://google.com/testing/audio_something", 11, true),
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};
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@ -135,37 +133,24 @@ const RtpExtension kAudioRtpExtension2[] = {
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RtpExtension("http://google.com/testing/both_audio_and_video", 7),
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};
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const RtpExtension kAudioRtpExtensionEncrypted2[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 2),
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RtpExtension("http://google.com/testing/audio_something", 13, true),
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RtpExtension("http://google.com/testing/audio_something_else", 5, true),
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};
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const RtpExtension kAudioRtpExtension3[] = {
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RtpExtension("http://google.com/testing/audio_something", 2),
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RtpExtension("http://google.com/testing/both_audio_and_video", 3),
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};
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const RtpExtension kAudioRtpExtension3ForEncryption[] = {
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RtpExtension("http://google.com/testing/audio_something", 2),
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// Use RTP extension that supports encryption.
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
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};
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const RtpExtension kAudioRtpExtension3ForEncryptionOffer[] = {
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RtpExtension("http://google.com/testing/audio_something", 2),
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
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RtpExtension("http://google.com/testing/audio_something", 14, true),
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 13, true),
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};
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const RtpExtension kVideoRtpExtension3ForEncryptionOffer[] = {
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RtpExtension("http://google.com/testing/video_something", 4),
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 3),
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RtpExtension("http://google.com/testing/video_something", 12, true),
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 13, true),
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};
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const RtpExtension kAudioRtpExtensionAnswer[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
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};
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const RtpExtension kAudioRtpExtensionEncryptedAnswer[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 12, true),
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RtpExtension("urn:ietf:params:rtp-hdrext:ssrc-audio-level", 8),
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RtpExtension("http://google.com/testing/audio_something", 11, true),
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};
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const RtpExtension kVideoRtpExtension1[] = {
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@ -175,8 +160,6 @@ const RtpExtension kVideoRtpExtension1[] = {
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const RtpExtension kVideoRtpExtensionEncrypted1[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
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RtpExtension("http://google.com/testing/video_something", 13),
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 9, true),
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RtpExtension("http://google.com/testing/video_something", 7, true),
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};
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@ -186,23 +169,24 @@ const RtpExtension kVideoRtpExtension2[] = {
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RtpExtension("http://google.com/testing/both_audio_and_video", 7),
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};
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const RtpExtension kVideoRtpExtensionEncrypted2[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 8),
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RtpExtension("http://google.com/testing/video_something", 10, true),
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RtpExtension("http://google.com/testing/video_something_else", 4, true),
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};
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const RtpExtension kVideoRtpExtension3[] = {
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RtpExtension("http://google.com/testing/video_something", 4),
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RtpExtension("http://google.com/testing/both_audio_and_video", 5),
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};
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const RtpExtension kVideoRtpExtension3ForEncryption[] = {
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RtpExtension("http://google.com/testing/video_something", 4),
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// Use RTP extension that supports encryption.
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 5),
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};
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const RtpExtension kVideoRtpExtensionAnswer[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
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};
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const RtpExtension kVideoRtpExtensionEncryptedAnswer[] = {
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 9, true),
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RtpExtension("urn:ietf:params:rtp-hdrext:toffset", 14),
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RtpExtension("http://google.com/testing/video_something", 7, true),
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};
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const RtpExtension kRtpExtensionTransportSequenceNumber01[] = {
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@ -602,7 +586,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
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std::vector<webrtc::RtpHeaderExtensionCapability> capabilities;
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for (const auto& extension : extensions) {
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webrtc::RtpHeaderExtensionCapability capability(
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extension.uri, extension.id,
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extension.uri, extension.id, extension.encrypt,
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webrtc::RtpTransceiverDirection::kSendRecv);
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capabilities.push_back(capability);
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}
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@ -612,8 +596,10 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
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void SetAudioVideoRtpHeaderExtensions(RtpHeaderExtensions audio_exts,
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RtpHeaderExtensions video_exts,
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MediaSessionOptions* opts) {
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auto audio_caps = HeaderExtensionCapabilitiesFromRtpExtensions(audio_exts);
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auto video_caps = HeaderExtensionCapabilitiesFromRtpExtensions(video_exts);
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std::vector<webrtc::RtpHeaderExtensionCapability> audio_caps =
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HeaderExtensionCapabilitiesFromRtpExtensions(audio_exts);
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std::vector<webrtc::RtpHeaderExtensionCapability> video_caps =
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HeaderExtensionCapabilitiesFromRtpExtensions(video_exts);
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for (auto& entry : opts->media_description_options) {
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switch (entry.type) {
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case MEDIA_TYPE_AUDIO:
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@ -2042,16 +2028,20 @@ TEST_F(MediaSessionDescriptionFactoryTest,
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MediaSessionOptions opts;
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AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
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// TODO: bugs.webrtc.org/358039777 - Reverse all the tests since the default
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// value will be set to true in the future.
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f1_.set_enable_encrypted_rtp_header_extensions(true);
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f2_.set_enable_encrypted_rtp_header_extensions(true);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtension1),
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MAKE_VECTOR(kVideoRtpExtension1), &opts);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
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MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
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&opts);
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std::unique_ptr<SessionDescription> offer =
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f1_.CreateOfferOrError(opts, nullptr).MoveValue();
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ASSERT_TRUE(offer.get());
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtension2),
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MAKE_VECTOR(kVideoRtpExtension2), &opts);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtensionEncrypted2),
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MAKE_VECTOR(kVideoRtpExtensionEncrypted2),
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&opts);
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std::unique_ptr<SessionDescription> answer =
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f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
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@ -2074,15 +2064,19 @@ TEST_F(MediaSessionDescriptionFactoryTest,
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MediaSessionOptions opts;
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AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
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// TODO: bugs.webrtc.org/358039777 - Reverse all the tests since the default
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// value will be true.
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f1_.set_enable_encrypted_rtp_header_extensions(true);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtension1),
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MAKE_VECTOR(kVideoRtpExtension1), &opts);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
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MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
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&opts);
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std::unique_ptr<SessionDescription> offer =
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f1_.CreateOfferOrError(opts, nullptr).MoveValue();
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ASSERT_TRUE(offer.get());
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtension2),
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MAKE_VECTOR(kVideoRtpExtension2), &opts);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtensionEncrypted2),
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MAKE_VECTOR(kVideoRtpExtensionEncrypted2),
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&opts);
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std::unique_ptr<SessionDescription> answer =
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f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
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@ -2105,23 +2099,27 @@ TEST_F(MediaSessionDescriptionFactoryTest,
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MediaSessionOptions opts;
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AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
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// TODO: bugs.webrtc.org/358039777 - Reverse all the tests since the default
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// value will be true.
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f2_.set_enable_encrypted_rtp_header_extensions(true);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtension1),
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MAKE_VECTOR(kVideoRtpExtension1), &opts);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
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MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
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&opts);
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std::unique_ptr<SessionDescription> offer =
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f1_.CreateOfferOrError(opts, nullptr).MoveValue();
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ASSERT_TRUE(offer.get());
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtension2),
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MAKE_VECTOR(kVideoRtpExtension2), &opts);
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SetAudioVideoRtpHeaderExtensions(MAKE_VECTOR(kAudioRtpExtensionEncrypted2),
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MAKE_VECTOR(kVideoRtpExtensionEncrypted2),
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&opts);
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std::unique_ptr<SessionDescription> answer =
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f2_.CreateAnswerOrError(offer.get(), opts, nullptr).MoveValue();
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EXPECT_EQ(
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MAKE_VECTOR(kAudioRtpExtension1),
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MAKE_VECTOR(kAudioRtpExtensionAnswer),
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GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
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EXPECT_EQ(
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MAKE_VECTOR(kVideoRtpExtension1),
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MAKE_VECTOR(kVideoRtpExtensionAnswer),
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GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
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EXPECT_EQ(
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MAKE_VECTOR(kAudioRtpExtensionAnswer),
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@ -3512,39 +3510,6 @@ TEST_F(MediaSessionDescriptionFactoryTest, RtpExtensionIdReused) {
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->rtp_header_extensions());
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}
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// Same as "RtpExtensionIdReused" above for encrypted RTP extensions.
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TEST_F(MediaSessionDescriptionFactoryTest, RtpExtensionIdReusedEncrypted) {
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MediaSessionOptions opts;
|
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AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
|
||||
|
||||
f1_.set_enable_encrypted_rtp_header_extensions(true);
|
||||
f2_.set_enable_encrypted_rtp_header_extensions(true);
|
||||
|
||||
SetAudioVideoRtpHeaderExtensions(
|
||||
MAKE_VECTOR(kAudioRtpExtension3ForEncryption),
|
||||
MAKE_VECTOR(kVideoRtpExtension3ForEncryption), &opts);
|
||||
std::unique_ptr<SessionDescription> offer =
|
||||
f1_.CreateOfferOrError(opts, nullptr).MoveValue();
|
||||
|
||||
EXPECT_EQ(
|
||||
MAKE_VECTOR(kAudioRtpExtension3ForEncryptionOffer),
|
||||
GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
|
||||
EXPECT_EQ(
|
||||
MAKE_VECTOR(kVideoRtpExtension3ForEncryptionOffer),
|
||||
GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
|
||||
|
||||
// Nothing should change when creating a new offer
|
||||
std::unique_ptr<SessionDescription> updated_offer(
|
||||
f1_.CreateOfferOrError(opts, offer.get()).MoveValue());
|
||||
|
||||
EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension3ForEncryptionOffer),
|
||||
GetFirstAudioContentDescription(updated_offer.get())
|
||||
->rtp_header_extensions());
|
||||
EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtension3ForEncryptionOffer),
|
||||
GetFirstVideoContentDescription(updated_offer.get())
|
||||
->rtp_header_extensions());
|
||||
}
|
||||
|
||||
TEST(MediaSessionDescription, CopySessionDescription) {
|
||||
SessionDescription source;
|
||||
ContentGroup group(CN_AUDIO);
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user