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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/jsep_transport.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/jsep_transport.cc')
-rw-r--r-- | third_party/libwebrtc/pc/jsep_transport.cc | 711 |
1 files changed, 711 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/jsep_transport.cc b/third_party/libwebrtc/pc/jsep_transport.cc new file mode 100644 index 0000000000..ec186626b7 --- /dev/null +++ b/third_party/libwebrtc/pc/jsep_transport.cc @@ -0,0 +1,711 @@ +/* + * Copyright 2018 The WebRTC Project Authors. All rights reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "pc/jsep_transport.h" + +#include <stddef.h> +#include <stdint.h> + +#include <functional> +#include <memory> +#include <string> +#include <utility> + +#include "api/array_view.h" +#include "api/candidate.h" +#include "p2p/base/p2p_constants.h" +#include "p2p/base/p2p_transport_channel.h" +#include "rtc_base/checks.h" +#include "rtc_base/copy_on_write_buffer.h" +#include "rtc_base/logging.h" +#include "rtc_base/strings/string_builder.h" +#include "rtc_base/trace_event.h" + +using webrtc::SdpType; + +namespace cricket { + +JsepTransportDescription::JsepTransportDescription() {} + +JsepTransportDescription::JsepTransportDescription( + bool rtcp_mux_enabled, + const std::vector<CryptoParams>& cryptos, + const std::vector<int>& encrypted_header_extension_ids, + int rtp_abs_sendtime_extn_id, + const TransportDescription& transport_desc) + : rtcp_mux_enabled(rtcp_mux_enabled), + cryptos(cryptos), + encrypted_header_extension_ids(encrypted_header_extension_ids), + rtp_abs_sendtime_extn_id(rtp_abs_sendtime_extn_id), + transport_desc(transport_desc) {} + +JsepTransportDescription::JsepTransportDescription( + const JsepTransportDescription& from) + : rtcp_mux_enabled(from.rtcp_mux_enabled), + cryptos(from.cryptos), + encrypted_header_extension_ids(from.encrypted_header_extension_ids), + rtp_abs_sendtime_extn_id(from.rtp_abs_sendtime_extn_id), + transport_desc(from.transport_desc) {} + +JsepTransportDescription::~JsepTransportDescription() = default; + +JsepTransportDescription& JsepTransportDescription::operator=( + const JsepTransportDescription& from) { + if (this == &from) { + return *this; + } + rtcp_mux_enabled = from.rtcp_mux_enabled; + cryptos = from.cryptos; + encrypted_header_extension_ids = from.encrypted_header_extension_ids; + rtp_abs_sendtime_extn_id = from.rtp_abs_sendtime_extn_id; + transport_desc = from.transport_desc; + + return *this; +} + +JsepTransport::JsepTransport( + const std::string& mid, + const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate, + rtc::scoped_refptr<webrtc::IceTransportInterface> ice_transport, + rtc::scoped_refptr<webrtc::IceTransportInterface> rtcp_ice_transport, + std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport, + std::unique_ptr<webrtc::SrtpTransport> sdes_transport, + std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport, + std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport, + std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport, + std::unique_ptr<SctpTransportInternal> sctp_transport, + std::function<void()> rtcp_mux_active_callback) + : network_thread_(rtc::Thread::Current()), + mid_(mid), + local_certificate_(local_certificate), + ice_transport_(std::move(ice_transport)), + rtcp_ice_transport_(std::move(rtcp_ice_transport)), + unencrypted_rtp_transport_(std::move(unencrypted_rtp_transport)), + sdes_transport_(std::move(sdes_transport)), + dtls_srtp_transport_(std::move(dtls_srtp_transport)), + rtp_dtls_transport_(rtp_dtls_transport + ? rtc::make_ref_counted<webrtc::DtlsTransport>( + std::move(rtp_dtls_transport)) + : nullptr), + rtcp_dtls_transport_(rtcp_dtls_transport + ? rtc::make_ref_counted<webrtc::DtlsTransport>( + std::move(rtcp_dtls_transport)) + : nullptr), + sctp_transport_(sctp_transport + ? rtc::make_ref_counted<webrtc::SctpTransport>( + std::move(sctp_transport)) + : nullptr), + rtcp_mux_active_callback_(std::move(rtcp_mux_active_callback)) { + TRACE_EVENT0("webrtc", "JsepTransport::JsepTransport"); + RTC_DCHECK(ice_transport_); + RTC_DCHECK(rtp_dtls_transport_); + // `rtcp_ice_transport_` must be present iff `rtcp_dtls_transport_` is + // present. + RTC_DCHECK_EQ((rtcp_ice_transport_ != nullptr), + (rtcp_dtls_transport_ != nullptr)); + // Verify the "only one out of these three can be set" invariant. + if (unencrypted_rtp_transport_) { + RTC_DCHECK(!sdes_transport); + RTC_DCHECK(!dtls_srtp_transport); + } else if (sdes_transport_) { + RTC_DCHECK(!unencrypted_rtp_transport); + RTC_DCHECK(!dtls_srtp_transport); + } else { + RTC_DCHECK(dtls_srtp_transport_); + RTC_DCHECK(!unencrypted_rtp_transport); + RTC_DCHECK(!sdes_transport); + } + + if (sctp_transport_) { + sctp_transport_->SetDtlsTransport(rtp_dtls_transport_); + } +} + +JsepTransport::~JsepTransport() { + TRACE_EVENT0("webrtc", "JsepTransport::~JsepTransport"); + if (sctp_transport_) { + sctp_transport_->Clear(); + } + + // Clear all DtlsTransports. There may be pointers to these from + // other places, so we can't assume they'll be deleted by the destructor. + rtp_dtls_transport_->Clear(); + if (rtcp_dtls_transport_) { + rtcp_dtls_transport_->Clear(); + } + + // ICE will be the last transport to be deleted. +} + +webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription( + const JsepTransportDescription& jsep_description, + SdpType type) { + webrtc::RTCError error; + TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription"); + RTC_DCHECK_RUN_ON(network_thread_); + + IceParameters ice_parameters = + jsep_description.transport_desc.GetIceParameters(); + webrtc::RTCError ice_parameters_result = ice_parameters.Validate(); + if (!ice_parameters_result.ok()) { + rtc::StringBuilder sb; + sb << "Invalid ICE parameters: " << ice_parameters_result.message(); + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + sb.Release()); + } + + if (!SetRtcpMux(jsep_description.rtcp_mux_enabled, type, + ContentSource::CS_LOCAL)) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + "Failed to setup RTCP mux."); + } + + // If doing SDES, setup the SDES crypto parameters. + if (sdes_transport_) { + RTC_DCHECK(!unencrypted_rtp_transport_); + RTC_DCHECK(!dtls_srtp_transport_); + if (!SetSdes(jsep_description.cryptos, + jsep_description.encrypted_header_extension_ids, type, + ContentSource::CS_LOCAL)) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + "Failed to setup SDES crypto parameters."); + } + } else if (dtls_srtp_transport_) { + RTC_DCHECK(!unencrypted_rtp_transport_); + RTC_DCHECK(!sdes_transport_); + dtls_srtp_transport_->UpdateRecvEncryptedHeaderExtensionIds( + jsep_description.encrypted_header_extension_ids); + } + bool ice_restarting = + local_description_ != nullptr && + IceCredentialsChanged(local_description_->transport_desc.ice_ufrag, + local_description_->transport_desc.ice_pwd, + ice_parameters.ufrag, ice_parameters.pwd); + local_description_.reset(new JsepTransportDescription(jsep_description)); + + rtc::SSLFingerprint* local_fp = + local_description_->transport_desc.identity_fingerprint.get(); + + if (!local_fp) { + local_certificate_ = nullptr; + } else { + error = VerifyCertificateFingerprint(local_certificate_.get(), local_fp); + if (!error.ok()) { + local_description_.reset(); + return error; + } + } + RTC_DCHECK(rtp_dtls_transport_->internal()); + rtp_dtls_transport_->internal()->ice_transport()->SetIceParameters( + ice_parameters); + + if (rtcp_dtls_transport_) { + RTC_DCHECK(rtcp_dtls_transport_->internal()); + rtcp_dtls_transport_->internal()->ice_transport()->SetIceParameters( + ice_parameters); + } + // If PRANSWER/ANSWER is set, we should decide transport protocol type. + if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) { + error = NegotiateAndSetDtlsParameters(type); + } + if (!error.ok()) { + local_description_.reset(); + return error; + } + + if (needs_ice_restart_ && ice_restarting) { + needs_ice_restart_ = false; + RTC_LOG(LS_VERBOSE) << "needs-ice-restart flag cleared for transport " + << mid(); + } + + return webrtc::RTCError::OK(); +} + +webrtc::RTCError JsepTransport::SetRemoteJsepTransportDescription( + const JsepTransportDescription& jsep_description, + webrtc::SdpType type) { + TRACE_EVENT0("webrtc", "JsepTransport::SetLocalJsepTransportDescription"); + webrtc::RTCError error; + + RTC_DCHECK_RUN_ON(network_thread_); + + IceParameters ice_parameters = + jsep_description.transport_desc.GetIceParameters(); + webrtc::RTCError ice_parameters_result = ice_parameters.Validate(); + if (!ice_parameters_result.ok()) { + remote_description_.reset(); + rtc::StringBuilder sb; + sb << "Invalid ICE parameters: " << ice_parameters_result.message(); + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + sb.Release()); + } + + if (!SetRtcpMux(jsep_description.rtcp_mux_enabled, type, + ContentSource::CS_REMOTE)) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + "Failed to setup RTCP mux."); + } + + // If doing SDES, setup the SDES crypto parameters. + if (sdes_transport_) { + RTC_DCHECK(!unencrypted_rtp_transport_); + RTC_DCHECK(!dtls_srtp_transport_); + if (!SetSdes(jsep_description.cryptos, + jsep_description.encrypted_header_extension_ids, type, + ContentSource::CS_REMOTE)) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + "Failed to setup SDES crypto parameters."); + } + sdes_transport_->CacheRtpAbsSendTimeHeaderExtension( + jsep_description.rtp_abs_sendtime_extn_id); + } else if (dtls_srtp_transport_) { + RTC_DCHECK(!unencrypted_rtp_transport_); + RTC_DCHECK(!sdes_transport_); + dtls_srtp_transport_->UpdateSendEncryptedHeaderExtensionIds( + jsep_description.encrypted_header_extension_ids); + dtls_srtp_transport_->CacheRtpAbsSendTimeHeaderExtension( + jsep_description.rtp_abs_sendtime_extn_id); + } + + remote_description_.reset(new JsepTransportDescription(jsep_description)); + RTC_DCHECK(rtp_dtls_transport()); + SetRemoteIceParameters(ice_parameters, rtp_dtls_transport()->ice_transport()); + + if (rtcp_dtls_transport()) { + SetRemoteIceParameters(ice_parameters, + rtcp_dtls_transport()->ice_transport()); + } + + // If PRANSWER/ANSWER is set, we should decide transport protocol type. + if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) { + error = NegotiateAndSetDtlsParameters(SdpType::kOffer); + } + if (!error.ok()) { + remote_description_.reset(); + return error; + } + return webrtc::RTCError::OK(); +} + +webrtc::RTCError JsepTransport::AddRemoteCandidates( + const Candidates& candidates) { + RTC_DCHECK_RUN_ON(network_thread_); + if (!local_description_ || !remote_description_) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_STATE, + mid() + + " is not ready to use the remote candidate " + "because the local or remote description is " + "not set."); + } + + for (const cricket::Candidate& candidate : candidates) { + auto transport = + candidate.component() == cricket::ICE_CANDIDATE_COMPONENT_RTP + ? rtp_dtls_transport_ + : rtcp_dtls_transport_; + if (!transport) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + "Candidate has an unknown component: " + + candidate.ToSensitiveString() + " for mid " + + mid()); + } + RTC_DCHECK(transport->internal() && transport->internal()->ice_transport()); + transport->internal()->ice_transport()->AddRemoteCandidate(candidate); + } + return webrtc::RTCError::OK(); +} + +void JsepTransport::SetNeedsIceRestartFlag() { + RTC_DCHECK_RUN_ON(network_thread_); + if (!needs_ice_restart_) { + needs_ice_restart_ = true; + RTC_LOG(LS_VERBOSE) << "needs-ice-restart flag set for transport " << mid(); + } +} + +absl::optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const { + RTC_DCHECK_RUN_ON(network_thread_); + RTC_DCHECK(rtp_dtls_transport_); + RTC_DCHECK(rtp_dtls_transport_->internal()); + rtc::SSLRole dtls_role; + if (!rtp_dtls_transport_->internal()->GetDtlsRole(&dtls_role)) { + return absl::optional<rtc::SSLRole>(); + } + + return absl::optional<rtc::SSLRole>(dtls_role); +} + +bool JsepTransport::GetStats(TransportStats* stats) { + TRACE_EVENT0("webrtc", "JsepTransport::GetStats"); + RTC_DCHECK_RUN_ON(network_thread_); + stats->transport_name = mid(); + stats->channel_stats.clear(); + RTC_DCHECK(rtp_dtls_transport_->internal()); + bool ret = GetTransportStats(rtp_dtls_transport_->internal(), + ICE_CANDIDATE_COMPONENT_RTP, stats); + + if (rtcp_dtls_transport_) { + RTC_DCHECK(rtcp_dtls_transport_->internal()); + ret &= GetTransportStats(rtcp_dtls_transport_->internal(), + ICE_CANDIDATE_COMPONENT_RTCP, stats); + } + return ret; +} + +webrtc::RTCError JsepTransport::VerifyCertificateFingerprint( + const rtc::RTCCertificate* certificate, + const rtc::SSLFingerprint* fingerprint) const { + TRACE_EVENT0("webrtc", "JsepTransport::VerifyCertificateFingerprint"); + RTC_DCHECK_RUN_ON(network_thread_); + if (!fingerprint) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + "No fingerprint"); + } + if (!certificate) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + "Fingerprint provided but no identity available."); + } + std::unique_ptr<rtc::SSLFingerprint> fp_tmp = + rtc::SSLFingerprint::CreateUnique(fingerprint->algorithm, + *certificate->identity()); + RTC_DCHECK(fp_tmp.get() != NULL); + if (*fp_tmp == *fingerprint) { + return webrtc::RTCError::OK(); + } + char ss_buf[1024]; + rtc::SimpleStringBuilder desc(ss_buf); + desc << "Local fingerprint does not match identity. Expected: "; + desc << fp_tmp->ToString(); + desc << " Got: " << fingerprint->ToString(); + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, + std::string(desc.str())); +} + +void JsepTransport::SetActiveResetSrtpParams(bool active_reset_srtp_params) { + RTC_DCHECK_RUN_ON(network_thread_); + if (dtls_srtp_transport_) { + RTC_LOG(LS_INFO) + << "Setting active_reset_srtp_params of DtlsSrtpTransport to: " + << active_reset_srtp_params; + dtls_srtp_transport_->SetActiveResetSrtpParams(active_reset_srtp_params); + } +} + +void JsepTransport::SetRemoteIceParameters( + const IceParameters& ice_parameters, + IceTransportInternal* ice_transport) { + TRACE_EVENT0("webrtc", "JsepTransport::SetRemoteIceParameters"); + RTC_DCHECK_RUN_ON(network_thread_); + RTC_DCHECK(ice_transport); + RTC_DCHECK(remote_description_); + ice_transport->SetRemoteIceParameters(ice_parameters); + ice_transport->SetRemoteIceMode(remote_description_->transport_desc.ice_mode); +} + +webrtc::RTCError JsepTransport::SetNegotiatedDtlsParameters( + DtlsTransportInternal* dtls_transport, + absl::optional<rtc::SSLRole> dtls_role, + rtc::SSLFingerprint* remote_fingerprint) { + RTC_DCHECK(dtls_transport); + return dtls_transport->SetRemoteParameters( + remote_fingerprint->algorithm, remote_fingerprint->digest.cdata(), + remote_fingerprint->digest.size(), dtls_role); +} + +bool JsepTransport::SetRtcpMux(bool enable, + webrtc::SdpType type, + ContentSource source) { + RTC_DCHECK_RUN_ON(network_thread_); + bool ret = false; + switch (type) { + case SdpType::kOffer: + ret = rtcp_mux_negotiator_.SetOffer(enable, source); + break; + case SdpType::kPrAnswer: + // This may activate RTCP muxing, but we don't yet destroy the transport + // because the final answer may deactivate it. + ret = rtcp_mux_negotiator_.SetProvisionalAnswer(enable, source); + break; + case SdpType::kAnswer: + ret = rtcp_mux_negotiator_.SetAnswer(enable, source); + if (ret && rtcp_mux_negotiator_.IsActive()) { + ActivateRtcpMux(); + } + break; + default: + RTC_DCHECK_NOTREACHED(); + } + + if (!ret) { + return false; + } + + auto transport = rtp_transport(); + transport->SetRtcpMuxEnabled(rtcp_mux_negotiator_.IsActive()); + return ret; +} + +void JsepTransport::ActivateRtcpMux() { + if (unencrypted_rtp_transport_) { + RTC_DCHECK(!sdes_transport_); + RTC_DCHECK(!dtls_srtp_transport_); + unencrypted_rtp_transport_->SetRtcpPacketTransport(nullptr); + } else if (sdes_transport_) { + RTC_DCHECK(!unencrypted_rtp_transport_); + RTC_DCHECK(!dtls_srtp_transport_); + sdes_transport_->SetRtcpPacketTransport(nullptr); + } else if (dtls_srtp_transport_) { + RTC_DCHECK(dtls_srtp_transport_); + RTC_DCHECK(!unencrypted_rtp_transport_); + RTC_DCHECK(!sdes_transport_); + dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport(), + /*rtcp_dtls_transport=*/nullptr); + } + rtcp_dtls_transport_ = nullptr; // Destroy this reference. + // Notify the JsepTransportController to update the aggregate states. + rtcp_mux_active_callback_(); +} + +bool JsepTransport::SetSdes(const std::vector<CryptoParams>& cryptos, + const std::vector<int>& encrypted_extension_ids, + webrtc::SdpType type, + ContentSource source) { + RTC_DCHECK_RUN_ON(network_thread_); + bool ret = false; + ret = sdes_negotiator_.Process(cryptos, type, source); + if (!ret) { + return ret; + } + + if (source == ContentSource::CS_LOCAL) { + recv_extension_ids_ = encrypted_extension_ids; + } else { + send_extension_ids_ = encrypted_extension_ids; + } + + // If setting an SDES answer succeeded, apply the negotiated parameters + // to the SRTP transport. + if ((type == SdpType::kPrAnswer || type == SdpType::kAnswer) && ret) { + if (sdes_negotiator_.send_cipher_suite() && + sdes_negotiator_.recv_cipher_suite()) { + RTC_DCHECK(send_extension_ids_); + RTC_DCHECK(recv_extension_ids_); + ret = sdes_transport_->SetRtpParams( + *(sdes_negotiator_.send_cipher_suite()), + sdes_negotiator_.send_key().data(), + static_cast<int>(sdes_negotiator_.send_key().size()), + *(send_extension_ids_), *(sdes_negotiator_.recv_cipher_suite()), + sdes_negotiator_.recv_key().data(), + static_cast<int>(sdes_negotiator_.recv_key().size()), + *(recv_extension_ids_)); + } else { + RTC_LOG(LS_INFO) << "No crypto keys are provided for SDES."; + if (type == SdpType::kAnswer) { + // Explicitly reset the `sdes_transport_` if no crypto param is + // provided in the answer. No need to call `ResetParams()` for + // `sdes_negotiator_` because it resets the params inside `SetAnswer`. + sdes_transport_->ResetParams(); + } + } + } + return ret; +} + +webrtc::RTCError JsepTransport::NegotiateAndSetDtlsParameters( + SdpType local_description_type) { + RTC_DCHECK_RUN_ON(network_thread_); + if (!local_description_ || !remote_description_) { + return webrtc::RTCError(webrtc::RTCErrorType::INVALID_STATE, + "Applying an answer transport description " + "without applying any offer."); + } + std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint; + absl::optional<rtc::SSLRole> negotiated_dtls_role; + + rtc::SSLFingerprint* local_fp = + local_description_->transport_desc.identity_fingerprint.get(); + rtc::SSLFingerprint* remote_fp = + remote_description_->transport_desc.identity_fingerprint.get(); + if (remote_fp && local_fp) { + remote_fingerprint = std::make_unique<rtc::SSLFingerprint>(*remote_fp); + webrtc::RTCError error = + NegotiateDtlsRole(local_description_type, + local_description_->transport_desc.connection_role, + remote_description_->transport_desc.connection_role, + &negotiated_dtls_role); + if (!error.ok()) { + return error; + } + } else if (local_fp && (local_description_type == SdpType::kAnswer)) { + return webrtc::RTCError( + webrtc::RTCErrorType::INVALID_PARAMETER, + "Local fingerprint supplied when caller didn't offer DTLS."); + } else { + // We are not doing DTLS + remote_fingerprint = std::make_unique<rtc::SSLFingerprint>( + "", rtc::ArrayView<const uint8_t>()); + } + // Now that we have negotiated everything, push it downward. + // Note that we cache the result so that if we have race conditions + // between future SetRemote/SetLocal invocations and new transport + // creation, we have the negotiation state saved until a new + // negotiation happens. + RTC_DCHECK(rtp_dtls_transport()); + webrtc::RTCError error = SetNegotiatedDtlsParameters( + rtp_dtls_transport(), negotiated_dtls_role, remote_fingerprint.get()); + if (!error.ok()) { + return error; + } + + if (rtcp_dtls_transport()) { + error = SetNegotiatedDtlsParameters( + rtcp_dtls_transport(), negotiated_dtls_role, remote_fingerprint.get()); + } + return error; +} + +webrtc::RTCError JsepTransport::NegotiateDtlsRole( + SdpType local_description_type, + ConnectionRole local_connection_role, + ConnectionRole remote_connection_role, + absl::optional<rtc::SSLRole>* negotiated_dtls_role) { + // From RFC 4145, section-4.1, The following are the values that the + // 'setup' attribute can take in an offer/answer exchange: + // Offer Answer + // ________________ + // active passive / holdconn + // passive active / holdconn + // actpass active / passive / holdconn + // holdconn holdconn + // + // Set the role that is most conformant with RFC 5763, Section 5, bullet 1 + // The endpoint MUST use the setup attribute defined in [RFC4145]. + // The endpoint that is the offerer MUST use the setup attribute + // value of setup:actpass and be prepared to receive a client_hello + // before it receives the answer. The answerer MUST use either a + // setup attribute value of setup:active or setup:passive. Note that + // if the answerer uses setup:passive, then the DTLS handshake will + // not begin until the answerer is received, which adds additional + // latency. setup:active allows the answer and the DTLS handshake to + // occur in parallel. Thus, setup:active is RECOMMENDED. Whichever + // party is active MUST initiate a DTLS handshake by sending a + // ClientHello over each flow (host/port quartet). + // IOW - actpass and passive modes should be treated as server and + // active as client. + // RFC 8842 section 5.3 updates this text, so that it is mandated + // for the responder to handle offers with "active" and "passive" + // as well as "actpass" + bool is_remote_server = false; + if (local_description_type == SdpType::kOffer) { + if (local_connection_role != CONNECTIONROLE_ACTPASS) { + return webrtc::RTCError( + webrtc::RTCErrorType::INVALID_PARAMETER, + "Offerer must use actpass value for setup attribute."); + } + + if (remote_connection_role == CONNECTIONROLE_ACTIVE || + remote_connection_role == CONNECTIONROLE_PASSIVE || + remote_connection_role == CONNECTIONROLE_NONE) { + is_remote_server = (remote_connection_role == CONNECTIONROLE_PASSIVE); + } else { + return webrtc::RTCError( + webrtc::RTCErrorType::INVALID_PARAMETER, + "Answerer must use either active or passive value " + "for setup attribute."); + } + // If remote is NONE or ACTIVE it will act as client. + } else { + if (remote_connection_role != CONNECTIONROLE_ACTPASS && + remote_connection_role != CONNECTIONROLE_NONE) { + // Accept a remote role attribute that's not "actpass", but matches the + // current negotiated role. This is allowed by dtls-sdp, though our + // implementation will never generate such an offer as it's not + // recommended. + // + // See https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-dtls-sdp, + // section 5.5. + auto current_dtls_role = GetDtlsRole(); + if (!current_dtls_role) { + // Role not assigned yet. Verify that local role fits with remote role. + switch (remote_connection_role) { + case CONNECTIONROLE_ACTIVE: + if (local_connection_role != CONNECTIONROLE_PASSIVE) { + return webrtc::RTCError( + webrtc::RTCErrorType::INVALID_PARAMETER, + "Answerer must be passive when offerer is active"); + } + break; + case CONNECTIONROLE_PASSIVE: + if (local_connection_role != CONNECTIONROLE_ACTIVE) { + return webrtc::RTCError( + webrtc::RTCErrorType::INVALID_PARAMETER, + "Answerer must be active when offerer is passive"); + } + break; + default: + RTC_DCHECK_NOTREACHED(); + break; + } + } else { + if ((*current_dtls_role == rtc::SSL_CLIENT && + remote_connection_role == CONNECTIONROLE_ACTIVE) || + (*current_dtls_role == rtc::SSL_SERVER && + remote_connection_role == CONNECTIONROLE_PASSIVE)) { + return webrtc::RTCError( + webrtc::RTCErrorType::INVALID_PARAMETER, + "Offerer must use current negotiated role for " + "setup attribute."); + } + } + } + + if (local_connection_role == CONNECTIONROLE_ACTIVE || + local_connection_role == CONNECTIONROLE_PASSIVE) { + is_remote_server = (local_connection_role == CONNECTIONROLE_ACTIVE); + } else { + return webrtc::RTCError( + webrtc::RTCErrorType::INVALID_PARAMETER, + "Answerer must use either active or passive value " + "for setup attribute."); + } + + // If local is passive, local will act as server. + } + + *negotiated_dtls_role = + (is_remote_server ? rtc::SSL_CLIENT : rtc::SSL_SERVER); + return webrtc::RTCError::OK(); +} + +bool JsepTransport::GetTransportStats(DtlsTransportInternal* dtls_transport, + int component, + TransportStats* stats) { + RTC_DCHECK_RUN_ON(network_thread_); + RTC_DCHECK(dtls_transport); + TransportChannelStats substats; + substats.component = component; + dtls_transport->GetSslVersionBytes(&substats.ssl_version_bytes); + dtls_transport->GetSrtpCryptoSuite(&substats.srtp_crypto_suite); + dtls_transport->GetSslCipherSuite(&substats.ssl_cipher_suite); + substats.dtls_state = dtls_transport->dtls_state(); + rtc::SSLRole dtls_role; + if (dtls_transport->GetDtlsRole(&dtls_role)) { + substats.dtls_role = dtls_role; + } + if (!dtls_transport->ice_transport()->GetStats( + &substats.ice_transport_stats)) { + return false; + } + stats->channel_stats.push_back(substats); + return true; +} + +} // namespace cricket |