| /* |
| * 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 <memory> |
| #include <type_traits> |
| #include <utility> // for std::pair |
| |
| #include "api/array_view.h" |
| #include "api/candidate.h" |
| #include "p2p/base/p2p_constants.h" |
| #include "p2p/base/p2p_transport_channel.h" |
| #include "pc/sctp_data_channel_transport.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" |
| |
| using webrtc::SdpType; |
| |
| namespace cricket { |
| |
| static bool VerifyIceParams(const JsepTransportDescription& jsep_description) { |
| // For legacy protocols. |
| // TODO(zhihuang): Remove this once the legacy protocol is no longer |
| // supported. |
| if (jsep_description.transport_desc.ice_ufrag.empty() && |
| jsep_description.transport_desc.ice_pwd.empty()) { |
| return true; |
| } |
| |
| if (jsep_description.transport_desc.ice_ufrag.length() < |
| ICE_UFRAG_MIN_LENGTH || |
| jsep_description.transport_desc.ice_ufrag.length() > |
| ICE_UFRAG_MAX_LENGTH) { |
| return false; |
| } |
| if (jsep_description.transport_desc.ice_pwd.length() < ICE_PWD_MIN_LENGTH || |
| jsep_description.transport_desc.ice_pwd.length() > ICE_PWD_MAX_LENGTH) { |
| return false; |
| } |
| return true; |
| } |
| |
| 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, |
| absl::optional<std::string> media_alt_protocol, |
| absl::optional<std::string> data_alt_protocol) |
| : 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), |
| media_alt_protocol(media_alt_protocol), |
| data_alt_protocol(data_alt_protocol) {} |
| |
| 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), |
| media_alt_protocol(from.media_alt_protocol), |
| data_alt_protocol(from.data_alt_protocol) {} |
| |
| 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; |
| media_alt_protocol = from.media_alt_protocol; |
| data_alt_protocol = from.data_alt_protocol; |
| |
| return *this; |
| } |
| |
| JsepTransport::JsepTransport( |
| const std::string& mid, |
| const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate, |
| std::unique_ptr<cricket::IceTransportInternal> ice_transport, |
| std::unique_ptr<cricket::IceTransportInternal> 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<webrtc::RtpTransportInternal> datagram_rtp_transport, |
| std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport, |
| std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport, |
| std::unique_ptr<SctpTransportInternal> sctp_transport, |
| std::unique_ptr<webrtc::MediaTransportInterface> media_transport, |
| std::unique_ptr<webrtc::DatagramTransportInterface> datagram_transport, |
| webrtc::DataChannelTransportInterface* data_channel_transport) |
| : 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 ? new rtc::RefCountedObject<webrtc::DtlsTransport>( |
| std::move(rtp_dtls_transport)) |
| : nullptr), |
| rtcp_dtls_transport_( |
| rtcp_dtls_transport |
| ? new rtc::RefCountedObject<webrtc::DtlsTransport>( |
| std::move(rtcp_dtls_transport)) |
| : nullptr), |
| sctp_data_channel_transport_( |
| sctp_transport ? std::make_unique<webrtc::SctpDataChannelTransport>( |
| sctp_transport.get()) |
| : nullptr), |
| sctp_transport_(sctp_transport |
| ? new rtc::RefCountedObject<webrtc::SctpTransport>( |
| std::move(sctp_transport)) |
| : nullptr), |
| media_transport_(std::move(media_transport)), |
| datagram_transport_(std::move(datagram_transport)), |
| datagram_rtp_transport_(std::move(datagram_rtp_transport)), |
| data_channel_transport_(data_channel_transport) { |
| 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)); |
| RTC_DCHECK(!datagram_transport_ || !media_transport_); |
| // 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_); |
| } |
| |
| if (datagram_rtp_transport_ && default_rtp_transport()) { |
| composite_rtp_transport_ = std::make_unique<webrtc::CompositeRtpTransport>( |
| std::vector<webrtc::RtpTransportInternal*>{ |
| datagram_rtp_transport_.get(), default_rtp_transport()}); |
| } |
| |
| if (media_transport_) { |
| media_transport_->SetMediaTransportStateCallback(this); |
| } |
| |
| if (data_channel_transport_ && sctp_data_channel_transport_) { |
| composite_data_channel_transport_ = |
| std::make_unique<webrtc::CompositeDataChannelTransport>( |
| std::vector<webrtc::DataChannelTransportInterface*>{ |
| data_channel_transport_, sctp_data_channel_transport_.get()}); |
| } |
| } |
| |
| JsepTransport::~JsepTransport() { |
| // Disconnect media transport state callbacks. |
| if (media_transport_) { |
| media_transport_->SetMediaTransportStateCallback(nullptr); |
| } |
| |
| 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; |
| |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (!VerifyIceParams(jsep_description)) { |
| return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, |
| "Invalid ice-ufrag or ice-pwd length."); |
| } |
| |
| 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. |
| { |
| rtc::CritScope scope(&accessor_lock_); |
| 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, |
| jsep_description.transport_desc.ice_ufrag, |
| jsep_description.transport_desc.ice_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_, local_fp); |
| if (!error.ok()) { |
| local_description_.reset(); |
| return error; |
| } |
| } |
| { |
| rtc::CritScope scope(&accessor_lock_); |
| RTC_DCHECK(rtp_dtls_transport_->internal()); |
| SetLocalIceParameters(rtp_dtls_transport_->internal()->ice_transport()); |
| |
| if (rtcp_dtls_transport_) { |
| RTC_DCHECK(rtcp_dtls_transport_->internal()); |
| SetLocalIceParameters(rtcp_dtls_transport_->internal()->ice_transport()); |
| } |
| } |
| // If PRANSWER/ANSWER is set, we should decide transport protocol type. |
| if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) { |
| error = NegotiateAndSetDtlsParameters(type); |
| NegotiateDatagramTransport(type); |
| } |
| if (!error.ok()) { |
| local_description_.reset(); |
| return error; |
| } |
| { |
| rtc::CritScope scope(&accessor_lock_); |
| 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) { |
| webrtc::RTCError error; |
| |
| RTC_DCHECK_RUN_ON(network_thread_); |
| if (!VerifyIceParams(jsep_description)) { |
| remote_description_.reset(); |
| return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, |
| "Invalid ice-ufrag or ice-pwd length."); |
| } |
| |
| 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. |
| { |
| rtc::CritScope lock(&accessor_lock_); |
| 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(rtp_dtls_transport()->ice_transport()); |
| |
| if (rtcp_dtls_transport()) { |
| SetRemoteIceParameters(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); |
| NegotiateDatagramTransport(type); |
| } |
| 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::CritScope scope(&accessor_lock_); |
| 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::CritScope scope(&accessor_lock_); |
| 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); |
| } |
| |
| absl::optional<OpaqueTransportParameters> |
| JsepTransport::GetTransportParameters() const { |
| rtc::CritScope scope(&accessor_lock_); |
| if (!datagram_transport()) { |
| return absl::nullopt; |
| } |
| |
| OpaqueTransportParameters params; |
| params.parameters = datagram_transport()->GetTransportParameters(); |
| return params; |
| } |
| |
| bool JsepTransport::GetStats(TransportStats* stats) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| rtc::CritScope scope(&accessor_lock_); |
| stats->transport_name = mid(); |
| stats->channel_stats.clear(); |
| RTC_DCHECK(rtp_dtls_transport_->internal()); |
| bool ret = GetTransportStats(rtp_dtls_transport_->internal(), stats); |
| if (rtcp_dtls_transport_) { |
| RTC_DCHECK(rtcp_dtls_transport_->internal()); |
| ret &= GetTransportStats(rtcp_dtls_transport_->internal(), stats); |
| } |
| return ret; |
| } |
| |
| webrtc::RTCError JsepTransport::VerifyCertificateFingerprint( |
| const rtc::RTCCertificate* certificate, |
| const rtc::SSLFingerprint* fingerprint) const { |
| 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_); |
| rtc::CritScope scope(&accessor_lock_); |
| if (dtls_srtp_transport_) { |
| RTC_LOG(INFO) |
| << "Setting active_reset_srtp_params of DtlsSrtpTransport to: " |
| << active_reset_srtp_params; |
| dtls_srtp_transport_->SetActiveResetSrtpParams(active_reset_srtp_params); |
| } |
| } |
| |
| void JsepTransport::SetLocalIceParameters(IceTransportInternal* ice_transport) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_DCHECK(ice_transport); |
| RTC_DCHECK(local_description_); |
| ice_transport->SetIceParameters( |
| local_description_->transport_desc.GetIceParameters()); |
| } |
| |
| void JsepTransport::SetRemoteIceParameters( |
| IceTransportInternal* ice_transport) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| RTC_DCHECK(ice_transport); |
| RTC_DCHECK(remote_description_); |
| ice_transport->SetRemoteIceParameters( |
| remote_description_->transport_desc.GetIceParameters()); |
| 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_RUN_ON(network_thread_); |
| RTC_DCHECK(dtls_transport); |
| // Set SSL role. Role must be set before fingerprint is applied, which |
| // initiates DTLS setup. |
| if (dtls_role && !dtls_transport->SetDtlsRole(*dtls_role)) { |
| return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, |
| "Failed to set SSL role for the transport."); |
| } |
| // Apply remote fingerprint. |
| if (!remote_fingerprint || |
| !dtls_transport->SetRemoteFingerprint( |
| remote_fingerprint->algorithm, remote_fingerprint->digest.cdata(), |
| remote_fingerprint->digest.size())) { |
| return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER, |
| "Failed to apply remote fingerprint."); |
| } |
| return webrtc::RTCError::OK(); |
| } |
| |
| 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_NOTREACHED(); |
| } |
| |
| if (!ret) { |
| return false; |
| } |
| |
| auto transport = rtp_transport(); |
| transport->SetRtcpMuxEnabled(rtcp_mux_negotiator_.IsActive()); |
| return ret; |
| } |
| |
| void JsepTransport::ActivateRtcpMux() { |
| { |
| // Don't hold the network_thread_ lock while calling other functions, |
| // since they might call other functions that call RTC_DCHECK_RUN_ON. |
| // TODO(https://crbug.com/webrtc/10318): Simplify when possible. |
| RTC_DCHECK_RUN_ON(network_thread_); |
| } |
| { |
| rtc::CritScope scope(&accessor_lock_); |
| 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. |
| SignalRtcpMuxActive(); |
| } |
| |
| 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_); |
| rtc::CritScope scope(&accessor_lock_); |
| 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. |
| 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 || |
| (*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 actpass value or 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, |
| TransportStats* stats) { |
| RTC_DCHECK_RUN_ON(network_thread_); |
| rtc::CritScope scope(&accessor_lock_); |
| RTC_DCHECK(dtls_transport); |
| TransportChannelStats substats; |
| if (rtcp_dtls_transport_) { |
| substats.component = dtls_transport == rtcp_dtls_transport_->internal() |
| ? ICE_CANDIDATE_COMPONENT_RTCP |
| : ICE_CANDIDATE_COMPONENT_RTP; |
| } else { |
| substats.component = ICE_CANDIDATE_COMPONENT_RTP; |
| } |
| dtls_transport->GetSrtpCryptoSuite(&substats.srtp_crypto_suite); |
| dtls_transport->GetSslCipherSuite(&substats.ssl_cipher_suite); |
| substats.dtls_state = dtls_transport->dtls_state(); |
| if (!dtls_transport->ice_transport()->GetStats( |
| &substats.ice_transport_stats)) { |
| return false; |
| } |
| stats->channel_stats.push_back(substats); |
| return true; |
| } |
| |
| void JsepTransport::OnStateChanged(webrtc::MediaTransportState state) { |
| // TODO(bugs.webrtc.org/9719) This method currently fires on the network |
| // thread, but media transport does not make such guarantees. We need to make |
| // sure this callback is guaranteed to be executed on the network thread. |
| RTC_DCHECK_RUN_ON(network_thread_); |
| { |
| rtc::CritScope scope(&accessor_lock_); |
| media_transport_state_ = state; |
| } |
| SignalMediaTransportStateChanged(); |
| } |
| |
| void JsepTransport::NegotiateDatagramTransport(SdpType type) { |
| RTC_DCHECK(type == SdpType::kAnswer || type == SdpType::kPrAnswer); |
| rtc::CritScope lock(&accessor_lock_); |
| if (!datagram_transport_) { |
| return; // No need to negotiate the use of datagram transport. |
| } |
| |
| bool compatible_datagram_transport = |
| remote_description_->transport_desc.opaque_parameters && |
| remote_description_->transport_desc.opaque_parameters == |
| local_description_->transport_desc.opaque_parameters; |
| |
| bool use_datagram_transport_for_media = |
| compatible_datagram_transport && |
| remote_description_->media_alt_protocol == |
| remote_description_->transport_desc.opaque_parameters->protocol && |
| remote_description_->media_alt_protocol == |
| local_description_->media_alt_protocol; |
| |
| bool use_datagram_transport_for_data = |
| compatible_datagram_transport && |
| remote_description_->data_alt_protocol == |
| remote_description_->transport_desc.opaque_parameters->protocol && |
| remote_description_->data_alt_protocol == |
| local_description_->data_alt_protocol; |
| |
| RTC_LOG(LS_INFO) |
| << "Negotiating datagram transport, use_datagram_transport_for_media=" |
| << use_datagram_transport_for_media |
| << ", use_datagram_transport_for_data=" << use_datagram_transport_for_data |
| << " answer type=" << (type == SdpType::kAnswer ? "answer" : "pr_answer"); |
| |
| // A provisional or full or answer lets the peer start sending on one of the |
| // transports. |
| if (composite_rtp_transport_) { |
| composite_rtp_transport_->SetSendTransport( |
| use_datagram_transport_for_media ? datagram_rtp_transport_.get() |
| : default_rtp_transport()); |
| } |
| if (composite_data_channel_transport_) { |
| composite_data_channel_transport_->SetSendTransport( |
| use_datagram_transport_for_data ? data_channel_transport_ |
| : sctp_data_channel_transport_.get()); |
| } |
| |
| if (type != SdpType::kAnswer) { |
| return; |
| } |
| |
| if (composite_rtp_transport_) { |
| if (use_datagram_transport_for_media) { |
| // Negotiated use of datagram transport for RTP, so remove the |
| // non-datagram RTP transport. |
| composite_rtp_transport_->RemoveTransport(default_rtp_transport()); |
| if (unencrypted_rtp_transport_) { |
| unencrypted_rtp_transport_ = nullptr; |
| } else if (sdes_transport_) { |
| sdes_transport_ = nullptr; |
| } else { |
| dtls_srtp_transport_ = nullptr; |
| } |
| } else { |
| composite_rtp_transport_->RemoveTransport(datagram_rtp_transport_.get()); |
| datagram_rtp_transport_ = nullptr; |
| } |
| } |
| |
| if (composite_data_channel_transport_) { |
| if (use_datagram_transport_for_data) { |
| // Negotiated use of datagram transport for data channels, so remove the |
| // non-datagram data channel transport. |
| composite_data_channel_transport_->RemoveTransport( |
| sctp_data_channel_transport_.get()); |
| sctp_data_channel_transport_ = nullptr; |
| sctp_transport_ = nullptr; |
| } else { |
| composite_data_channel_transport_->RemoveTransport( |
| data_channel_transport_); |
| data_channel_transport_ = nullptr; |
| } |
| } else if (data_channel_transport_ && !use_datagram_transport_for_data) { |
| // The datagram transport has been rejected without a fallback. We still |
| // need to inform the application and delete it. |
| SignalDataChannelTransportNegotiated(this, nullptr); |
| data_channel_transport_ = nullptr; |
| } |
| |
| if (!use_datagram_transport_for_media && !use_datagram_transport_for_data) { |
| // Datagram transport is not being used for anything, so clean it up. |
| datagram_transport_ = nullptr; |
| } |
| } |
| |
| } // namespace cricket |