| /* |
| * Copyright 2017 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/srtp_transport.h" |
| |
| #include <string.h> |
| |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/strings/match.h" |
| #include "media/base/rtp_utils.h" |
| #include "modules/rtp_rtcp/source/rtp_util.h" |
| #include "pc/rtp_transport.h" |
| #include "pc/srtp_session.h" |
| #include "rtc_base/async_packet_socket.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/third_party/base64/base64.h" |
| #include "rtc_base/trace_event.h" |
| #include "rtc_base/zero_memory.h" |
| |
| namespace webrtc { |
| |
| SrtpTransport::SrtpTransport(bool rtcp_mux_enabled, |
| const FieldTrialsView& field_trials) |
| : RtpTransport(rtcp_mux_enabled, field_trials), |
| field_trials_(field_trials) {} |
| |
| bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_ERROR) |
| << "Failed to send the packet because SRTP transport is inactive."; |
| return false; |
| } |
| rtc::PacketOptions updated_options = options; |
| TRACE_EVENT0("webrtc", "SRTP Encode"); |
| bool res; |
| uint8_t* data = packet->MutableData(); |
| int len = rtc::checked_cast<int>(packet->size()); |
| // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done |
| // inside libsrtp for a RTP packet. A external HMAC module will be writing |
| // a fake HMAC value. This is ONLY done for a RTP packet. |
| // Socket layer will update rtp sendtime extension header if present in |
| // packet with current time before updating the HMAC. |
| #if !defined(ENABLE_EXTERNAL_AUTH) |
| res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len); |
| #else |
| if (!IsExternalAuthActive()) { |
| res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len); |
| } else { |
| updated_options.packet_time_params.rtp_sendtime_extension_id = |
| rtp_abs_sendtime_extn_id_; |
| res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len, |
| &updated_options.packet_time_params.srtp_packet_index); |
| // If protection succeeds, let's get auth params from srtp. |
| if (res) { |
| uint8_t* auth_key = nullptr; |
| int key_len = 0; |
| res = GetRtpAuthParams( |
| &auth_key, &key_len, |
| &updated_options.packet_time_params.srtp_auth_tag_len); |
| if (res) { |
| updated_options.packet_time_params.srtp_auth_key.resize(key_len); |
| updated_options.packet_time_params.srtp_auth_key.assign( |
| auth_key, auth_key + key_len); |
| } |
| } |
| } |
| #endif |
| if (!res) { |
| uint16_t seq_num = ParseRtpSequenceNumber(*packet); |
| uint32_t ssrc = ParseRtpSsrc(*packet); |
| RTC_LOG(LS_ERROR) << "Failed to protect RTP packet: size=" << len |
| << ", seqnum=" << seq_num << ", SSRC=" << ssrc; |
| return false; |
| } |
| |
| // Update the length of the packet now that we've added the auth tag. |
| packet->SetSize(len); |
| return SendPacket(/*rtcp=*/false, packet, updated_options, flags); |
| } |
| |
| bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet, |
| const rtc::PacketOptions& options, |
| int flags) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_ERROR) |
| << "Failed to send the packet because SRTP transport is inactive."; |
| return false; |
| } |
| |
| TRACE_EVENT0("webrtc", "SRTP Encode"); |
| uint8_t* data = packet->MutableData(); |
| int len = rtc::checked_cast<int>(packet->size()); |
| if (!ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len)) { |
| int type = -1; |
| cricket::GetRtcpType(data, len, &type); |
| RTC_LOG(LS_ERROR) << "Failed to protect RTCP packet: size=" << len |
| << ", type=" << type; |
| return false; |
| } |
| // Update the length of the packet now that we've added the auth tag. |
| packet->SetSize(len); |
| |
| return SendPacket(/*rtcp=*/true, packet, options, flags); |
| } |
| |
| void SrtpTransport::OnRtpPacketReceived(const rtc::ReceivedPacket& packet) { |
| TRACE_EVENT0("webrtc", "SrtpTransport::OnRtpPacketReceived"); |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) |
| << "Inactive SRTP transport received an RTP packet. Drop it."; |
| return; |
| } |
| |
| rtc::CopyOnWriteBuffer payload(packet.payload()); |
| char* data = payload.MutableData<char>(); |
| int len = rtc::checked_cast<int>(payload.size()); |
| if (!UnprotectRtp(data, len, &len)) { |
| // Limit the error logging to avoid excessive logs when there are lots of |
| // bad packets. |
| const int kFailureLogThrottleCount = 100; |
| if (decryption_failure_count_ % kFailureLogThrottleCount == 0) { |
| RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len |
| << ", seqnum=" << ParseRtpSequenceNumber(payload) |
| << ", SSRC=" << ParseRtpSsrc(payload) |
| << ", previous failure count: " |
| << decryption_failure_count_; |
| } |
| ++decryption_failure_count_; |
| return; |
| } |
| payload.SetSize(len); |
| DemuxPacket(std::move(payload), |
| packet.arrival_time().value_or(Timestamp::MinusInfinity()), |
| packet.ecn()); |
| } |
| |
| void SrtpTransport::OnRtcpPacketReceived(const rtc::ReceivedPacket& packet) { |
| TRACE_EVENT0("webrtc", "SrtpTransport::OnRtcpPacketReceived"); |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) |
| << "Inactive SRTP transport received an RTCP packet. Drop it."; |
| return; |
| } |
| rtc::CopyOnWriteBuffer payload(packet.payload()); |
| char* data = payload.MutableData<char>(); |
| int len = rtc::checked_cast<int>(payload.size()); |
| if (!UnprotectRtcp(data, len, &len)) { |
| int type = -1; |
| cricket::GetRtcpType(data, len, &type); |
| RTC_LOG(LS_ERROR) << "Failed to unprotect RTCP packet: size=" << len |
| << ", type=" << type; |
| return; |
| } |
| payload.SetSize(len); |
| SendRtcpPacketReceived( |
| &payload, packet.arrival_time() ? packet.arrival_time()->us() : -1); |
| } |
| |
| void SrtpTransport::OnNetworkRouteChanged( |
| std::optional<rtc::NetworkRoute> network_route) { |
| // Only append the SRTP overhead when there is a selected network route. |
| if (network_route) { |
| int srtp_overhead = 0; |
| if (IsSrtpActive()) { |
| GetSrtpOverhead(&srtp_overhead); |
| } |
| network_route->packet_overhead += srtp_overhead; |
| } |
| SendNetworkRouteChanged(network_route); |
| } |
| |
| void SrtpTransport::OnWritableState( |
| rtc::PacketTransportInternal* packet_transport) { |
| SendWritableState(IsWritable(/*rtcp=*/false) && IsWritable(/*rtcp=*/true)); |
| } |
| |
| bool SrtpTransport::SetRtpParams(int send_crypto_suite, |
| const uint8_t* send_key, |
| int send_key_len, |
| const std::vector<int>& send_extension_ids, |
| int recv_crypto_suite, |
| const uint8_t* recv_key, |
| int recv_key_len, |
| const std::vector<int>& recv_extension_ids) { |
| // If parameters are being set for the first time, we should create new SRTP |
| // sessions and call "SetSend/SetRecv". Otherwise we should call |
| // "UpdateSend"/"UpdateRecv" on the existing sessions, which will internally |
| // call "srtp_update". |
| bool new_sessions = false; |
| if (!send_session_) { |
| RTC_DCHECK(!recv_session_); |
| CreateSrtpSessions(); |
| new_sessions = true; |
| } |
| bool ret = new_sessions |
| ? send_session_->SetSend(send_crypto_suite, send_key, |
| send_key_len, send_extension_ids) |
| : send_session_->UpdateSend(send_crypto_suite, send_key, |
| send_key_len, send_extension_ids); |
| if (!ret) { |
| ResetParams(); |
| return false; |
| } |
| |
| ret = new_sessions |
| ? recv_session_->SetRecv(recv_crypto_suite, recv_key, recv_key_len, |
| recv_extension_ids) |
| : recv_session_->UpdateRecv(recv_crypto_suite, recv_key, |
| recv_key_len, recv_extension_ids); |
| if (!ret) { |
| ResetParams(); |
| return false; |
| } |
| |
| RTC_LOG(LS_INFO) << "SRTP " << (new_sessions ? "activated" : "updated") |
| << " with negotiated parameters: send crypto_suite " |
| << send_crypto_suite << " recv crypto_suite " |
| << recv_crypto_suite; |
| MaybeUpdateWritableState(); |
| return true; |
| } |
| |
| bool SrtpTransport::SetRtcpParams(int send_crypto_suite, |
| const uint8_t* send_key, |
| int send_key_len, |
| const std::vector<int>& send_extension_ids, |
| int recv_crypto_suite, |
| const uint8_t* recv_key, |
| int recv_key_len, |
| const std::vector<int>& recv_extension_ids) { |
| // This can only be called once, but can be safely called after |
| // SetRtpParams |
| if (send_rtcp_session_ || recv_rtcp_session_) { |
| RTC_LOG(LS_ERROR) << "Tried to set SRTCP Params when filter already active"; |
| return false; |
| } |
| |
| send_rtcp_session_.reset(new cricket::SrtpSession(field_trials_)); |
| if (!send_rtcp_session_->SetSend(send_crypto_suite, send_key, send_key_len, |
| send_extension_ids)) { |
| return false; |
| } |
| |
| recv_rtcp_session_.reset(new cricket::SrtpSession(field_trials_)); |
| if (!recv_rtcp_session_->SetRecv(recv_crypto_suite, recv_key, recv_key_len, |
| recv_extension_ids)) { |
| return false; |
| } |
| |
| RTC_LOG(LS_INFO) << "SRTCP activated with negotiated parameters:" |
| " send crypto_suite " |
| << send_crypto_suite << " recv crypto_suite " |
| << recv_crypto_suite; |
| MaybeUpdateWritableState(); |
| return true; |
| } |
| |
| bool SrtpTransport::IsSrtpActive() const { |
| return send_session_ && recv_session_; |
| } |
| |
| bool SrtpTransport::IsWritable(bool rtcp) const { |
| return IsSrtpActive() && RtpTransport::IsWritable(rtcp); |
| } |
| |
| void SrtpTransport::ResetParams() { |
| send_session_ = nullptr; |
| recv_session_ = nullptr; |
| send_rtcp_session_ = nullptr; |
| recv_rtcp_session_ = nullptr; |
| MaybeUpdateWritableState(); |
| RTC_LOG(LS_INFO) << "The params in SRTP transport are reset."; |
| } |
| |
| void SrtpTransport::CreateSrtpSessions() { |
| send_session_.reset(new cricket::SrtpSession(field_trials_)); |
| recv_session_.reset(new cricket::SrtpSession(field_trials_)); |
| if (external_auth_enabled_) { |
| send_session_->EnableExternalAuth(); |
| } |
| } |
| |
| bool SrtpTransport::ProtectRtp(void* p, int in_len, int max_len, int* out_len) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active"; |
| return false; |
| } |
| RTC_CHECK(send_session_); |
| return send_session_->ProtectRtp(p, in_len, max_len, out_len); |
| } |
| |
| bool SrtpTransport::ProtectRtp(void* p, |
| int in_len, |
| int max_len, |
| int* out_len, |
| int64_t* index) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active"; |
| return false; |
| } |
| RTC_CHECK(send_session_); |
| return send_session_->ProtectRtp(p, in_len, max_len, out_len, index); |
| } |
| |
| bool SrtpTransport::ProtectRtcp(void* p, |
| int in_len, |
| int max_len, |
| int* out_len) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) << "Failed to ProtectRtcp: SRTP not active"; |
| return false; |
| } |
| if (send_rtcp_session_) { |
| return send_rtcp_session_->ProtectRtcp(p, in_len, max_len, out_len); |
| } else { |
| RTC_CHECK(send_session_); |
| return send_session_->ProtectRtcp(p, in_len, max_len, out_len); |
| } |
| } |
| |
| bool SrtpTransport::UnprotectRtp(void* p, int in_len, int* out_len) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active"; |
| return false; |
| } |
| RTC_CHECK(recv_session_); |
| return recv_session_->UnprotectRtp(p, in_len, out_len); |
| } |
| |
| bool SrtpTransport::UnprotectRtcp(void* p, int in_len, int* out_len) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) << "Failed to UnprotectRtcp: SRTP not active"; |
| return false; |
| } |
| if (recv_rtcp_session_) { |
| return recv_rtcp_session_->UnprotectRtcp(p, in_len, out_len); |
| } else { |
| RTC_CHECK(recv_session_); |
| return recv_session_->UnprotectRtcp(p, in_len, out_len); |
| } |
| } |
| |
| bool SrtpTransport::GetRtpAuthParams(uint8_t** key, |
| int* key_len, |
| int* tag_len) { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) << "Failed to GetRtpAuthParams: SRTP not active"; |
| return false; |
| } |
| |
| RTC_CHECK(send_session_); |
| return send_session_->GetRtpAuthParams(key, key_len, tag_len); |
| } |
| |
| bool SrtpTransport::GetSrtpOverhead(int* srtp_overhead) const { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) << "Failed to GetSrtpOverhead: SRTP not active"; |
| return false; |
| } |
| |
| RTC_CHECK(send_session_); |
| *srtp_overhead = send_session_->GetSrtpOverhead(); |
| return true; |
| } |
| |
| void SrtpTransport::EnableExternalAuth() { |
| RTC_DCHECK(!IsSrtpActive()); |
| external_auth_enabled_ = true; |
| } |
| |
| bool SrtpTransport::IsExternalAuthEnabled() const { |
| return external_auth_enabled_; |
| } |
| |
| bool SrtpTransport::IsExternalAuthActive() const { |
| if (!IsSrtpActive()) { |
| RTC_LOG(LS_WARNING) |
| << "Failed to check IsExternalAuthActive: SRTP not active"; |
| return false; |
| } |
| |
| RTC_CHECK(send_session_); |
| return send_session_->IsExternalAuthActive(); |
| } |
| |
| bool SrtpTransport::MaybeSetKeyParams() { |
| if (!send_crypto_suite_ || !recv_crypto_suite_) { |
| return true; |
| } |
| |
| return SetRtpParams(*send_crypto_suite_, send_key_.data(), |
| static_cast<int>(send_key_.size()), std::vector<int>(), |
| *recv_crypto_suite_, recv_key_.data(), |
| static_cast<int>(recv_key_.size()), std::vector<int>()); |
| } |
| |
| bool SrtpTransport::ParseKeyParams(const std::string& key_params, |
| uint8_t* key, |
| size_t len) { |
| // example key_params: "inline:YUJDZGVmZ2hpSktMbW9QUXJzVHVWd3l6MTIzNDU2" |
| |
| // Fail if key-method is wrong. |
| if (!absl::StartsWith(key_params, "inline:")) { |
| return false; |
| } |
| |
| // Fail if base64 decode fails, or the key is the wrong size. |
| std::string key_b64(key_params.substr(7)), key_str; |
| if (!rtc::Base64::Decode(key_b64, rtc::Base64::DO_STRICT, &key_str, |
| nullptr) || |
| key_str.size() != len) { |
| return false; |
| } |
| |
| memcpy(key, key_str.c_str(), len); |
| // TODO(bugs.webrtc.org/8905): Switch to ZeroOnFreeBuffer for storing |
| // sensitive data. |
| rtc::ExplicitZeroMemory(&key_str[0], key_str.size()); |
| return true; |
| } |
| |
| void SrtpTransport::MaybeUpdateWritableState() { |
| bool writable = IsWritable(/*rtcp=*/true) && IsWritable(/*rtcp=*/false); |
| // Only fire the signal if the writable state changes. |
| if (writable_ != writable) { |
| writable_ = writable; |
| SendWritableState(writable_); |
| } |
| } |
| |
| bool SrtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) { |
| if (recv_session_ && |
| field_trials_.IsEnabled("WebRTC-SrtpRemoveReceiveStream")) { |
| // Remove the SSRCs explicitly registered with the demuxer |
| // (via SDP negotiation) from the SRTP session. |
| for (const auto ssrc : GetSsrcsForSink(sink)) { |
| if (!recv_session_->RemoveSsrcFromSession(ssrc)) { |
| RTC_LOG(LS_WARNING) |
| << "Could not remove SSRC " << ssrc << " from SRTP session."; |
| } |
| } |
| } |
| return RtpTransport::UnregisterRtpDemuxerSink(sink); |
| } |
| |
| } // namespace webrtc |
