/* * 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/srtptransport.h" #include #include #include "absl/memory/memory.h" #include "media/base/rtputils.h" #include "pc/rtptransport.h" #include "pc/srtpsession.h" #include "rtc_base/asyncpacketsocket.h" #include "rtc_base/copyonwritebuffer.h" #include "rtc_base/numerics/safe_conversions.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) : RtpTransport(rtcp_mux_enabled) {} RTCError SrtpTransport::SetSrtpSendKey(const cricket::CryptoParams& params) { if (send_params_) { LOG_AND_RETURN_ERROR( webrtc::RTCErrorType::UNSUPPORTED_OPERATION, "Setting the SRTP send key twice is currently unsupported."); } if (recv_params_ && recv_params_->cipher_suite != params.cipher_suite) { LOG_AND_RETURN_ERROR( webrtc::RTCErrorType::UNSUPPORTED_OPERATION, "The send key and receive key must have the same cipher suite."); } send_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(params.cipher_suite); if (*send_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Invalid SRTP crypto suite"); } int send_key_len, send_salt_len; if (!rtc::GetSrtpKeyAndSaltLengths(*send_cipher_suite_, &send_key_len, &send_salt_len)) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Could not get lengths for crypto suite(s):" " send cipher_suite "); } send_key_ = rtc::ZeroOnFreeBuffer(send_key_len + send_salt_len); if (!ParseKeyParams(params.key_params, send_key_.data(), send_key_.size())) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Failed to parse the crypto key params"); } if (!MaybeSetKeyParams()) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Failed to set the crypto key params"); } send_params_ = params; return RTCError::OK(); } RTCError SrtpTransport::SetSrtpReceiveKey(const cricket::CryptoParams& params) { if (recv_params_) { LOG_AND_RETURN_ERROR( webrtc::RTCErrorType::UNSUPPORTED_OPERATION, "Setting the SRTP send key twice is currently unsupported."); } if (send_params_ && send_params_->cipher_suite != params.cipher_suite) { LOG_AND_RETURN_ERROR( webrtc::RTCErrorType::UNSUPPORTED_OPERATION, "The send key and receive key must have the same cipher suite."); } recv_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(params.cipher_suite); if (*recv_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Invalid SRTP crypto suite"); } int recv_key_len, recv_salt_len; if (!rtc::GetSrtpKeyAndSaltLengths(*recv_cipher_suite_, &recv_key_len, &recv_salt_len)) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Could not get lengths for crypto suite(s):" " recv cipher_suite "); } recv_key_ = rtc::ZeroOnFreeBuffer(recv_key_len + recv_salt_len); if (!ParseKeyParams(params.key_params, recv_key_.data(), recv_key_.size())) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Failed to parse the crypto key params"); } if (!MaybeSetKeyParams()) { return RTCError(RTCErrorType::INVALID_PARAMETER, "Failed to set the crypto key params"); } recv_params_ = params; return RTCError::OK(); } 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->data(); int len = rtc::checked_cast(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(packet->capacity()), &len); #else if (!IsExternalAuthActive()) { res = ProtectRtp(data, len, static_cast(packet->capacity()), &len); } else { updated_options.packet_time_params.rtp_sendtime_extension_id = rtp_abs_sendtime_extn_id_; res = ProtectRtp(data, len, static_cast(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) { int seq_num = -1; uint32_t ssrc = 0; cricket::GetRtpSeqNum(data, len, &seq_num); cricket::GetRtpSsrc(data, len, &ssrc); 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->data(); int len = rtc::checked_cast(packet->size()); if (!ProtectRtcp(data, len, static_cast(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(rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { if (!IsSrtpActive()) { RTC_LOG(LS_WARNING) << "Inactive SRTP transport received an RTP packet. Drop it."; return; } TRACE_EVENT0("webrtc", "SRTP Decode"); char* data = packet->data(); int len = rtc::checked_cast(packet->size()); if (!UnprotectRtp(data, len, &len)) { int seq_num = -1; uint32_t ssrc = 0; cricket::GetRtpSeqNum(data, len, &seq_num); cricket::GetRtpSsrc(data, len, &ssrc); RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len << ", seqnum=" << seq_num << ", SSRC=" << ssrc; return; } packet->SetSize(len); DemuxPacket(packet, packet_time); } void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) { if (!IsSrtpActive()) { RTC_LOG(LS_WARNING) << "Inactive SRTP transport received an RTCP packet. Drop it."; return; } TRACE_EVENT0("webrtc", "SRTP Decode"); char* data = packet->data(); int len = rtc::checked_cast(packet->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; } packet->SetSize(len); SignalRtcpPacketReceived(packet, packet_time); } void SrtpTransport::OnNetworkRouteChanged( absl::optional 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; } SignalNetworkRouteChanged(network_route); } void SrtpTransport::OnWritableState( rtc::PacketTransportInternal* packet_transport) { SignalWritableState(IsWritable(/*rtcp=*/true) && IsWritable(/*rtcp=*/true)); } bool SrtpTransport::SetRtpParams(int send_cs, const uint8_t* send_key, int send_key_len, const std::vector& send_extension_ids, int recv_cs, const uint8_t* recv_key, int recv_key_len, const std::vector& 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_cs, send_key, send_key_len, send_extension_ids) : send_session_->UpdateSend(send_cs, send_key, send_key_len, send_extension_ids); if (!ret) { ResetParams(); return false; } ret = new_sessions ? recv_session_->SetRecv(recv_cs, recv_key, recv_key_len, recv_extension_ids) : recv_session_->UpdateRecv( recv_cs, 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 cipher_suite " << send_cs << " recv cipher_suite " << recv_cs; MaybeUpdateWritableState(); return true; } bool SrtpTransport::SetRtcpParams(int send_cs, const uint8_t* send_key, int send_key_len, const std::vector& send_extension_ids, int recv_cs, const uint8_t* recv_key, int recv_key_len, const std::vector& 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()); if (!send_rtcp_session_->SetSend(send_cs, send_key, send_key_len, send_extension_ids)) { return false; } recv_rtcp_session_.reset(new cricket::SrtpSession()); if (!recv_rtcp_session_->SetRecv(recv_cs, recv_key, recv_key_len, recv_extension_ids)) { return false; } RTC_LOG(LS_INFO) << "SRTCP activated with negotiated parameters:" " send cipher_suite " << send_cs << " recv cipher_suite " << recv_cs; 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()); recv_session_.reset(new cricket::SrtpSession()); 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_cipher_suite_ || !recv_cipher_suite_) { return true; } return SetRtpParams(*send_cipher_suite_, send_key_.data(), static_cast(send_key_.size()), std::vector(), *recv_cipher_suite_, recv_key_.data(), static_cast(recv_key_.size()), std::vector()); } 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 (key_params.find("inline:") != 0) { 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; SignalWritableState(writable_); } } } // namespace webrtc