/* * 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 #include #include #include #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) {} 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(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) { 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(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, int64_t packet_time_us) { TRACE_EVENT0("webrtc", "SrtpTransport::OnRtpPacketReceived"); if (!IsSrtpActive()) { RTC_LOG(LS_WARNING) << "Inactive SRTP transport received an RTP packet. Drop it."; return; } char* data = packet.MutableData(); int len = rtc::checked_cast(packet.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(packet) << ", SSRC=" << ParseRtpSsrc(packet) << ", previous failure count: " << decryption_failure_count_; } ++decryption_failure_count_; return; } packet.SetSize(len); DemuxPacket(std::move(packet), packet_time_us); } void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us) { TRACE_EVENT0("webrtc", "SrtpTransport::OnRtcpPacketReceived"); if (!IsSrtpActive()) { RTC_LOG(LS_WARNING) << "Inactive SRTP transport received an RTCP packet. Drop it."; return; } char* data = packet.MutableData(); 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); SendRtcpPacketReceived(&packet, packet_time_us); } 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; } 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& send_extension_ids, int recv_crypto_suite, 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_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& send_extension_ids, int recv_crypto_suite, 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(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(send_key_.size()), std::vector(), *recv_crypto_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 (!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