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/*
* 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) {}
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(
absl::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