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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/srtptransport.h"
#include <string>
#include <vector>
#include "media/base/rtputils.h"
#include "pc/rtptransport.h"
#include "pc/srtpsession.h"
#include "rtc_base/asyncpacketsocket.h"
#include "rtc_base/base64.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/trace_event.h"
namespace webrtc {
SrtpTransport::SrtpTransport(bool rtcp_mux_enabled)
: RtpTransportInternalAdapter(new RtpTransport(rtcp_mux_enabled)) {
// Own the raw pointer |transport| from the base class.
rtp_transport_.reset(transport_);
RTC_DCHECK(rtp_transport_);
ConnectToRtpTransport();
}
SrtpTransport::SrtpTransport(
std::unique_ptr<RtpTransportInternal> rtp_transport)
: RtpTransportInternalAdapter(rtp_transport.get()),
rtp_transport_(std::move(rtp_transport)) {
RTC_DCHECK(rtp_transport_);
ConnectToRtpTransport();
}
void SrtpTransport::ConnectToRtpTransport() {
rtp_transport_->SignalPacketReceived.connect(
this, &SrtpTransport::OnPacketReceived);
rtp_transport_->SignalReadyToSend.connect(this,
&SrtpTransport::OnReadyToSend);
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &SrtpTransport::OnNetworkRouteChanged);
rtp_transport_->SignalWritableState.connect(this,
&SrtpTransport::OnWritableState);
rtp_transport_->SignalSentPacket.connect(this, &SrtpTransport::OnSentPacket);
}
bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
return SendPacket(false, packet, options, flags);
}
bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
return SendPacket(true, packet, options, flags);
}
bool SrtpTransport::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
if (!IsActive()) {
RTC_LOG(LS_ERROR)
<< "Failed to send the packet because SRTP transport is inactive.";
return false;
}
rtc::PacketOptions updated_options = options;
rtc::CopyOnWriteBuffer cp = *packet;
TRACE_EVENT0("webrtc", "SRTP Encode");
bool res;
uint8_t* data = packet->data();
int len = static_cast<int>(packet->size());
if (!rtcp) {
// 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 = NULL;
int key_len;
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;
}
} else {
res = ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len);
if (!res) {
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 rtcp ? rtp_transport_->SendRtcpPacket(packet, updated_options, flags)
: rtp_transport_->SendRtpPacket(packet, updated_options, flags);
}
void SrtpTransport::OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
if (!IsActive()) {
RTC_LOG(LS_WARNING)
<< "Inactive SRTP transport received a packet. Drop it.";
return;
}
TRACE_EVENT0("webrtc", "SRTP Decode");
char* data = packet->data<char>();
int len = static_cast<int>(packet->size());
bool res;
if (!rtcp) {
res = UnprotectRtp(data, len, &len);
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 unprotect RTP packet: size=" << len
<< ", seqnum=" << seq_num << ", SSRC=" << ssrc;
return;
}
} else {
res = UnprotectRtcp(data, len, &len);
if (!res) {
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);
SignalPacketReceived(rtcp, packet, packet_time);
}
void SrtpTransport::OnNetworkRouteChanged(
rtc::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 (IsActive()) {
GetSrtpOverhead(&srtp_overhead);
}
network_route->packet_overhead += srtp_overhead;
}
SignalNetworkRouteChanged(network_route);
}
bool SrtpTransport::SetRtpParams(int send_cs,
const uint8_t* send_key,
int send_key_len,
const std::vector<int>& send_extension_ids,
int recv_cs,
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_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;
return true;
}
bool SrtpTransport::SetRtcpParams(int send_cs,
const uint8_t* send_key,
int send_key_len,
const std::vector<int>& send_extension_ids,
int recv_cs,
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());
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;
return true;
}
bool SrtpTransport::IsActive() const {
return send_session_ && recv_session_;
}
void SrtpTransport::ResetParams() {
send_session_ = nullptr;
recv_session_ = nullptr;
send_rtcp_session_ = nullptr;
recv_rtcp_session_ = nullptr;
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 (!IsActive()) {
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 (!IsActive()) {
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 (!IsActive()) {
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 (!IsActive()) {
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 (!IsActive()) {
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 (!IsActive()) {
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 (!IsActive()) {
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(!IsActive());
external_auth_enabled_ = true;
}
bool SrtpTransport::IsExternalAuthEnabled() const {
return external_auth_enabled_;
}
bool SrtpTransport::IsExternalAuthActive() const {
if (!IsActive()) {
RTC_LOG(LS_WARNING)
<< "Failed to check IsExternalAuthActive: SRTP not active";
return false;
}
RTC_CHECK(send_session_);
return send_session_->IsExternalAuthActive();
}
} // namespace webrtc