webrtc_m130/pc/srtptransport.cc
Zhi Huang c99b6c7936 Remove the SetEncryptedHeaderExtensionIds methods.
The existing methods SetEncrypedHeaderExtensionIds in SrtpTransport and SrtpSession
are removed because those methods could be confusing. When these methods are called
the head extension IDs are not actually updated and the user need to call SetRtpParams
again to make that happen. The existing setter just caches the new IDs.

To make it less confusing, the SetEncryptedHeaderExtensionIds is removed and the new
extension IDs will be set immediately when setting the crypto params.

For SDES, the crypto params and the header extension IDs will be set at the same time.

For DTLS, the new header extensions are cached in BaseChannel and will be set when
the DTLS handshake is completed.

Another major change is that when doing DTLS-SRTP, the encrypted header extension
IDs will be updated only when they are changed.

Bug: webrtc:7013
Change-Id: Ib70d4797456ae5ecb61b3dfff15c7e3e7ede89bd
Reviewed-on: https://webrtc-review.googlesource.com/15860
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20639}
2017-11-11 01:14:35 +00:00

376 lines
13 KiB
C++

/*
* 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,
const std::string& content_name)
: content_name_(content_name),
rtp_transport_(rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled)) {
ConnectToRtpTransport();
}
SrtpTransport::SrtpTransport(std::unique_ptr<RtpTransportInternal> transport,
const std::string& content_name)
: content_name_(content_name), rtp_transport_(std::move(transport)) {
ConnectToRtpTransport();
}
void SrtpTransport::ConnectToRtpTransport() {
rtp_transport_->SignalPacketReceived.connect(
this, &SrtpTransport::OnPacketReceived);
rtp_transport_->SignalReadyToSend.connect(this,
&SrtpTransport::OnReadyToSend);
}
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 " << content_name_
<< " 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 " << content_name_
<< " 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 " << content_name_
<< " 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 " << content_name_
<< " RTCP packet: size=" << len << ", type=" << type;
return;
}
}
packet->SetSize(len);
SignalPacketReceived(rtcp, packet, packet_time);
}
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