webrtc_m130/pc/srtptransport.cc
Zhi Huang 2dfc42d7b6 Prepare to make BaseChannel depend on RtpTransportInternal only.
Eventually we want BaseChannel to depend on the RtpTransportInternal
instead of DtlsTransportInternal and share RtpTransport when bundling.
This CL is the first step.

Add SetRtpTransport and Init_w(RtptransportInternal*) to BaseChannel.
These two methods would replace the existing SetTransports and Init_w
methods.

Add new CreateVoice/VideoChannel methods to the ChannelManager which
 take RtpTransportInternal instead of Dtls/PacketTransportInternal.

|cotnent_name| is removed from the SrtpTransport to simplify to code
since it is only used for debugging.

InitNetwork_n is removed from BaseChannel in CL as well.

Bug: webrtc:7013
Change-Id: I35b1565958548bd4896854c49e61d3ee160b7634
Reviewed-on: https://webrtc-review.googlesource.com/27840
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21057}
2017-12-04 22:27:39 +00:00

396 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)
: 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