Reason for revert: Speculative revert since all Android bots on WebRTC FYI started to fail when this CL landed. https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29 Original issue's description: > If SRTP sessions exist, don't create new ones when applying answer. > > Instead, call the "Update" methods of SrtpSession, which will just call > srtp_update, instead of wiping out the session state completely. > > This was causing decryption to stop working when subsequent > offers/answers are applied. We don't know enough about SRTP to > understand the root cause, and I wasn't able to write an integration > test that reproduces the issue... But at least this fixes the bug that > can be reproduced reliably using Hangouts. > > BUG=webrtc:8251 > > Review-Url: https://codereview.webrtc.org/3019443002 > Cr-Commit-Position: refs/heads/master@{#19874} > Committed: https://webrtc.googlesource.com/src/+/5ada7acf603e90e71990e9d4ff8f49389f24958c TBR=zhihuang@webrtc.org,deadbeef@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:8251 NOTRY=TRUE Review-Url: https://codereview.webrtc.org/3017543002 Cr-Commit-Position: refs/heads/master@{#19882}
365 lines
12 KiB
C++
365 lines
12 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/srtptransport.h"
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#include <string>
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#include "media/base/rtputils.h"
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#include "pc/rtptransport.h"
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#include "pc/srtpsession.h"
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#include "rtc_base/asyncpacketsocket.h"
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#include "rtc_base/base64.h"
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#include "rtc_base/copyonwritebuffer.h"
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#include "rtc_base/ptr_util.h"
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#include "rtc_base/trace_event.h"
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namespace webrtc {
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SrtpTransport::SrtpTransport(bool rtcp_mux_enabled,
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const std::string& content_name)
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: content_name_(content_name),
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rtp_transport_(rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled)) {
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ConnectToRtpTransport();
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}
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SrtpTransport::SrtpTransport(std::unique_ptr<RtpTransportInternal> transport,
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const std::string& content_name)
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: content_name_(content_name), rtp_transport_(std::move(transport)) {
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ConnectToRtpTransport();
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}
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void SrtpTransport::ConnectToRtpTransport() {
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rtp_transport_->SignalPacketReceived.connect(
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this, &SrtpTransport::OnPacketReceived);
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rtp_transport_->SignalReadyToSend.connect(this,
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&SrtpTransport::OnReadyToSend);
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}
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bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) {
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return SendPacket(false, packet, options, flags);
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}
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bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) {
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return SendPacket(true, packet, options, flags);
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}
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bool SrtpTransport::SendPacket(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags) {
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if (!IsActive()) {
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LOG(LS_ERROR)
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<< "Failed to send the packet because SRTP transport is inactive.";
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return false;
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}
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rtc::PacketOptions updated_options = options;
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rtc::CopyOnWriteBuffer cp = *packet;
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TRACE_EVENT0("webrtc", "SRTP Encode");
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bool res;
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uint8_t* data = packet->data();
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int len = static_cast<int>(packet->size());
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if (!rtcp) {
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// If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
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// inside libsrtp for a RTP packet. A external HMAC module will be writing
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// a fake HMAC value. This is ONLY done for a RTP packet.
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// Socket layer will update rtp sendtime extension header if present in
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// packet with current time before updating the HMAC.
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#if !defined(ENABLE_EXTERNAL_AUTH)
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res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
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#else
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if (!IsExternalAuthActive()) {
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res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
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} else {
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updated_options.packet_time_params.rtp_sendtime_extension_id =
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rtp_abs_sendtime_extn_id_;
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res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len,
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&updated_options.packet_time_params.srtp_packet_index);
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// If protection succeeds, let's get auth params from srtp.
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if (res) {
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uint8_t* auth_key = NULL;
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int key_len;
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res = GetRtpAuthParams(
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&auth_key, &key_len,
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&updated_options.packet_time_params.srtp_auth_tag_len);
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if (res) {
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updated_options.packet_time_params.srtp_auth_key.resize(key_len);
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updated_options.packet_time_params.srtp_auth_key.assign(
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auth_key, auth_key + key_len);
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}
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}
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}
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#endif
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if (!res) {
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int seq_num = -1;
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uint32_t ssrc = 0;
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cricket::GetRtpSeqNum(data, len, &seq_num);
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cricket::GetRtpSsrc(data, len, &ssrc);
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LOG(LS_ERROR) << "Failed to protect " << content_name_
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<< " RTP packet: size=" << len << ", seqnum=" << seq_num
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<< ", SSRC=" << ssrc;
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return false;
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}
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} else {
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res = ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len);
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if (!res) {
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int type = -1;
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cricket::GetRtcpType(data, len, &type);
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LOG(LS_ERROR) << "Failed to protect " << content_name_
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<< " RTCP packet: size=" << len << ", type=" << type;
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return false;
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}
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}
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// Update the length of the packet now that we've added the auth tag.
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packet->SetSize(len);
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return rtcp ? rtp_transport_->SendRtcpPacket(packet, updated_options, flags)
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: rtp_transport_->SendRtpPacket(packet, updated_options, flags);
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}
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void SrtpTransport::OnPacketReceived(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketTime& packet_time) {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Inactive SRTP transport received a packet. Drop it.";
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return;
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}
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TRACE_EVENT0("webrtc", "SRTP Decode");
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char* data = packet->data<char>();
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int len = static_cast<int>(packet->size());
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bool res;
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if (!rtcp) {
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res = UnprotectRtp(data, len, &len);
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if (!res) {
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int seq_num = -1;
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uint32_t ssrc = 0;
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cricket::GetRtpSeqNum(data, len, &seq_num);
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cricket::GetRtpSsrc(data, len, &ssrc);
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LOG(LS_ERROR) << "Failed to unprotect " << content_name_
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<< " RTP packet: size=" << len << ", seqnum=" << seq_num
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<< ", SSRC=" << ssrc;
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return;
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}
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} else {
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res = UnprotectRtcp(data, len, &len);
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if (!res) {
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int type = -1;
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cricket::GetRtcpType(data, len, &type);
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LOG(LS_ERROR) << "Failed to unprotect " << content_name_
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<< " RTCP packet: size=" << len << ", type=" << type;
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return;
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}
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}
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packet->SetSize(len);
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SignalPacketReceived(rtcp, packet, packet_time);
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}
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bool SrtpTransport::SetRtpParams(int send_cs,
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const uint8_t* send_key,
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int send_key_len,
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int recv_cs,
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const uint8_t* recv_key,
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int recv_key_len) {
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CreateSrtpSessions();
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send_session_->SetEncryptedHeaderExtensionIds(
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send_encrypted_header_extension_ids_);
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if (external_auth_enabled_) {
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send_session_->EnableExternalAuth();
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}
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if (!send_session_->SetSend(send_cs, send_key, send_key_len)) {
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ResetParams();
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return false;
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}
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recv_session_->SetEncryptedHeaderExtensionIds(
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recv_encrypted_header_extension_ids_);
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if (!recv_session_->SetRecv(recv_cs, recv_key, recv_key_len)) {
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ResetParams();
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return false;
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}
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LOG(LS_INFO) << "SRTP activated with negotiated parameters:"
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<< " send cipher_suite " << send_cs << " recv cipher_suite "
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<< recv_cs;
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return true;
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}
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bool SrtpTransport::SetRtcpParams(int send_cs,
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const uint8_t* send_key,
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int send_key_len,
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int recv_cs,
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const uint8_t* recv_key,
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int recv_key_len) {
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// This can only be called once, but can be safely called after
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// SetRtpParams
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if (send_rtcp_session_ || recv_rtcp_session_) {
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LOG(LS_ERROR) << "Tried to set SRTCP Params when filter already active";
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return false;
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}
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send_rtcp_session_.reset(new cricket::SrtpSession());
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if (!send_rtcp_session_->SetRecv(send_cs, send_key, send_key_len)) {
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return false;
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}
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recv_rtcp_session_.reset(new cricket::SrtpSession());
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if (!recv_rtcp_session_->SetRecv(recv_cs, recv_key, recv_key_len)) {
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return false;
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}
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LOG(LS_INFO) << "SRTCP activated with negotiated parameters:"
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<< " send cipher_suite " << send_cs << " recv cipher_suite "
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<< recv_cs;
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return true;
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}
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bool SrtpTransport::IsActive() const {
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return send_session_ && recv_session_;
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}
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void SrtpTransport::ResetParams() {
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send_session_ = nullptr;
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recv_session_ = nullptr;
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send_rtcp_session_ = nullptr;
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recv_rtcp_session_ = nullptr;
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LOG(LS_INFO) << "The params in SRTP transport are reset.";
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}
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void SrtpTransport::SetEncryptedHeaderExtensionIds(
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cricket::ContentSource source,
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const std::vector<int>& extension_ids) {
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if (source == cricket::CS_LOCAL) {
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recv_encrypted_header_extension_ids_ = extension_ids;
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} else {
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send_encrypted_header_extension_ids_ = extension_ids;
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}
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}
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void SrtpTransport::CreateSrtpSessions() {
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send_session_.reset(new cricket::SrtpSession());
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recv_session_.reset(new cricket::SrtpSession());
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if (external_auth_enabled_) {
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send_session_->EnableExternalAuth();
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}
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}
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bool SrtpTransport::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
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return false;
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}
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RTC_CHECK(send_session_);
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return send_session_->ProtectRtp(p, in_len, max_len, out_len);
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}
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bool SrtpTransport::ProtectRtp(void* p,
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int in_len,
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int max_len,
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int* out_len,
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int64_t* index) {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
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return false;
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}
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RTC_CHECK(send_session_);
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return send_session_->ProtectRtp(p, in_len, max_len, out_len, index);
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}
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bool SrtpTransport::ProtectRtcp(void* p,
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int in_len,
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int max_len,
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int* out_len) {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to ProtectRtcp: SRTP not active";
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return false;
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}
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if (send_rtcp_session_) {
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return send_rtcp_session_->ProtectRtcp(p, in_len, max_len, out_len);
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} else {
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RTC_CHECK(send_session_);
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return send_session_->ProtectRtcp(p, in_len, max_len, out_len);
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}
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}
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bool SrtpTransport::UnprotectRtp(void* p, int in_len, int* out_len) {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active";
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return false;
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}
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RTC_CHECK(recv_session_);
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return recv_session_->UnprotectRtp(p, in_len, out_len);
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}
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bool SrtpTransport::UnprotectRtcp(void* p, int in_len, int* out_len) {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to UnprotectRtcp: SRTP not active";
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return false;
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}
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if (recv_rtcp_session_) {
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return recv_rtcp_session_->UnprotectRtcp(p, in_len, out_len);
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} else {
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RTC_CHECK(recv_session_);
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return recv_session_->UnprotectRtcp(p, in_len, out_len);
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}
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}
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bool SrtpTransport::GetRtpAuthParams(uint8_t** key,
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int* key_len,
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int* tag_len) {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to GetRtpAuthParams: SRTP not active";
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return false;
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}
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RTC_CHECK(send_session_);
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return send_session_->GetRtpAuthParams(key, key_len, tag_len);
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}
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bool SrtpTransport::GetSrtpOverhead(int* srtp_overhead) const {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to GetSrtpOverhead: SRTP not active";
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return false;
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}
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RTC_CHECK(send_session_);
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*srtp_overhead = send_session_->GetSrtpOverhead();
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return true;
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}
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void SrtpTransport::EnableExternalAuth() {
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RTC_DCHECK(!IsActive());
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external_auth_enabled_ = true;
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}
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bool SrtpTransport::IsExternalAuthEnabled() const {
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return external_auth_enabled_;
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}
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bool SrtpTransport::IsExternalAuthActive() const {
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if (!IsActive()) {
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LOG(LS_WARNING) << "Failed to check IsExternalAuthActive: SRTP not active";
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return false;
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}
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RTC_CHECK(send_session_);
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return send_session_->IsExternalAuthActive();
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}
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} // namespace webrtc
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