Before this cl, ReadyToSend signaled false if sending a packet failed and transport->GetError() returns ECONN. ECONN may be reported by the TCP connection (TcpConnection) if the remote closed the connection. TcpConnection will attempt to reconnect and should change the writable state if it fail. Changing the state in the context of sending packets may cause recursive calls and seems to cause problems with incorrect states. It is simpler if RtpTransport::SendPacket ignore these failures and upper layers treat these lost packets similar to if the packets had been lost via UDP. For safety, this change can be reverted by field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/. Bug: webrtc:361124449 b/359989715 Change-Id: I8e7016dfb4301862286215c4512aa8ac03a16685 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360120 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42868}
463 lines
16 KiB
C++
463 lines
16 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/srtp_transport.h"
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#include <string.h>
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#include <string>
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#include <utility>
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#include <vector>
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#include "absl/strings/match.h"
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#include "media/base/rtp_utils.h"
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#include "modules/rtp_rtcp/source/rtp_util.h"
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#include "pc/rtp_transport.h"
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#include "pc/srtp_session.h"
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#include "rtc_base/async_packet_socket.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/copy_on_write_buffer.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/ssl_stream_adapter.h"
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#include "rtc_base/third_party/base64/base64.h"
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#include "rtc_base/trace_event.h"
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#include "rtc_base/zero_memory.h"
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namespace webrtc {
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SrtpTransport::SrtpTransport(bool rtcp_mux_enabled,
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const FieldTrialsView& field_trials)
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: RtpTransport(rtcp_mux_enabled, field_trials),
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field_trials_(field_trials) {}
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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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if (!IsSrtpActive()) {
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RTC_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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TRACE_EVENT0("webrtc", "SRTP Encode");
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bool res;
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uint8_t* data = packet->MutableData();
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int len = rtc::checked_cast<int>(packet->size());
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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 = nullptr;
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int key_len = 0;
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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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uint16_t seq_num = ParseRtpSequenceNumber(*packet);
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uint32_t ssrc = ParseRtpSsrc(*packet);
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RTC_LOG(LS_ERROR) << "Failed to protect RTP packet: size=" << len
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<< ", seqnum=" << seq_num << ", SSRC=" << ssrc;
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return false;
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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 SendPacket(/*rtcp=*/false, packet, updated_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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if (!IsSrtpActive()) {
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RTC_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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TRACE_EVENT0("webrtc", "SRTP Encode");
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uint8_t* data = packet->MutableData();
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int len = rtc::checked_cast<int>(packet->size());
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if (!ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len)) {
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int type = -1;
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cricket::GetRtcpType(data, len, &type);
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RTC_LOG(LS_ERROR) << "Failed to protect RTCP packet: size=" << len
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<< ", type=" << type;
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return false;
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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 SendPacket(/*rtcp=*/true, packet, options, flags);
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}
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void SrtpTransport::OnRtpPacketReceived(const rtc::ReceivedPacket& packet) {
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TRACE_EVENT0("webrtc", "SrtpTransport::OnRtpPacketReceived");
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if (!IsSrtpActive()) {
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RTC_LOG(LS_WARNING)
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<< "Inactive SRTP transport received an RTP packet. Drop it.";
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return;
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}
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rtc::CopyOnWriteBuffer payload(packet.payload());
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char* data = payload.MutableData<char>();
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int len = rtc::checked_cast<int>(payload.size());
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if (!UnprotectRtp(data, len, &len)) {
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// Limit the error logging to avoid excessive logs when there are lots of
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// bad packets.
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const int kFailureLogThrottleCount = 100;
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if (decryption_failure_count_ % kFailureLogThrottleCount == 0) {
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RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
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<< ", seqnum=" << ParseRtpSequenceNumber(payload)
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<< ", SSRC=" << ParseRtpSsrc(payload)
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<< ", previous failure count: "
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<< decryption_failure_count_;
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}
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++decryption_failure_count_;
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return;
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}
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payload.SetSize(len);
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DemuxPacket(std::move(payload),
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packet.arrival_time().value_or(Timestamp::MinusInfinity()),
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packet.ecn());
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}
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void SrtpTransport::OnRtcpPacketReceived(const rtc::ReceivedPacket& packet) {
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TRACE_EVENT0("webrtc", "SrtpTransport::OnRtcpPacketReceived");
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if (!IsSrtpActive()) {
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RTC_LOG(LS_WARNING)
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<< "Inactive SRTP transport received an RTCP packet. Drop it.";
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return;
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}
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rtc::CopyOnWriteBuffer payload(packet.payload());
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char* data = payload.MutableData<char>();
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int len = rtc::checked_cast<int>(payload.size());
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if (!UnprotectRtcp(data, len, &len)) {
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int type = -1;
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cricket::GetRtcpType(data, len, &type);
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RTC_LOG(LS_ERROR) << "Failed to unprotect RTCP packet: size=" << len
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<< ", type=" << type;
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return;
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}
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payload.SetSize(len);
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SendRtcpPacketReceived(
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&payload, packet.arrival_time() ? packet.arrival_time()->us() : -1);
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}
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void SrtpTransport::OnNetworkRouteChanged(
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absl::optional<rtc::NetworkRoute> network_route) {
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// Only append the SRTP overhead when there is a selected network route.
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if (network_route) {
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int srtp_overhead = 0;
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if (IsSrtpActive()) {
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GetSrtpOverhead(&srtp_overhead);
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}
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network_route->packet_overhead += srtp_overhead;
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}
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SendNetworkRouteChanged(network_route);
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}
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void SrtpTransport::OnWritableState(
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rtc::PacketTransportInternal* packet_transport) {
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SendWritableState(IsWritable(/*rtcp=*/false) && IsWritable(/*rtcp=*/true));
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}
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bool SrtpTransport::SetRtpParams(int send_crypto_suite,
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const uint8_t* send_key,
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int send_key_len,
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const std::vector<int>& send_extension_ids,
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int recv_crypto_suite,
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const uint8_t* recv_key,
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int recv_key_len,
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const std::vector<int>& recv_extension_ids) {
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// If parameters are being set for the first time, we should create new SRTP
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// sessions and call "SetSend/SetRecv". Otherwise we should call
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// "UpdateSend"/"UpdateRecv" on the existing sessions, which will internally
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// call "srtp_update".
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bool new_sessions = false;
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if (!send_session_) {
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RTC_DCHECK(!recv_session_);
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CreateSrtpSessions();
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new_sessions = true;
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}
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bool ret = new_sessions
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? send_session_->SetSend(send_crypto_suite, send_key,
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send_key_len, send_extension_ids)
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: send_session_->UpdateSend(send_crypto_suite, send_key,
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send_key_len, send_extension_ids);
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if (!ret) {
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ResetParams();
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return false;
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}
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ret = new_sessions
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? recv_session_->SetRecv(recv_crypto_suite, recv_key, recv_key_len,
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recv_extension_ids)
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: recv_session_->UpdateRecv(recv_crypto_suite, recv_key,
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recv_key_len, recv_extension_ids);
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if (!ret) {
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ResetParams();
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return false;
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}
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RTC_LOG(LS_INFO) << "SRTP " << (new_sessions ? "activated" : "updated")
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<< " with negotiated parameters: send crypto_suite "
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<< send_crypto_suite << " recv crypto_suite "
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<< recv_crypto_suite;
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MaybeUpdateWritableState();
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return true;
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}
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bool SrtpTransport::SetRtcpParams(int send_crypto_suite,
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const uint8_t* send_key,
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int send_key_len,
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const std::vector<int>& send_extension_ids,
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int recv_crypto_suite,
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const uint8_t* recv_key,
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int recv_key_len,
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const std::vector<int>& recv_extension_ids) {
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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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RTC_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(field_trials_));
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if (!send_rtcp_session_->SetSend(send_crypto_suite, send_key, send_key_len,
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send_extension_ids)) {
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return false;
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}
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recv_rtcp_session_.reset(new cricket::SrtpSession(field_trials_));
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if (!recv_rtcp_session_->SetRecv(recv_crypto_suite, recv_key, recv_key_len,
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recv_extension_ids)) {
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return false;
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}
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RTC_LOG(LS_INFO) << "SRTCP activated with negotiated parameters:"
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" send crypto_suite "
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<< send_crypto_suite << " recv crypto_suite "
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<< recv_crypto_suite;
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MaybeUpdateWritableState();
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return true;
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}
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bool SrtpTransport::IsSrtpActive() const {
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return send_session_ && recv_session_;
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}
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bool SrtpTransport::IsWritable(bool rtcp) const {
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return IsSrtpActive() && RtpTransport::IsWritable(rtcp);
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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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MaybeUpdateWritableState();
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RTC_LOG(LS_INFO) << "The params in SRTP transport are reset.";
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}
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void SrtpTransport::CreateSrtpSessions() {
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send_session_.reset(new cricket::SrtpSession(field_trials_));
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recv_session_.reset(new cricket::SrtpSession(field_trials_));
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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 (!IsSrtpActive()) {
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RTC_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 (!IsSrtpActive()) {
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RTC_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 (!IsSrtpActive()) {
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RTC_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 (!IsSrtpActive()) {
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RTC_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 (!IsSrtpActive()) {
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RTC_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 (!IsSrtpActive()) {
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RTC_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 (!IsSrtpActive()) {
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RTC_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(!IsSrtpActive());
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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 (!IsSrtpActive()) {
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RTC_LOG(LS_WARNING)
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<< "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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bool SrtpTransport::MaybeSetKeyParams() {
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if (!send_crypto_suite_ || !recv_crypto_suite_) {
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return true;
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}
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return SetRtpParams(*send_crypto_suite_, send_key_.data(),
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static_cast<int>(send_key_.size()), std::vector<int>(),
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*recv_crypto_suite_, recv_key_.data(),
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static_cast<int>(recv_key_.size()), std::vector<int>());
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}
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bool SrtpTransport::ParseKeyParams(const std::string& key_params,
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uint8_t* key,
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size_t len) {
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// example key_params: "inline:YUJDZGVmZ2hpSktMbW9QUXJzVHVWd3l6MTIzNDU2"
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// Fail if key-method is wrong.
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if (!absl::StartsWith(key_params, "inline:")) {
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return false;
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}
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// Fail if base64 decode fails, or the key is the wrong size.
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std::string key_b64(key_params.substr(7)), key_str;
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if (!rtc::Base64::Decode(key_b64, rtc::Base64::DO_STRICT, &key_str,
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nullptr) ||
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key_str.size() != len) {
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return false;
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}
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memcpy(key, key_str.c_str(), len);
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// TODO(bugs.webrtc.org/8905): Switch to ZeroOnFreeBuffer for storing
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// sensitive data.
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rtc::ExplicitZeroMemory(&key_str[0], key_str.size());
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return true;
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}
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void SrtpTransport::MaybeUpdateWritableState() {
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bool writable = IsWritable(/*rtcp=*/true) && IsWritable(/*rtcp=*/false);
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// Only fire the signal if the writable state changes.
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if (writable_ != writable) {
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writable_ = writable;
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SendWritableState(writable_);
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}
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}
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bool SrtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
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if (recv_session_ &&
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field_trials_.IsEnabled("WebRTC-SrtpRemoveReceiveStream")) {
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// Remove the SSRCs explicitly registered with the demuxer
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// (via SDP negotiation) from the SRTP session.
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for (const auto ssrc : GetSsrcsForSink(sink)) {
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if (!recv_session_->RemoveSsrcFromSession(ssrc)) {
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RTC_LOG(LS_WARNING)
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<< "Could not remove SSRC " << ssrc << " from SRTP session.";
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|
}
|
|
}
|
|
}
|
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return RtpTransport::UnregisterRtpDemuxerSink(sink);
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|
}
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} // namespace webrtc
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