The SrtpTransport takes the SRTP responsibilities from the BaseChannel and SrtpFilter. SrtpTransport is now responsible for setting the crypto keys, protecting and unprotecting the packets. SrtpTransport doesn't know if the keys are from SDES or DTLS handshake. BaseChannel is now only responsible setting the offer/answer for SDES or extracting the key from DtlsTransport and configuring the SrtpTransport. SrtpFilter is used by BaseChannel as a helper for SDES negotiation. BUG=webrtc:7013 Change-Id: If61489dfbdf23481a1f1831ad181fbf45eaadb3e Reviewed-on: https://webrtc-review.googlesource.com/2560 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19977}
378 lines
12 KiB
C++
378 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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// 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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send_session_->SetEncryptedHeaderExtensionIds(
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send_encrypted_header_extension_ids_);
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bool ret = new_sessions
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? send_session_->SetSend(send_cs, send_key, send_key_len)
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: send_session_->UpdateSend(send_cs, send_key, send_key_len);
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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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recv_session_->SetEncryptedHeaderExtensionIds(
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recv_encrypted_header_extension_ids_);
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ret = new_sessions
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? recv_session_->SetRecv(recv_cs, recv_key, recv_key_len)
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: recv_session_->UpdateRecv(recv_cs, recv_key, recv_key_len);
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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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LOG(LS_INFO) << "SRTP " << (new_sessions ? "updated" : "activated")
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<< " 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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