admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport.""

Browse Source 
This reverts commit 27f3bf512827b483f9e0c67ce76362d83faa1950.

Reason for revert: Broken internal project.

Original change's description:
> Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."
> 
> This reverts commit 97d5e5b32c77bf550f1d788454f2db10ac9fbb1c.
> 
> Reason for revert: <INSERT REASONING HERE>
> 
> Original change's description:
> > Revert "Replace BundleFilter with RtpDemuxer in RtpTransport."
> > 
> > This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e.
> > 
> > Reason for revert: Broke chromium tests.
> > Original change's description:
> > > Replace BundleFilter with RtpDemuxer in RtpTransport.
> > > 
> > > BundleFilter is replaced by RtpDemuxer in RtpTransport for payload
> > > type-based demuxing. RtpTransport will support MID-based demuxing later.
> > > 
> > > Each BaseChannel has its own RTP demuxing criteria and when connecting
> > > to the RtpTransport, BaseChannel will register itself as a demuxer sink.
> > > 
> > > The inheritance model is changed. New inheritance chain:
> > > DtlsSrtpTransport->SrtpTransport->RtpTranpsort
> > > 
> > > NOTE:
> > > When RTCP packets are received, Call::DeliverRtcp will be called for
> > > multiple times (webrtc:9035) which is an existing issue. With this CL,
> > > it will become more of a problem and should be fixed.
> > > 
> > > Bug: webrtc:8587
> > > Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0
> > > Reviewed-on: https://webrtc-review.googlesource.com/61360
> > > Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22613}
> > 
> > TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
> > 
> > Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8587
> > Reviewed-on: https://webrtc-review.googlesource.com/64860
> > Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> > Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22614}
> 
> TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org
> 
> Change-Id: I3c272588ab4388ecadc4edc6786d5195c701855f
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8587
> Reviewed-on: https://webrtc-review.googlesource.com/64862
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22615}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8587
Change-Id: I694ce9a039ed52c5961cdc0cba57587bed4cbde4
Reviewed-on: https://webrtc-review.googlesource.com/65381
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22665}
This commit is contained in:
Zhi Huang 2018-03-29 00:08:03 +00:00 committed by Commit Bot
parent dea6889ef6
commit 95e7dbb7c7
28 changed files with 713 additions and 592 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

10
media/base/rtputils.cc
View File

@ -275,16 +275,6 @@ bool IsRtpPacket(const void* data, size_t len) {
return (static_cast<const uint8_t*>(data)[0] >> 6) == kRtpVersion;
}
// Check the RTP payload type. If 63 < payload type < 96, it's RTCP.
// For additional details, see http://tools.ietf.org/html/rfc5761.
bool IsRtcp(const char* data, int len) {
if (len < 2) {
return false;
}
char pt = data[1] & 0x7F;
return (63 < pt) && (pt < 96);
}
bool IsValidRtpPayloadType(int payload_type) {
return payload_type >= 0 && payload_type <= 127;
}

1
media/base/rtputils.h
View File

@ -55,7 +55,6 @@ bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
bool IsRtpPacket(const void* data, size_t len);
bool IsRtcp(const char* data, int len);
// True if |payload type| is 0-127.
bool IsValidRtpPayloadType(int payload_type);

8
pc/BUILD.gn
View File

@ -30,6 +30,8 @@ rtc_static_library("rtc_pc_base") {
defines = []
sources = [
"audiomonitor.h",
"bundlefilter.cc",
"bundlefilter.h",
"channel.cc",
"channel.h",
"channelmanager.cc",
@ -78,13 +80,10 @@ rtc_static_library("rtc_pc_base") {
"../api:optional",
"../api:ortc_api",
"../api:video_frame_api",
"../call:rtp_interfaces",
"../call:rtp_receiver",
"../common_video:common_video",
"../media:rtc_data",
"../media:rtc_h264_profile_id",
"../media:rtc_media_base",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:rtc_p2p",
"../rtc_base:checks",
"../rtc_base:rtc_base",
@ -275,6 +274,7 @@ if (rtc_include_tests) {
testonly = true
sources = [
"bundlefilter_unittest.cc",
"channel_unittest.cc",
"channelmanager_unittest.cc",
"currentspeakermonitor_unittest.cc",
@ -314,11 +314,9 @@ if (rtc_include_tests) {
"../api:array_view",
"../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
"../call:rtp_interfaces",
"../logging:rtc_event_log_api",
"../media:rtc_media_base",
"../media:rtc_media_tests_utils",
"../modules/rtp_rtcp:rtp_rtcp_format",
"../p2p:p2p_test_utils",
"../p2p:rtc_p2p",
"../rtc_base:checks",

49
pc/bundlefilter.cc Normal file
View File

@ -0,0 +1,49 @@
/*
* Copyright 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/bundlefilter.h"
#include "media/base/rtputils.h"
#include "rtc_base/logging.h"
namespace cricket {
BundleFilter::BundleFilter() {
}
BundleFilter::~BundleFilter() {
}
bool BundleFilter::DemuxPacket(const uint8_t* data, size_t len) {
// For RTP packets, we check whether the payload type can be found.
if (!IsRtpPacket(data, len)) {
return false;
}
int payload_type = 0;
if (!GetRtpPayloadType(data, len, &payload_type)) {
return false;
}
return FindPayloadType(payload_type);
}
void BundleFilter::AddPayloadType(int payload_type) {
payload_types_.insert(payload_type);
}
bool BundleFilter::FindPayloadType(int pl_type) const {
return payload_types_.find(pl_type) != payload_types_.end();
}
void BundleFilter::ClearAllPayloadTypes() {
payload_types_.clear();
}
} // namespace cricket

54
pc/bundlefilter.h Normal file
View File

@ -0,0 +1,54 @@
/*
* Copyright 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef PC_BUNDLEFILTER_H_
#define PC_BUNDLEFILTER_H_
#include <stdint.h>
#include <set>
#include <vector>
#include "media/base/streamparams.h"
#include "rtc_base/basictypes.h"
namespace cricket {
// In case of single RTP session and single transport channel, all session
// (or media) channels share a common transport channel. Hence they all get
// SignalReadPacket when packet received on transport channel. This requires
// cricket::BaseChannel to know all the valid sources, else media channel
// will decode invalid packets.
//
// This class determines whether a packet is destined for cricket::BaseChannel.
// This is only to be used for RTP packets as RTCP packets are not filtered.
// For RTP packets, this is decided based on the payload type.
class BundleFilter {
public:
BundleFilter();
~BundleFilter();
// Determines if a RTP packet belongs to valid cricket::BaseChannel.
bool DemuxPacket(const uint8_t* data, size_t len);
// Adds the supported payload type.
void AddPayloadType(int payload_type);
// Public for unittests.
bool FindPayloadType(int pl_type) const;
void ClearAllPayloadTypes();
private:
std::set<int> payload_types_;
};
} // namespace cricket
#endif // PC_BUNDLEFILTER_H_

72
pc/bundlefilter_unittest.cc Normal file
View File

@ -0,0 +1,72 @@
/*
* Copyright 2004 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "pc/bundlefilter.h"
#include "rtc_base/gunit.h"
using cricket::StreamParams;
static const int kPayloadType1 = 0x11;
static const int kPayloadType2 = 0x22;
static const int kPayloadType3 = 0x33;
// SSRC = 0x1111, Payload type = 0x11
static const unsigned char kRtpPacketPt1Ssrc1[] = {
0x80, kPayloadType1, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x11,
0x11,
};
// SSRC = 0x2222, Payload type = 0x22
static const unsigned char kRtpPacketPt2Ssrc2[] = {
0x80, 0x80 + kPayloadType2, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
0x22, 0x22,
};
// SSRC = 0x2222, Payload type = 0x33
static const unsigned char kRtpPacketPt3Ssrc2[] = {
0x80, kPayloadType3, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x22,
0x22,
};
// An SCTP packet.
static const unsigned char kSctpPacket[] = {
0x00, 0x01, 0x00, 0x01,
0xff, 0xff, 0xff, 0xff,
0x00, 0x00, 0x00, 0x00,
0x03, 0x00, 0x00, 0x04,
0x00, 0x00, 0x00, 0x00,
};
TEST(BundleFilterTest, RtpPacketTest) {
cricket::BundleFilter bundle_filter;
bundle_filter.AddPayloadType(kPayloadType1);
EXPECT_TRUE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1,
sizeof(kRtpPacketPt1Ssrc1)));
bundle_filter.AddPayloadType(kPayloadType2);
EXPECT_TRUE(bundle_filter.DemuxPacket(kRtpPacketPt2Ssrc2,
sizeof(kRtpPacketPt2Ssrc2)));
// Payload type 0x33 is not added.
EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt3Ssrc2,
sizeof(kRtpPacketPt3Ssrc2)));
// Size is too small.
EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1, 11));
bundle_filter.ClearAllPayloadTypes();
EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt1Ssrc1,
sizeof(kRtpPacketPt1Ssrc1)));
EXPECT_FALSE(bundle_filter.DemuxPacket(kRtpPacketPt2Ssrc2,
sizeof(kRtpPacketPt2Ssrc2)));
}
TEST(BundleFilterTest, InvalidRtpPacket) {
cricket::BundleFilter bundle_filter;
EXPECT_FALSE(bundle_filter.DemuxPacket(kSctpPacket, sizeof(kSctpPacket)));
}

128
pc/channel.cc
View File

@ -29,7 +29,6 @@
// Adding 'nogncheck' to disable the gn include headers check to support modular
// WebRTC build targets.
#include "media/engine/webrtcvoiceengine.h" // nogncheck
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/packettransportinternal.h"
#include "pc/channelmanager.h"
#include "pc/rtpmediautils.h"
@ -111,7 +110,6 @@ BaseChannel::BaseChannel(rtc::Thread* worker_thread,
srtp_required_(srtp_required),
media_channel_(std::move(media_channel)) {
RTC_DCHECK_RUN_ON(worker_thread_);
demuxer_criteria_.mid = content_name;
rtp_transport_ = unencrypted_rtp_transport_.get();
ConnectToRtpTransport();
RTC_LOG(LS_INFO) << "Created channel for " << content_name;
@ -133,12 +131,13 @@ BaseChannel::~BaseChannel() {
void BaseChannel::ConnectToRtpTransport() {
RTC_DCHECK(rtp_transport_);
bool success = RegisterRtpDemuxerSink();
RTC_DCHECK(success);
rtp_transport_->SignalReadyToSend.connect(
this, &BaseChannel::OnTransportReadyToSend);
rtp_transport_->SignalRtcpPacketReceived.connect(
this, &BaseChannel::OnRtcpPacketReceived);
// TODO(zstein): RtpTransport::SignalPacketReceived will probably be replaced
// with a callback interface later so that the demuxer can select which
// channel to signal.
rtp_transport_->SignalPacketReceived.connect(this,
&BaseChannel::OnPacketReceived);
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &BaseChannel::OnNetworkRouteChanged);
rtp_transport_->SignalWritableState.connect(this,
@ -155,9 +154,8 @@ void BaseChannel::ConnectToRtpTransport() {
void BaseChannel::DisconnectFromRtpTransport() {
RTC_DCHECK(rtp_transport_);
rtp_transport_->UnregisterRtpDemuxerSink(this);
rtp_transport_->SignalReadyToSend.disconnect(this);
rtp_transport_->SignalRtcpPacketReceived.disconnect(this);
rtp_transport_->SignalPacketReceived.disconnect(this);
rtp_transport_->SignalNetworkRouteChanged.disconnect(this);
rtp_transport_->SignalWritableState.disconnect(this);
rtp_transport_->SignalSentPacket.disconnect(this);
@ -205,26 +203,17 @@ void BaseChannel::Deinit() {
// functions, so need to stop this process in Deinit that is called in
// derived classes destructor.
network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
if (rtp_transport_) {
FlushRtcpMessages_n();
if (dtls_srtp_transport_) {
dtls_srtp_transport_->SetDtlsTransports(nullptr, nullptr);
} else {
rtp_transport_->SetRtpPacketTransport(nullptr);
rtp_transport_->SetRtcpPacketTransport(nullptr);
}
DisconnectFromRtpTransport();
}
FlushRtcpMessages_n();
if (dtls_srtp_transport_) {
dtls_srtp_transport_->SetDtlsTransports(nullptr, nullptr);
} else {
rtp_transport_->SetRtpPacketTransport(nullptr);
rtp_transport_->SetRtcpPacketTransport(nullptr);
}
// Clear pending read packets/messages.
network_thread_->Clear(&invoker_);
network_thread_->Clear(this);
// Because RTP level transports are accessed from the |network_thread_|,
// it's safer to release them from the |network_thread_| as well.
unencrypted_rtp_transport_.reset();
sdes_transport_.reset();
dtls_srtp_transport_.reset();
rtp_transport_ = nullptr;
});
}
@ -235,10 +224,8 @@ void BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
return;
});
}
RTC_DCHECK(rtp_transport);
if (rtp_transport == rtp_transport_) {
return;
}
if (rtp_transport_) {
DisconnectFromRtpTransport();
@ -594,37 +581,12 @@ bool BaseChannel::SendPacket(bool rtcp,
: rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS);
}
void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
// Reconstruct the PacketTime from the |parsed_packet|.
// RtpPacketReceived.arrival_time_ms = (PacketTime + 500) / 1000;
// Note: The |not_before| field is always 0 here. This field is not currently
// used, so it should be fine.
int64_t timestamp = -1;
if (parsed_packet.arrival_time_ms() > 0) {
timestamp = parsed_packet.arrival_time_ms() * 1000;
}
rtc::PacketTime packet_time(timestamp, /*not_before=*/0);
OnPacketReceived(/*rtcp=*/false, parsed_packet.Buffer(), packet_time);
}
void BaseChannel::UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions) {
RTC_DCHECK(rtp_transport_);
rtp_transport_->UpdateRtpHeaderExtensionMap(header_extensions);
}
bool BaseChannel::RegisterRtpDemuxerSink() {
RTC_DCHECK(rtp_transport_);
return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this);
}
void BaseChannel::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
OnPacketReceived(/*rtcp=*/true, *packet, packet_time);
bool BaseChannel::HandlesPayloadType(int packet_type) const {
return rtp_transport_->HandlesPayloadType(packet_type);
}
void BaseChannel::OnPacketReceived(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
if (!has_received_packet_ && !rtcp) {
has_received_packet_ = true;
@ -651,7 +613,7 @@ void BaseChannel::OnPacketReceived(bool rtcp,
invoker_.AsyncInvoke<void>(
RTC_FROM_HERE, worker_thread_,
Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time));
Bind(&BaseChannel::ProcessPacket, this, rtcp, *packet, packet_time));
}
void BaseChannel::ProcessPacket(bool rtcp,
@ -793,17 +755,14 @@ void BaseChannel::EnableSdes_n() {
// DtlsSrtpTransport and SrtpTransport shouldn't be enabled at the same
// time.
RTC_DCHECK(!dtls_srtp_transport_);
sdes_transport_ = rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_required_);
RTC_DCHECK(unencrypted_rtp_transport_);
sdes_transport_ = rtc::MakeUnique<webrtc::SrtpTransport>(
std::move(unencrypted_rtp_transport_));
#if defined(ENABLE_EXTERNAL_AUTH)
sdes_transport_->EnableExternalAuth();
#endif
sdes_transport_->SetRtpPacketTransport(
rtp_transport_->rtp_packet_transport());
sdes_transport_->SetRtcpPacketTransport(
rtp_transport_->rtcp_packet_transport());
SetRtpTransport(sdes_transport_.get());
RTC_LOG(LS_INFO) << "SrtpTransport is created for SDES.";
RTC_LOG(LS_INFO) << "Wrapping RtpTransport in SrtpTransport.";
}
void BaseChannel::EnableDtlsSrtp_n() {
@ -813,12 +772,15 @@ void BaseChannel::EnableDtlsSrtp_n() {
// DtlsSrtpTransport and SrtpTransport shouldn't be enabled at the same
// time.
RTC_DCHECK(!sdes_transport_);
RTC_DCHECK(unencrypted_rtp_transport_);
dtls_srtp_transport_ =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(rtcp_mux_required_);
auto srtp_transport = rtc::MakeUnique<webrtc::SrtpTransport>(
std::move(unencrypted_rtp_transport_));
#if defined(ENABLE_EXTERNAL_AUTH)
dtls_srtp_transport_->EnableExternalAuth();
srtp_transport->EnableExternalAuth();
#endif
dtls_srtp_transport_ =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(srtp_transport));
SetRtpTransport(dtls_srtp_transport_.get());
if (cached_send_extension_ids_) {
@ -834,7 +796,8 @@ void BaseChannel::EnableDtlsSrtp_n() {
RTC_DCHECK(rtp_dtls_transport_);
dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport_,
rtcp_dtls_transport_);
RTC_LOG(LS_INFO) << "DtlsSrtpTransport is created for DTLS-SRTP.";
RTC_LOG(LS_INFO) << "Wrapping SrtpTransport in DtlsSrtpTransport.";
}
bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
@ -1114,7 +1077,7 @@ void BaseChannel::OnMessage(rtc::Message *pmsg) {
}
void BaseChannel::AddHandledPayloadType(int payload_type) {
demuxer_criteria_.payload_types.insert(static_cast<uint8_t>(payload_type));
rtp_transport_->AddHandledPayloadType(payload_type);
}
void BaseChannel::FlushRtcpMessages_n() {
@ -1244,7 +1207,6 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
error_desc)) {
@ -1258,16 +1220,9 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
error_desc);
return false;
}
for (const AudioCodec& codec : audio->codecs()) {
AddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to set up audio demuxing.";
return false;
}
last_recv_params_ = recv_params;
// TODO(pthatcher): Move local streams into AudioSendParameters, and
@ -1302,7 +1257,6 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
if (!SetRtpTransportParameters(content, type, CS_REMOTE,
rtp_header_extensions, error_desc)) {
return false;
@ -1358,6 +1312,7 @@ VideoChannel::~VideoChannel() {
TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
@ -1395,7 +1350,6 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
error_desc)) {
@ -1409,16 +1363,9 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
error_desc);
return false;
}
for (const VideoCodec& codec : video->codecs()) {
AddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to set up video demuxing.";
return false;
}
last_recv_params_ = recv_params;
// TODO(pthatcher): Move local streams into VideoSendParameters, and
@ -1453,7 +1400,6 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
if (!SetRtpTransportParameters(content, type, CS_REMOTE,
rtp_header_extensions, error_desc)) {
return false;
@ -1513,6 +1459,7 @@ RtpDataChannel::~RtpDataChannel() {
TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel");
// this can't be done in the base class, since it calls a virtual
DisableMedia_w();
Deinit();
}
@ -1581,7 +1528,6 @@ bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
error_desc)) {
@ -1595,16 +1541,9 @@ bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
error_desc);
return false;
}
for (const DataCodec& codec : data->codecs()) {
AddHandledPayloadType(codec.id);
}
// Need to re-register the sink to update the handled payload.
if (!RegisterRtpDemuxerSink()) {
RTC_LOG(LS_ERROR) << "Failed to set up data demuxing.";
return false;
}
last_recv_params_ = recv_params;
// TODO(pthatcher): Move local streams into DataSendParameters, and
@ -1648,7 +1587,6 @@ bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content,
RtpHeaderExtensions rtp_header_extensions =
GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
UpdateRtpHeaderExtensionMap(rtp_header_extensions);
RTC_LOG(LS_INFO) << "Setting remote data description";
if (!SetRtpTransportParameters(content, type, CS_REMOTE,
rtp_header_extensions, error_desc)) {

30
pc/channel.h
View File

@ -23,7 +23,6 @@
#include "api/rtpreceiverinterface.h"
#include "api/videosinkinterface.h"
#include "api/videosourceinterface.h"
#include "call/rtp_packet_sink_interface.h"
#include "media/base/mediachannel.h"
#include "media/base/mediaengine.h"
#include "media/base/streamparams.h"
@ -70,10 +69,9 @@ class MediaContentDescription;
// vtable, and the media channel's thread using BaseChannel as the
// NetworkInterface.
class BaseChannel : public rtc::MessageHandler,
public sigslot::has_slots<>,
public MediaChannel::NetworkInterface,
public webrtc::RtpPacketSinkInterface {
class BaseChannel
: public rtc::MessageHandler, public sigslot::has_slots<>,
public MediaChannel::NetworkInterface {
public:
// If |srtp_required| is true, the channel will not send or receive any
// RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
@ -195,8 +193,10 @@ class BaseChannel : public rtc::MessageHandler,
virtual cricket::MediaType media_type() = 0;
// RtpPacketSinkInterface overrides.
void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
// Public for testing.
// TODO(zstein): Remove this once channels register themselves with
// an RtpTransport in a more explicit way.
bool HandlesPayloadType(int payload_type) const;
// Used by the RTCStatsCollector tests to set the transport name without
// creating RtpTransports.
@ -264,10 +264,12 @@ class BaseChannel : public rtc::MessageHandler,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options);
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
// TODO(zstein): packet can be const once the RtpTransport handles protection.
void OnPacketReceived(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void ProcessPacket(bool rtcp,
const rtc::CopyOnWriteBuffer& packet,
@ -358,11 +360,6 @@ class BaseChannel : public rtc::MessageHandler,
void AddHandledPayloadType(int payload_type);
void UpdateRtpHeaderExtensionMap(
const RtpHeaderExtensions& header_extensions);
bool RegisterRtpDemuxerSink();
private:
void ConnectToRtpTransport();
void DisconnectFromRtpTransport();
@ -442,9 +439,6 @@ class BaseChannel : public rtc::MessageHandler,
// The cached encrypted header extension IDs.
rtc::Optional<std::vector<int>> cached_send_extension_ids_;
rtc::Optional<std::vector<int>> cached_recv_extension_ids_;
bool encryption_disabled_ = false;
webrtc::RtpDemuxerCriteria demuxer_criteria_;
};
// VoiceChannel is a specialization that adds support for early media, DTMF,

8
pc/channel_unittest.cc
View File

@ -1609,6 +1609,10 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
EXPECT_TRUE(SendAccept());
EXPECT_EQ(rtcp_mux, !channel1_->NeedsRtcpTransport());
EXPECT_EQ(rtcp_mux, !channel2_->NeedsRtcpTransport());
EXPECT_TRUE(channel1_->HandlesPayloadType(pl_type1));
EXPECT_TRUE(channel2_->HandlesPayloadType(pl_type1));
EXPECT_FALSE(channel1_->HandlesPayloadType(pl_type2));
EXPECT_FALSE(channel2_->HandlesPayloadType(pl_type2));
// Both channels can receive pl_type1 only.
SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type1);
@ -1619,15 +1623,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
EXPECT_TRUE(CheckNoRtp1());
EXPECT_TRUE(CheckNoRtp2());
EXPECT_TRUE(SendInitiate());
EXPECT_TRUE(SendAccept());
// RTCP test
SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type2);
SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type2);
WaitForThreads();
EXPECT_FALSE(CheckCustomRtp2(kSsrc1, sequence_number1_1, pl_type2));
EXPECT_FALSE(CheckCustomRtp1(kSsrc2, sequence_number2_2, pl_type2));
// RTCP test
SendCustomRtcp1(kSsrc1);
SendCustomRtcp2(kSsrc2);
WaitForThreads();

1
pc/channelmanager.cc
View File

@ -507,6 +507,7 @@ void ChannelManager::DestroyRtpDataChannel(RtpDataChannel* data_channel) {
if (it == data_channels_.end()) {
return;
}
data_channels_.erase(it);
}

32
pc/channelmanager_unittest.cc
View File

@ -195,14 +195,11 @@ class ChannelManagerTestWithRtpTransport
RTPTransportType type = GetParam();
switch (type) {
case RTPTransportType::kRtp:
return rtc::MakeUnique<webrtc::RtpTransport>(
/*rtcp_mux_required=*/true);
return CreatePlainRtpTransport();
case RTPTransportType::kSrtp:
return rtc::MakeUnique<webrtc::SrtpTransport>(
/*rtcp_mux_required=*/true);
return CreateSrtpTransport();
case RTPTransportType::kDtlsSrtp:
return rtc::MakeUnique<webrtc::DtlsSrtpTransport>(
/*rtcp_mux_required=*/true);
return CreateDtlsSrtpTransport();
}
return nullptr;
}
@ -227,6 +224,29 @@ class ChannelManagerTestWithRtpTransport
cm_->DestroyRtpDataChannel(rtp_data_channel);
cm_->Terminate();
}
private:
std::unique_ptr<webrtc::RtpTransportInternal> CreatePlainRtpTransport() {
return rtc::MakeUnique<webrtc::RtpTransport>(/*rtcp_mux_required=*/true);
}
std::unique_ptr<webrtc::RtpTransportInternal> CreateSrtpTransport() {
auto rtp_transport =
rtc::MakeUnique<webrtc::RtpTransport>(/*rtcp_mux_required=*/true);
auto srtp_transport =
rtc::MakeUnique<webrtc::SrtpTransport>(std::move(rtp_transport));
return srtp_transport;
}
std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
auto rtp_transport =
rtc::MakeUnique<webrtc::RtpTransport>(/*rtcp_mux_required=*/true);
auto srtp_transport =
rtc::MakeUnique<webrtc::SrtpTransport>(std::move(rtp_transport));
auto dtls_srtp_transport_ =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(srtp_transport));
return dtls_srtp_transport_;
}
};
TEST_P(ChannelManagerTestWithRtpTransport, CreateDestroyChannels) {

71
pc/dtlssrtptransport.cc
View File

@ -24,8 +24,22 @@ static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
namespace webrtc {
DtlsSrtpTransport::DtlsSrtpTransport(bool rtcp_mux_enabled)
: SrtpTransport(rtcp_mux_enabled) {}
DtlsSrtpTransport::DtlsSrtpTransport(
std::unique_ptr<webrtc::SrtpTransport> srtp_transport)
: RtpTransportInternalAdapter(srtp_transport.get()) {
srtp_transport_ = std::move(srtp_transport);
RTC_DCHECK(srtp_transport_);
srtp_transport_->SignalPacketReceived.connect(
this, &DtlsSrtpTransport::OnPacketReceived);
srtp_transport_->SignalReadyToSend.connect(this,
&DtlsSrtpTransport::OnReadyToSend);
srtp_transport_->SignalNetworkRouteChanged.connect(
this, &DtlsSrtpTransport::OnNetworkRouteChanged);
srtp_transport_->SignalWritableState.connect(
this, &DtlsSrtpTransport::OnWritableState);
srtp_transport_->SignalSentPacket.connect(this,
&DtlsSrtpTransport::OnSentPacket);
}
void DtlsSrtpTransport::SetDtlsTransports(
cricket::DtlsTransportInternal* rtp_dtls_transport,
@ -40,7 +54,7 @@ void DtlsSrtpTransport::SetDtlsTransports(
// DtlsTransport changes and wait until the DTLS handshake is complete to set
// the newly negotiated parameters.
if (IsActive()) {
ResetParams();
srtp_transport_->ResetParams();
}
const std::string transport_name =
@ -66,7 +80,7 @@ void DtlsSrtpTransport::SetDtlsTransports(
}
void DtlsSrtpTransport::SetRtcpMuxEnabled(bool enable) {
SrtpTransport::SetRtcpMuxEnabled(enable);
srtp_transport_->SetRtcpMuxEnabled(enable);
if (enable) {
UpdateWritableStateAndMaybeSetupDtlsSrtp();
}
@ -114,9 +128,10 @@ bool DtlsSrtpTransport::IsDtlsConnected() {
}
bool DtlsSrtpTransport::IsDtlsWritable() {
auto rtp_packet_transport = srtp_transport_->rtp_packet_transport();
auto rtcp_packet_transport =
rtcp_mux_enabled() ? nullptr : rtcp_dtls_transport_;
return rtp_dtls_transport_ && rtp_dtls_transport_->writable() &&
rtcp_mux_enabled() ? nullptr : srtp_transport_->rtcp_packet_transport();
return rtp_packet_transport && rtp_packet_transport->writable() &&
(!rtcp_packet_transport || rtcp_packet_transport->writable());
}
@ -155,10 +170,11 @@ void DtlsSrtpTransport::SetupRtpDtlsSrtp() {
if (!ExtractParams(rtp_dtls_transport_, &selected_crypto_suite, &send_key,
&recv_key) ||
!SetRtpParams(selected_crypto_suite, &send_key[0],
static_cast<int>(send_key.size()), send_extension_ids,
selected_crypto_suite, &recv_key[0],
static_cast<int>(recv_key.size()), recv_extension_ids)) {
!srtp_transport_->SetRtpParams(
selected_crypto_suite, &send_key[0],
static_cast<int>(send_key.size()), send_extension_ids,
selected_crypto_suite, &recv_key[0],
static_cast<int>(recv_key.size()), recv_extension_ids)) {
SignalDtlsSrtpSetupFailure(this, /*rtcp=*/false);
RTC_LOG(LS_WARNING) << "DTLS-SRTP key installation for RTP failed";
}
@ -186,11 +202,11 @@ void DtlsSrtpTransport::SetupRtcpDtlsSrtp() {
rtc::ZeroOnFreeBuffer<unsigned char> rtcp_recv_key;
if (!ExtractParams(rtcp_dtls_transport_, &selected_crypto_suite,
&rtcp_send_key, &rtcp_recv_key) ||
!SetRtcpParams(selected_crypto_suite, &rtcp_send_key[0],
static_cast<int>(rtcp_send_key.size()), send_extension_ids,
selected_crypto_suite, &rtcp_recv_key[0],
static_cast<int>(rtcp_recv_key.size()),
recv_extension_ids)) {
!srtp_transport_->SetRtcpParams(
selected_crypto_suite, &rtcp_send_key[0],
static_cast<int>(rtcp_send_key.size()), send_extension_ids,
selected_crypto_suite, &rtcp_recv_key[0],
static_cast<int>(rtcp_recv_key.size()), recv_extension_ids)) {
SignalDtlsSrtpSetupFailure(this, /*rtcp=*/true);
RTC_LOG(LS_WARNING) << "DTLS-SRTP key installation for RTCP failed";
}
@ -313,20 +329,37 @@ void DtlsSrtpTransport::OnDtlsState(cricket::DtlsTransportInternal* transport,
transport == rtcp_dtls_transport_);
if (state != cricket::DTLS_TRANSPORT_CONNECTED) {
ResetParams();
srtp_transport_->ResetParams();
return;
}
MaybeSetupDtlsSrtp();
}
void DtlsSrtpTransport::OnWritableState(
rtc::PacketTransportInternal* packet_transport) {
bool writable = IsTransportWritable();
void DtlsSrtpTransport::OnWritableState(bool writable) {
SetWritable(writable);
if (writable) {
MaybeSetupDtlsSrtp();
}
}
void DtlsSrtpTransport::OnSentPacket(const rtc::SentPacket& sent_packet) {
SignalSentPacket(sent_packet);
}
void DtlsSrtpTransport::OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
SignalPacketReceived(rtcp, packet, packet_time);
}
void DtlsSrtpTransport::OnReadyToSend(bool ready) {
SignalReadyToSend(ready);
}
void DtlsSrtpTransport::OnNetworkRouteChanged(
rtc::Optional<rtc::NetworkRoute> network_route) {
SignalNetworkRouteChanged(network_route);
}
} // namespace webrtc

33
pc/dtlssrtptransport.h
View File

@ -16,17 +16,20 @@
#include <vector>
#include "p2p/base/dtlstransportinternal.h"
#include "pc/rtptransportinternaladapter.h"
#include "pc/srtptransport.h"
#include "rtc_base/buffer.h"
namespace webrtc {
// The subclass of SrtpTransport is used for DTLS-SRTP. When the DTLS handshake
// is finished, it extracts the keying materials from DtlsTransport and
// configures the SrtpSessions in the base class.
class DtlsSrtpTransport : public SrtpTransport {
// This class is intended to be used as an RtpTransport and it wraps both an
// SrtpTransport and DtlsTransports(RTP/RTCP). When the DTLS handshake is
// finished, it extracts the keying materials from DtlsTransport and sets them
// to SrtpTransport.
class DtlsSrtpTransport : public RtpTransportInternalAdapter {
public:
explicit DtlsSrtpTransport(bool rtcp_mux_enabled);
explicit DtlsSrtpTransport(
std::unique_ptr<webrtc::SrtpTransport> srtp_transport);
// Set P2P layer RTP/RTCP DtlsTransports. When using RTCP-muxing,
// |rtcp_dtls_transport| is null.
@ -42,6 +45,15 @@ class DtlsSrtpTransport : public SrtpTransport {
void UpdateRecvEncryptedHeaderExtensionIds(
const std::vector<int>& recv_extension_ids);
bool IsActive() { return srtp_transport_->IsActive(); }
// Cache RTP Absoulute SendTime extension header ID. This is only used when
// external authentication is enabled.
void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) {
srtp_transport_->CacheRtpAbsSendTimeHeaderExtension(
rtp_abs_sendtime_extn_id);
}
// TODO(zhihuang): Remove this when we remove RtpTransportAdapter.
RtpTransportAdapter* GetInternal() override { return nullptr; }
@ -71,11 +83,16 @@ class DtlsSrtpTransport : public SrtpTransport {
void OnDtlsState(cricket::DtlsTransportInternal* dtls_transport,
cricket::DtlsTransportState state);
// Override the RtpTransport::OnWritableState.
void OnWritableState(rtc::PacketTransportInternal* packet_transport) override;
void OnWritableState(bool writable);
void OnSentPacket(const rtc::SentPacket& sent_packet);
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void OnReadyToSend(bool ready);
void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route);
bool writable_ = false;
std::unique_ptr<SrtpTransport> srtp_transport_;
// Owned by the TransportController.
cricket::DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
cricket::DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;

69
pc/dtlssrtptransport_unittest.cc
View File

@ -33,26 +33,50 @@ using webrtc::RtpTransport;
const int kRtpAuthTagLen = 10;
class TransportObserver : public sigslot::has_slots<> {
public:
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
rtcp ? last_recv_rtcp_packet_ = *packet : last_recv_rtp_packet_ = *packet;
}
void OnReadyToSend(bool ready) { ready_to_send_ = ready; }
rtc::CopyOnWriteBuffer last_recv_rtp_packet() {
return last_recv_rtp_packet_;
}
rtc::CopyOnWriteBuffer last_recv_rtcp_packet() {
return last_recv_rtcp_packet_;
}
bool ready_to_send() { return ready_to_send_; }
private:
rtc::CopyOnWriteBuffer last_recv_rtp_packet_;
rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
bool ready_to_send_ = false;
};
class DtlsSrtpTransportTest : public testing::Test,
public sigslot::has_slots<> {
protected:
DtlsSrtpTransportTest() {}
~DtlsSrtpTransportTest() {
if (dtls_srtp_transport1_) {
dtls_srtp_transport1_->UnregisterRtpDemuxerSink(&transport_observer1_);
}
if (dtls_srtp_transport2_) {
dtls_srtp_transport2_->UnregisterRtpDemuxerSink(&transport_observer2_);
}
}
std::unique_ptr<DtlsSrtpTransport> MakeDtlsSrtpTransport(
FakeDtlsTransport* rtp_dtls,
FakeDtlsTransport* rtcp_dtls,
bool rtcp_mux_enabled) {
auto rtp_transport = rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled);
rtp_transport->AddHandledPayloadType(0x00);
rtp_transport->AddHandledPayloadType(0xc9);
auto srtp_transport =
rtc::MakeUnique<SrtpTransport>(std::move(rtp_transport));
auto dtls_srtp_transport =
rtc::MakeUnique<DtlsSrtpTransport>(rtcp_mux_enabled);
rtc::MakeUnique<DtlsSrtpTransport>(std::move(srtp_transport));
dtls_srtp_transport->SetDtlsTransports(rtp_dtls, rtcp_dtls);
@ -69,24 +93,15 @@ class DtlsSrtpTransportTest : public testing::Test,
dtls_srtp_transport2_ =
MakeDtlsSrtpTransport(rtp_dtls2, rtcp_dtls2, rtcp_mux_enabled);
dtls_srtp_transport1_->SignalRtcpPacketReceived.connect(
&transport_observer1_,
&webrtc::TransportObserver::OnRtcpPacketReceived);
dtls_srtp_transport1_->SignalPacketReceived.connect(
&transport_observer1_, &TransportObserver::OnPacketReceived);
dtls_srtp_transport1_->SignalReadyToSend.connect(
&transport_observer1_, &webrtc::TransportObserver::OnReadyToSend);
&transport_observer1_, &TransportObserver::OnReadyToSend);
dtls_srtp_transport2_->SignalRtcpPacketReceived.connect(
&transport_observer2_,
&webrtc::TransportObserver::OnRtcpPacketReceived);
dtls_srtp_transport2_->SignalPacketReceived.connect(
&transport_observer2_, &TransportObserver::OnPacketReceived);
dtls_srtp_transport2_->SignalReadyToSend.connect(
&transport_observer2_, &webrtc::TransportObserver::OnReadyToSend);
webrtc::RtpDemuxerCriteria demuxer_criteria;
// 0x00 is the payload type used in kPcmuFrame.
demuxer_criteria.payload_types = {0x00};
dtls_srtp_transport1_->RegisterRtpDemuxerSink(demuxer_criteria,
&transport_observer1_);
dtls_srtp_transport2_->RegisterRtpDemuxerSink(demuxer_criteria,
&transport_observer2_);
&transport_observer2_, &TransportObserver::OnReadyToSend);
}
void CompleteDtlsHandshake(FakeDtlsTransport* fake_dtls1,
@ -236,8 +251,8 @@ class DtlsSrtpTransportTest : public testing::Test,
std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport1_;
std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport2_;
webrtc::TransportObserver transport_observer1_;
webrtc::TransportObserver transport_observer2_;
TransportObserver transport_observer1_;
TransportObserver transport_observer2_;
int sequence_number_ = 0;
};

12
pc/jseptransport2_unittest.cc
View File

@ -43,8 +43,9 @@ class JsepTransport2Test : public testing::Test, public sigslot::has_slots<> {
const std::string& transport_name,
rtc::PacketTransportInternal* rtp_packet_transport,
rtc::PacketTransportInternal* rtcp_packet_transport) {
auto srtp_transport = rtc::MakeUnique<webrtc::SrtpTransport>(
rtcp_packet_transport == nullptr);
bool rtcp_mux_enabled = (rtcp_packet_transport == nullptr);
auto srtp_transport =
rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_enabled);
srtp_transport->SetRtpPacketTransport(rtp_packet_transport);
if (rtcp_packet_transport) {
@ -57,8 +58,11 @@ class JsepTransport2Test : public testing::Test, public sigslot::has_slots<> {
const std::string& transport_name,
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport) {
auto dtls_srtp_transport = rtc::MakeUnique<webrtc::DtlsSrtpTransport>(
rtcp_dtls_transport == nullptr);
bool rtcp_mux_enabled = (rtcp_dtls_transport == nullptr);
auto srtp_transport =
rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_enabled);
auto dtls_srtp_transport =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(srtp_transport));
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
rtcp_dtls_transport);

11
pc/jseptransportcontroller.cc
View File

@ -457,13 +457,16 @@ JsepTransportController::CreateDtlsSrtpTransport(
cricket::DtlsTransportInternal* rtp_dtls_transport,
cricket::DtlsTransportInternal* rtcp_dtls_transport) {
RTC_DCHECK(network_thread_->IsCurrent());
auto dtls_srtp_transport = rtc::MakeUnique<webrtc::DtlsSrtpTransport>(
rtcp_dtls_transport == nullptr);
bool rtcp_mux_enabled = rtcp_dtls_transport == nullptr;
auto srtp_transport =
rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_enabled);
if (config_.enable_external_auth) {
dtls_srtp_transport->EnableExternalAuth();
srtp_transport->EnableExternalAuth();
}
auto dtls_srtp_transport =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(srtp_transport));
dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
rtcp_dtls_transport);
return dtls_srtp_transport;

5
pc/peerconnection.cc
View File

@ -6195,9 +6195,12 @@ void PeerConnection::DestroyDataChannel() {
void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
RTC_DCHECK(channel);
RTC_DCHECK(channel->rtp_dtls_transport());
// Need to cache these before destroying the base channel so that we do not
// access uninitialized memory.
const std::string transport_name = channel->transport_name();
const std::string transport_name =
channel->rtp_dtls_transport()->transport_name();
const bool need_to_delete_rtcp = (channel->rtcp_dtls_transport() != nullptr);
switch (channel->media_type()) {

8
pc/peerconnection_media_unittest.cc
View File

@ -959,8 +959,8 @@ void RenameContent(cricket::SessionDescription* desc,
// Tests that an answer responds with the same MIDs as the offer.
TEST_P(PeerConnectionMediaTest, AnswerHasSameMidsAsOffer) {
const std::string kAudioMid = "notdefault1";
const std::string kVideoMid = "notdefault2";
const std::string kAudioMid = "not default1";
const std::string kVideoMid = "not default2";
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();
@ -980,8 +980,8 @@ TEST_P(PeerConnectionMediaTest, AnswerHasSameMidsAsOffer) {
// Test that if the callee creates a re-offer, the MIDs are the same as the
// original offer.
TEST_P(PeerConnectionMediaTest, ReOfferHasSameMidsAsFirstOffer) {
const std::string kAudioMid = "notdefault1";
const std::string kVideoMid = "notdefault2";
const std::string kAudioMid = "not default1";
const std::string kVideoMid = "not default2";
auto caller = CreatePeerConnectionWithAudioVideo();
auto callee = CreatePeerConnectionWithAudioVideo();

136
pc/rtptransport.cc
View File

@ -10,10 +10,7 @@
#include "pc/rtptransport.h"
#include <utility>
#include "media/base/rtputils.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "p2p/base/p2pconstants.h"
#include "p2p/base/packettransportinterface.h"
#include "rtc_base/checks.h"
@ -47,7 +44,7 @@ void RtpTransport::SetRtpPacketTransport(
new_packet_transport->SignalReadPacket.connect(this,
&RtpTransport::OnReadPacket);
new_packet_transport->SignalNetworkRouteChanged.connect(
this, &RtpTransport::OnNetworkRouteChanged);
this, &RtpTransport::OnNetworkRouteChange);
new_packet_transport->SignalWritableState.connect(
this, &RtpTransport::OnWritableState);
new_packet_transport->SignalSentPacket.connect(this,
@ -83,7 +80,7 @@ void RtpTransport::SetRtcpPacketTransport(
new_packet_transport->SignalReadPacket.connect(this,
&RtpTransport::OnReadPacket);
new_packet_transport->SignalNetworkRouteChanged.connect(
this, &RtpTransport::OnNetworkRouteChanged);
this, &RtpTransport::OnNetworkRouteChange);
new_packet_transport->SignalWritableState.connect(
this, &RtpTransport::OnWritableState);
new_packet_transport->SignalSentPacket.connect(this,
@ -137,27 +134,16 @@ bool RtpTransport::SendPacket(bool rtcp,
return true;
}
void RtpTransport::UpdateRtpHeaderExtensionMap(
const cricket::RtpHeaderExtensions& header_extensions) {
header_extension_map_ = RtpHeaderExtensionMap(header_extensions);
bool RtpTransport::HandlesPacket(const uint8_t* data, size_t len) {
return bundle_filter_.DemuxPacket(data, len);
}
bool RtpTransport::RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
RtpPacketSinkInterface* sink) {
rtp_demuxer_.RemoveSink(sink);
if (!rtp_demuxer_.AddSink(criteria, sink)) {
RTC_LOG(LS_ERROR) << "Failed to register the sink for RTP demuxer.";
return false;
}
return true;
bool RtpTransport::HandlesPayloadType(int payload_type) const {
return bundle_filter_.FindPayloadType(payload_type);
}
bool RtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
if (!rtp_demuxer_.RemoveSink(sink)) {
RTC_LOG(LS_ERROR) << "Failed to unregister the sink for RTP demuxer.";
return false;
}
return true;
void RtpTransport::AddHandledPayloadType(int payload_type) {
bundle_filter_.AddPayloadType(payload_type);
}
PacketTransportInterface* RtpTransport::GetRtpPacketTransport() const {
@ -194,26 +180,11 @@ RtpTransportParameters RtpTransport::GetParameters() const {
return parameters_;
}
void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& time) {
webrtc::RtpPacketReceived parsed_packet(&header_extension_map_);
if (!parsed_packet.Parse(std::move(*packet))) {
RTC_LOG(LS_ERROR)
<< "Failed to parse the incoming RTP packet before demuxing. Drop it.";
return;
}
if (time.timestamp != -1) {
parsed_packet.set_arrival_time_ms((time.timestamp + 500) / 1000);
}
rtp_demuxer_.OnRtpPacket(parsed_packet);
}
RtpTransportAdapter* RtpTransport::GetInternal() {
return nullptr;
}
bool RtpTransport::IsTransportWritable() {
bool RtpTransport::IsRtpTransportWritable() {
auto rtcp_packet_transport =
rtcp_mux_enabled_ ? nullptr : rtcp_packet_transport_;
return rtp_packet_transport_ && rtp_packet_transport_->writable() &&
@ -224,7 +195,7 @@ void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
SetReadyToSend(transport == rtcp_packet_transport_, true);
}
void RtpTransport::OnNetworkRouteChanged(
void RtpTransport::OnNetworkRouteChange(
rtc::Optional<rtc::NetworkRoute> network_route) {
SignalNetworkRouteChanged(network_route);
}
@ -233,7 +204,7 @@ void RtpTransport::OnWritableState(
rtc::PacketTransportInternal* packet_transport) {
RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
packet_transport == rtcp_packet_transport_);
SignalWritableState(IsTransportWritable());
SignalWritableState(IsRtpTransportWritable());
}
void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
@ -243,44 +214,6 @@ void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
SignalSentPacket(sent_packet);
}
void RtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
DemuxPacket(packet, packet_time);
}
void RtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
SignalRtcpPacketReceived(packet, packet_time);
}
void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const rtc::PacketTime& packet_time,
int flags) {
TRACE_EVENT0("webrtc", "RtpTransport::OnReadPacket");
// When using RTCP multiplexing we might get RTCP packets on the RTP
// transport. We check the RTP payload type to determine if it is RTCP.
bool rtcp = transport == rtcp_packet_transport() ||
cricket::IsRtcp(data, static_cast<int>(len));
rtc::CopyOnWriteBuffer packet(data, len);
// Protect ourselves against crazy data.
if (!cricket::IsValidRtpRtcpPacketSize(rtcp, packet.size())) {
RTC_LOG(LS_ERROR) << "Dropping incoming "
<< cricket::RtpRtcpStringLiteral(rtcp)
<< " packet: wrong size=" << packet.size();
return;
}
if (rtcp) {
OnRtcpPacketReceived(&packet, packet_time);
} else {
OnRtpPacketReceived(&packet, packet_time);
}
}
void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
if (rtcp) {
rtcp_ready_to_send_ = ready;
@ -300,4 +233,51 @@ void RtpTransport::MaybeSignalReadyToSend() {
}
}
// Check the RTP payload type. If 63 < payload type < 96, it's RTCP.
// For additional details, see http://tools.ietf.org/html/rfc5761.
bool IsRtcp(const char* data, int len) {
if (len < 2) {
return false;
}
char pt = data[1] & 0x7F;
return (63 < pt) && (pt < 96);
}
void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const rtc::PacketTime& packet_time,
int flags) {
TRACE_EVENT0("webrtc", "RtpTransport::OnReadPacket");
// When using RTCP multiplexing we might get RTCP packets on the RTP
// transport. We check the RTP payload type to determine if it is RTCP.
bool rtcp = transport == rtcp_packet_transport() ||
IsRtcp(data, static_cast<int>(len));
rtc::CopyOnWriteBuffer packet(data, len);
if (!WantsPacket(rtcp, &packet)) {
return;
}
// This mutates |packet| if it is protected.
SignalPacketReceived(rtcp, &packet, packet_time);
}
bool RtpTransport::WantsPacket(bool rtcp,
const rtc::CopyOnWriteBuffer* packet) {
// Protect ourselves against crazy data.
if (!packet || !cricket::IsValidRtpRtcpPacketSize(rtcp, packet->size())) {
RTC_LOG(LS_ERROR) << "Dropping incoming "
<< cricket::RtpRtcpStringLiteral(rtcp)
<< " packet: wrong size=" << packet->size();
return false;
}
if (rtcp) {
// Permit all (seemingly valid) RTCP packets.
return true;
}
// Check whether we handle this payload.
return HandlesPacket(packet->data(), packet->size());
}
} // namespace webrtc

68
pc/rtptransport.h
View File

@ -13,8 +13,7 @@
#include <string>
#include "call/rtp_demuxer.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "pc/bundlefilter.h"
#include "pc/rtptransportinternal.h"
#include "rtc_base/sigslot.h"
@ -67,13 +66,9 @@ class RtpTransport : public RtpTransportInternal {
const rtc::PacketOptions& options,
int flags) override;
void UpdateRtpHeaderExtensionMap(
const cricket::RtpHeaderExtensions& header_extensions) override;
bool HandlesPayloadType(int payload_type) const override;
bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
RtpPacketSinkInterface* sink) override;
bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
void AddHandledPayloadType(int payload_type) override;
void SetMetricsObserver(
rtc::scoped_refptr<MetricsObserverInterface> metrics_observer) override {}
@ -82,35 +77,15 @@ class RtpTransport : public RtpTransportInternal {
// TODO(zstein): Remove this when we remove RtpTransportAdapter.
RtpTransportAdapter* GetInternal() override;
// These methods will be used in the subclasses.
void DemuxPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& time);
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags);
bool IsTransportWritable();
// Overridden by SrtpTransport.
virtual void OnNetworkRouteChanged(
rtc::Optional<rtc::NetworkRoute> network_route);
virtual void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
virtual void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
// Overridden by DtlsSrtpTransport.
virtual void OnWritableState(rtc::PacketTransportInternal* packet_transport);
private:
bool IsRtpTransportWritable();
bool HandlesPacket(const uint8_t* data, size_t len);
void OnReadyToSend(rtc::PacketTransportInternal* transport);
void OnNetworkRouteChange(rtc::Optional<rtc::NetworkRoute> network_route);
void OnWritableState(rtc::PacketTransportInternal* packet_transport);
void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
const rtc::SentPacket& sent_packet);
void OnReadPacket(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const rtc::PacketTime& packet_time,
int flags);
// Updates "ready to send" for an individual channel and fires
// SignalReadyToSend.
@ -118,17 +93,18 @@ class RtpTransport : public RtpTransportInternal {
void MaybeSignalReadyToSend();
// SRTP specific methods.
// TODO(zhihuang): Improve the inheritance model so that the RtpTransport
// doesn't need to implement SRTP specfic methods.
RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override {
RTC_NOTREACHED();
return RTCError::OK();
}
RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override {
RTC_NOTREACHED();
return RTCError::OK();
}
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags);
void OnReadPacket(rtc::PacketTransportInternal* transport,
const char* data,
size_t len,
const rtc::PacketTime& packet_time,
int flags);
bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
bool rtcp_mux_enabled_;
@ -140,10 +116,8 @@ class RtpTransport : public RtpTransportInternal {
bool rtcp_ready_to_send_ = false;
RtpTransportParameters parameters_;
RtpDemuxer rtp_demuxer_;
// Used for identifying the MID for RtpDemuxer.
RtpHeaderExtensionMap header_extension_map_;
cricket::BundleFilter bundle_filter_;
};
} // namespace webrtc

42
pc/rtptransport_unittest.cc
View File

@ -197,37 +197,49 @@ TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
EXPECT_FALSE(observer.network_route());
}
class SignalCounter : public sigslot::has_slots<> {
public:
explicit SignalCounter(RtpTransport* transport) {
transport->SignalReadyToSend.connect(this, &SignalCounter::OnReadyToSend);
}
int count() const { return count_; }
void OnReadyToSend(bool ready) { ++count_; }
private:
int count_ = 0;
};
TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
RtpTransport transport(kMuxEnabled);
TransportObserver observer(&transport);
SignalCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetWritable(true);
// State changes, so we should signal.
transport.SetRtpPacketTransport(&fake_rtp);
EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
EXPECT_EQ(observer.count(), 1);
// State does not change, so we should not signal.
transport.SetRtpPacketTransport(&fake_rtp);
EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
EXPECT_EQ(observer.count(), 1);
// State does not change, so we should not signal.
transport.SetRtcpMuxEnabled(true);
EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
EXPECT_EQ(observer.count(), 1);
// State changes, so we should signal.
transport.SetRtcpMuxEnabled(false);
EXPECT_EQ(observer.ready_to_send_signal_count(), 2);
EXPECT_EQ(observer.count(), 2);
}
// Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
// received.
TEST(RtpTransportTest, SignalDemuxedRtcp) {
RtpTransport transport(kMuxDisabled);
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
TransportObserver observer(&transport);
// An rtcp packet.
const char data[] = {0, 73, 0, 0};
@ -247,15 +259,11 @@ static const int kRtpLen = 12;
// handled payload type is received.
TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
RtpTransport transport(kMuxDisabled);
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
// Disable the encryption to allow raw RTP data.
transport.SetRtpPacketTransport(&fake_rtp);
TransportObserver observer(&transport);
RtpDemuxerCriteria demuxer_criteria;
// Add a handled payload type.
demuxer_criteria.payload_types = {0x11};
transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
transport.AddHandledPayloadType(0x11);
// An rtp packet.
const rtc::PacketOptions options;
@ -264,22 +272,16 @@ TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
EXPECT_EQ(1, observer.rtp_count());
EXPECT_EQ(0, observer.rtcp_count());
// Remove the sink before destroying the transport.
transport.UnregisterRtpDemuxerSink(&observer);
}
// Test that SignalPacketReceived does not fire when a RTP packet with an
// unhandled payload type is received.
TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
RtpTransport transport(kMuxDisabled);
SignalPacketReceivedCounter observer(&transport);
rtc::FakePacketTransport fake_rtp("fake_rtp");
fake_rtp.SetDestination(&fake_rtp, true);
transport.SetRtpPacketTransport(&fake_rtp);
TransportObserver observer(&transport);
RtpDemuxerCriteria demuxer_criteria;
// Add an unhandled payload type.
demuxer_criteria.payload_types = {0x12};
transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
const rtc::PacketOptions options;
const int flags = 0;
@ -287,8 +289,6 @@ TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
EXPECT_EQ(0, observer.rtp_count());
EXPECT_EQ(0, observer.rtcp_count());
// Remove the sink before destroying the transport.
transport.UnregisterRtpDemuxerSink(&observer);
}
} // namespace webrtc

34
pc/rtptransportinternal.h
View File

@ -13,11 +13,9 @@
#include <string>
#include "api/ortc/srtptransportinterface.h"
#include "api/ortc/rtptransportinterface.h"
#include "api/umametrics.h"
#include "call/rtp_demuxer.h"
#include "p2p/base/icetransportinternal.h"
#include "pc/sessiondescription.h"
#include "rtc_base/networkroute.h"
#include "rtc_base/sigslot.h"
@ -29,11 +27,11 @@ struct PacketTime;
namespace webrtc {
// This represents the internal interface beneath SrtpTransportInterface;
// This represents the internal interface beneath RtpTransportInterface;
// it is not accessible to API consumers but is accessible to internal classes
// in order to send and receive RTP and RTCP packets belonging to a single RTP
// session. Additional convenience and configuration methods are also provided.
class RtpTransportInternal : public SrtpTransportInterface,
class RtpTransportInternal : public RtpTransportInterface,
public sigslot::has_slots<> {
public:
virtual void SetRtcpMuxEnabled(bool enable) = 0;
@ -54,11 +52,11 @@ class RtpTransportInternal : public SrtpTransportInterface,
// than just "writable"; it means the last send didn't return ENOTCONN.
sigslot::signal1<bool> SignalReadyToSend;
// Called whenever an RTCP packet is received. There is no equivalent signal
// for RTP packets because they would be forwarded to the BaseChannel through
// the RtpDemuxer callback.
sigslot::signal2<rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
SignalRtcpPacketReceived;
// TODO(zstein): Consider having two signals - RtpPacketReceived and
// RtcpPacketReceived.
// The first argument is true for RTCP packets and false for RTP packets.
sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
SignalPacketReceived;
// Called whenever the network route of the P2P layer transport changes.
// The argument is an optional network route.
@ -82,22 +80,12 @@ class RtpTransportInternal : public SrtpTransportInterface,
const rtc::PacketOptions& options,
int flags) = 0;
// This method updates the RTP header extension map so that the RTP transport
// can parse the received packets and identify the MID. This is called by the
// BaseChannel when setting the content description.
//
// Note: This doesn't take the BUNDLE case in account meaning the RTP header
// extension maps are not merged when BUNDLE is enabled. This is fine because
// the ID for MID should be consistent among all the RTP transports.
virtual void UpdateRtpHeaderExtensionMap(
const cricket::RtpHeaderExtensions& header_extensions) = 0;
virtual bool HandlesPayloadType(int payload_type) const = 0;
virtual void AddHandledPayloadType(int payload_type) = 0;
virtual void SetMetricsObserver(
rtc::scoped_refptr<MetricsObserverInterface> metrics_observer) = 0;
virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
RtpPacketSinkInterface* sink) = 0;
virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
};
} // namespace webrtc

18
pc/rtptransportinternaladapter.h
View File

@ -67,22 +67,12 @@ class RtpTransportInternalAdapter : public RtpTransportInternal {
return transport_->SendRtcpPacket(packet, options, flags);
}
void UpdateRtpHeaderExtensionMap(
const cricket::RtpHeaderExtensions& header_extensions) override {
transport_->UpdateRtpHeaderExtensionMap(header_extensions);
bool HandlesPayloadType(int payload_type) const override {
return transport_->HandlesPayloadType(payload_type);
}
void RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
RtpPacketSinkInterface* sink) override {
transport_->RegisterRtpDemuxerSink(criteria, sink);
}
void UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override {
transport_->UnregisterRtpDemuxerSink(sink);
}
void SetEncryptionDisabled(bool encryption_disabled) override {
transport_->SetEncryptionDisabled(encryption_disabled);
void AddHandledPayloadType(int payload_type) override {
return transport_->AddHandledPayloadType(payload_type);
}
// RtpTransportInterface overrides.

65
pc/rtptransporttestutil.h
View File

@ -11,67 +11,32 @@
#ifndef PC_RTPTRANSPORTTESTUTIL_H_
#define PC_RTPTRANSPORTTESTUTIL_H_
#include "call/rtp_packet_sink_interface.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "pc/rtptransportinternal.h"
#include "rtc_base/sigslot.h"
namespace webrtc {
// Used to handle the signals when the RtpTransport receives an RTP/RTCP packet.
// Used in Rtp/Srtp/DtlsTransport unit tests.
class TransportObserver : public RtpPacketSinkInterface,
public sigslot::has_slots<> {
class SignalPacketReceivedCounter : public sigslot::has_slots<> {
public:
TransportObserver() {}
explicit TransportObserver(RtpTransportInternal* rtp_transport) {
rtp_transport->SignalRtcpPacketReceived.connect(
this, &TransportObserver::OnRtcpPacketReceived);
rtp_transport->SignalReadyToSend.connect(this,
&TransportObserver::OnReadyToSend);
explicit SignalPacketReceivedCounter(RtpTransportInternal* transport) {
transport->SignalPacketReceived.connect(
this, &SignalPacketReceivedCounter::OnPacketReceived);
}
// RtpPacketInterface override.
void OnRtpPacket(const RtpPacketReceived& packet) override {
rtp_count_++;
last_recv_rtp_packet_ = packet.Buffer();
}
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
RTC_LOG(LS_INFO) << "Received an RTCP packet.";
rtcp_count_++;
last_recv_rtcp_packet_ = *packet;
}
int rtp_count() const { return rtp_count_; }
int rtcp_count() const { return rtcp_count_; }
rtc::CopyOnWriteBuffer last_recv_rtp_packet() {
return last_recv_rtp_packet_;
}
rtc::CopyOnWriteBuffer last_recv_rtcp_packet() {
return last_recv_rtcp_packet_;
}
void OnReadyToSend(bool ready) {
ready_to_send_signal_count_++;
ready_to_send_ = ready;
}
bool ready_to_send() { return ready_to_send_; }
int ready_to_send_signal_count() { return ready_to_send_signal_count_; }
int rtp_count() const { return rtp_count_; }
private:
bool ready_to_send_ = false;
int rtp_count_ = 0;
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer*,
const rtc::PacketTime&) {
if (rtcp) {
++rtcp_count_;
} else {
++rtp_count_;
}
}
int rtcp_count_ = 0;
int ready_to_send_signal_count_ = 0;
rtc::CopyOnWriteBuffer last_recv_rtp_packet_;
rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
int rtp_count_ = 0;
};
} // namespace webrtc

195
pc/srtptransport.cc
View File

@ -25,133 +25,155 @@
namespace webrtc {
SrtpTransport::SrtpTransport(bool rtcp_mux_enabled)
: RtpTransport(rtcp_mux_enabled) {}
: RtpTransportInternalAdapter(new RtpTransport(rtcp_mux_enabled)) {
// Own the raw pointer |transport| from the base class.
rtp_transport_.reset(transport_);
RTC_DCHECK(rtp_transport_);
ConnectToRtpTransport();
}
SrtpTransport::SrtpTransport(
std::unique_ptr<RtpTransportInternal> rtp_transport)
: RtpTransportInternalAdapter(rtp_transport.get()),
rtp_transport_(std::move(rtp_transport)) {
RTC_DCHECK(rtp_transport_);
ConnectToRtpTransport();
}
void SrtpTransport::ConnectToRtpTransport() {
rtp_transport_->SignalPacketReceived.connect(
this, &SrtpTransport::OnPacketReceived);
rtp_transport_->SignalReadyToSend.connect(this,
&SrtpTransport::OnReadyToSend);
rtp_transport_->SignalNetworkRouteChanged.connect(
this, &SrtpTransport::OnNetworkRouteChanged);
rtp_transport_->SignalWritableState.connect(this,
&SrtpTransport::OnWritableState);
rtp_transport_->SignalSentPacket.connect(this, &SrtpTransport::OnSentPacket);
}
bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
return SendPacket(false, packet, options, flags);
}
bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
return SendPacket(true, packet, options, flags);
}
bool SrtpTransport::SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
if (!IsActive()) {
RTC_LOG(LS_ERROR)
<< "Failed to send the packet because SRTP transport is inactive.";
return false;
}
rtc::PacketOptions updated_options = options;
rtc::CopyOnWriteBuffer cp = *packet;
TRACE_EVENT0("webrtc", "SRTP Encode");
bool res;
uint8_t* data = packet->data();
int len = static_cast<int>(packet->size());
if (!rtcp) {
// If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
// inside libsrtp for a RTP packet. A external HMAC module will be writing
// a fake HMAC value. This is ONLY done for a RTP packet.
// Socket layer will update rtp sendtime extension header if present in
// packet with current time before updating the HMAC.
#if !defined(ENABLE_EXTERNAL_AUTH)
res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
#else
if (!IsExternalAuthActive()) {
res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
} else {
updated_options.packet_time_params.rtp_sendtime_extension_id =
rtp_abs_sendtime_extn_id_;
res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len,
&updated_options.packet_time_params.srtp_packet_index);
// If protection succeeds, let's get auth params from srtp.
if (res) {
uint8_t* auth_key = NULL;
int key_len;
res = GetRtpAuthParams(
&auth_key, &key_len,
&updated_options.packet_time_params.srtp_auth_tag_len);
#else
if (!IsExternalAuthActive()) {
res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
} else {
updated_options.packet_time_params.rtp_sendtime_extension_id =
rtp_abs_sendtime_extn_id_;
res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len,
&updated_options.packet_time_params.srtp_packet_index);
// If protection succeeds, let's get auth params from srtp.
if (res) {
updated_options.packet_time_params.srtp_auth_key.resize(key_len);
updated_options.packet_time_params.srtp_auth_key.assign(
auth_key, auth_key + key_len);
uint8_t* auth_key = NULL;
int key_len;
res = GetRtpAuthParams(
&auth_key, &key_len,
&updated_options.packet_time_params.srtp_auth_tag_len);
if (res) {
updated_options.packet_time_params.srtp_auth_key.resize(key_len);
updated_options.packet_time_params.srtp_auth_key.assign(
auth_key, auth_key + key_len);
}
}
}
}
#endif
if (!res) {
int seq_num = -1;
uint32_t ssrc = 0;
cricket::GetRtpSeqNum(data, len, &seq_num);
cricket::GetRtpSsrc(data, len, &ssrc);
RTC_LOG(LS_ERROR) << "Failed to protect RTP packet: size=" << len
<< ", seqnum=" << seq_num << ", SSRC=" << ssrc;
return false;
if (!res) {
int seq_num = -1;
uint32_t ssrc = 0;
cricket::GetRtpSeqNum(data, len, &seq_num);
cricket::GetRtpSsrc(data, len, &ssrc);
RTC_LOG(LS_ERROR) << "Failed to protect RTP packet: size=" << len
<< ", seqnum=" << seq_num << ", SSRC=" << ssrc;
return false;
}
} else {
res = ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len);
if (!res) {
int type = -1;
cricket::GetRtcpType(data, len, &type);
RTC_LOG(LS_ERROR) << "Failed to protect RTCP packet: size=" << len
<< ", type=" << type;
return false;
}
}
// Update the length of the packet now that we've added the auth tag.
packet->SetSize(len);
return SendPacket(/*rtcp=*/false, packet, updated_options, flags);
return rtcp ? rtp_transport_->SendRtcpPacket(packet, updated_options, flags)
: rtp_transport_->SendRtpPacket(packet, updated_options, flags);
}
bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) {
if (!IsActive()) {
RTC_LOG(LS_ERROR)
<< "Failed to send the packet because SRTP transport is inactive.";
return false;
}
TRACE_EVENT0("webrtc", "SRTP Encode");
uint8_t* data = packet->data();
int len = static_cast<int>(packet->size());
if (!ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len)) {
int type = -1;
cricket::GetRtcpType(data, len, &type);
RTC_LOG(LS_ERROR) << "Failed to protect RTCP packet: size=" << len
<< ", type=" << type;
return false;
}
// Update the length of the packet now that we've added the auth tag.
packet->SetSize(len);
return SendPacket(/*rtcp=*/true, packet, options, flags);
}
void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
void SrtpTransport::OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
if (!IsActive()) {
RTC_LOG(LS_WARNING)
<< "Inactive SRTP transport received an RTP packet. Drop it.";
<< "Inactive SRTP transport received a packet. Drop it.";
return;
}
TRACE_EVENT0("webrtc", "SRTP Decode");
char* data = packet->data<char>();
int len = static_cast<int>(packet->size());
if (!UnprotectRtp(data, len, &len)) {
int seq_num = -1;
uint32_t ssrc = 0;
cricket::GetRtpSeqNum(data, len, &seq_num);
cricket::GetRtpSsrc(data, len, &ssrc);
RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
<< ", seqnum=" << seq_num << ", SSRC=" << ssrc;
return;
bool res;
if (!rtcp) {
res = UnprotectRtp(data, len, &len);
if (!res) {
int seq_num = -1;
uint32_t ssrc = 0;
cricket::GetRtpSeqNum(data, len, &seq_num);
cricket::GetRtpSsrc(data, len, &ssrc);
RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
<< ", seqnum=" << seq_num << ", SSRC=" << ssrc;
return;
}
} else {
res = UnprotectRtcp(data, len, &len);
if (!res) {
int type = -1;
cricket::GetRtcpType(data, len, &type);
RTC_LOG(LS_ERROR) << "Failed to unprotect RTCP packet: size=" << len
<< ", type=" << type;
return;
}
}
packet->SetSize(len);
DemuxPacket(packet, packet_time);
}
void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
if (!IsActive()) {
RTC_LOG(LS_WARNING)
<< "Inactive SRTP transport received an RTCP packet. Drop it.";
return;
}
TRACE_EVENT0("webrtc", "SRTP Decode");
char* data = packet->data<char>();
int len = static_cast<int>(packet->size());
if (!UnprotectRtcp(data, len, &len)) {
int type = -1;
cricket::GetRtcpType(data, len, &type);
RTC_LOG(LS_ERROR) << "Failed to unprotect RTCP packet: size=" << len
<< ", type=" << type;
return;
}
packet->SetSize(len);
SignalRtcpPacketReceived(packet, packet_time);
SignalPacketReceived(rtcp, packet, packet_time);
}
void SrtpTransport::OnNetworkRouteChanged(
@ -392,6 +414,7 @@ void SrtpTransport::SetMetricsObserver(
if (recv_rtcp_session_) {
recv_rtcp_session_->SetMetricsObserver(metrics_observer_);
}
rtp_transport_->SetMetricsObserver(metrics_observer);
}
} // namespace webrtc

44
pc/srtptransport.h
View File

@ -17,20 +17,21 @@
#include <vector>
#include "p2p/base/icetransportinternal.h"
#include "pc/rtptransport.h"
#include "pc/rtptransportinternaladapter.h"
#include "pc/srtpfilter.h"
#include "pc/srtpsession.h"
#include "rtc_base/checks.h"
namespace webrtc {
// This subclass of the RtpTransport is used for SRTP which is reponsible for
// protecting/unprotecting the packets. It provides interfaces to set the crypto
// parameters for the SrtpSession underneath.
class SrtpTransport : public RtpTransport {
// This class will eventually be a wrapper around RtpTransportInternal
// that protects and unprotects sent and received RTP packets.
class SrtpTransport : public RtpTransportInternalAdapter {
public:
explicit SrtpTransport(bool rtcp_mux_enabled);
explicit SrtpTransport(std::unique_ptr<RtpTransportInternal> rtp_transport);
bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags) override;
@ -39,16 +40,6 @@ class SrtpTransport : public RtpTransport {
const rtc::PacketOptions& options,
int flags) override;
// SrtpTransportInterface override.
// TODO(zhihuang): Implement these methods and replace the RtpTransportAdapter
// object.
RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override {
return RTCError::OK();
}
RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override {
return RTCError::OK();
}
// The transport becomes active if the send_session_ and recv_session_ are
// created.
bool IsActive() const;
@ -114,12 +105,22 @@ class SrtpTransport : public RtpTransport {
void ConnectToRtpTransport();
void CreateSrtpSessions();
void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) override;
void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) override;
void OnNetworkRouteChanged(
rtc::Optional<rtc::NetworkRoute> network_route) override;
bool SendPacket(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketOptions& options,
int flags);
void OnPacketReceived(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time);
void OnReadyToSend(bool ready) { SignalReadyToSend(ready); }
void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route);
void OnWritableState(bool writable) { SignalWritableState(writable); }
void OnSentPacket(const rtc::SentPacket& sent_packet) {
SignalSentPacket(sent_packet);
}
bool ProtectRtp(void* data, int in_len, int max_len, int* out_len);
@ -138,6 +139,7 @@ class SrtpTransport : public RtpTransport {
bool UnprotectRtcp(void* data, int in_len, int* out_len);
const std::string content_name_;
std::unique_ptr<RtpTransportInternal> rtp_transport_;
std::unique_ptr<cricket::SrtpSession> send_session_;
std::unique_ptr<cricket::SrtpSession> recv_session_;

91
pc/srtptransport_unittest.cc
View File

@ -42,6 +42,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
protected:
SrtpTransportTest() {
bool rtcp_mux_enabled = true;
auto rtp_transport1 = rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled);
auto rtp_transport2 = rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled);
rtp_packet_transport1_ =
rtc::MakeUnique<rtc::FakePacketTransport>("fake_packet_transport1");
@ -52,32 +54,38 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
rtp_packet_transport1_->SetDestination(rtp_packet_transport2_.get(),
asymmetric);
srtp_transport1_ = rtc::MakeUnique<SrtpTransport>(rtcp_mux_enabled);
srtp_transport2_ = rtc::MakeUnique<SrtpTransport>(rtcp_mux_enabled);
rtp_transport1->SetRtpPacketTransport(rtp_packet_transport1_.get());
rtp_transport2->SetRtpPacketTransport(rtp_packet_transport2_.get());
srtp_transport1_->SetRtpPacketTransport(rtp_packet_transport1_.get());
srtp_transport2_->SetRtpPacketTransport(rtp_packet_transport2_.get());
// Add payload type for RTP packet and RTCP packet.
rtp_transport1->AddHandledPayloadType(0x00);
rtp_transport2->AddHandledPayloadType(0x00);
rtp_transport1->AddHandledPayloadType(0xc9);
rtp_transport2->AddHandledPayloadType(0xc9);
srtp_transport1_->SignalRtcpPacketReceived.connect(
&rtp_sink1_, &TransportObserver::OnRtcpPacketReceived);
srtp_transport2_->SignalRtcpPacketReceived.connect(
&rtp_sink2_, &TransportObserver::OnRtcpPacketReceived);
srtp_transport1_ =
rtc::MakeUnique<SrtpTransport>(std::move(rtp_transport1));
srtp_transport2_ =
rtc::MakeUnique<SrtpTransport>(std::move(rtp_transport2));
RtpDemuxerCriteria demuxer_criteria;
// 0x00 is the payload type used in kPcmuFrame.
demuxer_criteria.payload_types = {0x00};
srtp_transport1_->RegisterRtpDemuxerSink(demuxer_criteria, &rtp_sink1_);
srtp_transport2_->RegisterRtpDemuxerSink(demuxer_criteria, &rtp_sink2_);
srtp_transport1_->SignalPacketReceived.connect(
this, &SrtpTransportTest::OnPacketReceived1);
srtp_transport2_->SignalPacketReceived.connect(
this, &SrtpTransportTest::OnPacketReceived2);
}
~SrtpTransportTest() {
if (srtp_transport1_) {
srtp_transport1_->UnregisterRtpDemuxerSink(&rtp_sink1_);
}
if (srtp_transport2_) {
srtp_transport2_->UnregisterRtpDemuxerSink(&rtp_sink2_);
}
void OnPacketReceived1(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
RTC_LOG(LS_INFO) << "SrtpTransport1 Received a packet.";
last_recv_packet1_ = *packet;
}
void OnPacketReceived2(bool rtcp,
rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
RTC_LOG(LS_INFO) << "SrtpTransport2 Received a packet.";
last_recv_packet2_ = *packet;
}
// With external auth enabled, SRTP doesn't write the auth tag and
@ -134,9 +142,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
if (srtp_transport1_->IsExternalAuthActive()) {
TestRtpAuthParams(srtp_transport1_.get(), cipher_suite_name);
} else {
ASSERT_TRUE(rtp_sink2_.last_recv_rtp_packet().data());
EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtp_packet().data(),
original_rtp_data, rtp_len));
ASSERT_TRUE(last_recv_packet2_.data());
EXPECT_EQ(0,
memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len));
// Get the encrypted packet from underneath packet transport and verify
// the data is actually encrypted.
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
@ -151,9 +159,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
if (srtp_transport2_->IsExternalAuthActive()) {
TestRtpAuthParams(srtp_transport2_.get(), cipher_suite_name);
} else {
ASSERT_TRUE(rtp_sink1_.last_recv_rtp_packet().data());
EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtp_packet().data(),
original_rtp_data, rtp_len));
ASSERT_TRUE(last_recv_packet1_.data());
EXPECT_EQ(0,
memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len));
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport2_->rtp_packet_transport());
EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
@ -162,12 +170,12 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
}
void TestSendRecvRtcpPacket(const std::string& cipher_suite_name) {
size_t rtcp_len = sizeof(::kRtcpReport);
size_t rtcp_len = sizeof(kRtcpReport);
size_t packet_size =
rtcp_len + 4 + rtc::rtcp_auth_tag_len(cipher_suite_name);
rtc::Buffer rtcp_packet_buffer(packet_size);
char* rtcp_packet_data = rtcp_packet_buffer.data<char>();
memcpy(rtcp_packet_data, ::kRtcpReport, rtcp_len);
memcpy(rtcp_packet_data, kRtcpReport, rtcp_len);
rtc::CopyOnWriteBuffer rtcp_packet1to2(rtcp_packet_data, rtcp_len,
packet_size);
@ -179,9 +187,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
// that the packet can be successfully received and decrypted.
ASSERT_TRUE(srtp_transport1_->SendRtcpPacket(&rtcp_packet1to2, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(rtp_sink2_.last_recv_rtcp_packet().data());
EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtcp_packet().data(),
rtcp_packet_data, rtcp_len));
ASSERT_TRUE(last_recv_packet2_.data());
EXPECT_EQ(0, memcmp(last_recv_packet2_.data(), rtcp_packet_data, rtcp_len));
// Get the encrypted packet from underneath packet transport and verify the
// data is actually encrypted.
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
@ -192,9 +199,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
// Do the same thing in the opposite direction;
ASSERT_TRUE(srtp_transport2_->SendRtcpPacket(&rtcp_packet2to1, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(rtp_sink1_.last_recv_rtcp_packet().data());
EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtcp_packet().data(),
rtcp_packet_data, rtcp_len));
ASSERT_TRUE(last_recv_packet1_.data());
EXPECT_EQ(0, memcmp(last_recv_packet1_.data(), rtcp_packet_data, rtcp_len));
fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport2_->rtp_packet_transport());
EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
@ -261,9 +267,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
// that the packet can be successfully received and decrypted.
ASSERT_TRUE(srtp_transport1_->SendRtpPacket(&rtp_packet1to2, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(rtp_sink2_.last_recv_rtp_packet().data());
EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtp_packet().data(),
original_rtp_data, rtp_len));
ASSERT_TRUE(last_recv_packet2_.data());
EXPECT_EQ(0, memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len));
// Get the encrypted packet from underneath packet transport and verify the
// data and header extension are actually encrypted.
auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
@ -279,9 +284,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
// Do the same thing in the opposite direction;
ASSERT_TRUE(srtp_transport2_->SendRtpPacket(&rtp_packet2to1, options,
cricket::PF_SRTP_BYPASS));
ASSERT_TRUE(rtp_sink1_.last_recv_rtp_packet().data());
EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtp_packet().data(),
original_rtp_data, rtp_len));
ASSERT_TRUE(last_recv_packet1_.data());
EXPECT_EQ(0, memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len));
fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
srtp_transport2_->rtp_packet_transport());
EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
@ -324,9 +328,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
std::unique_ptr<rtc::FakePacketTransport> rtp_packet_transport1_;
std::unique_ptr<rtc::FakePacketTransport> rtp_packet_transport2_;
TransportObserver rtp_sink1_;
TransportObserver rtp_sink2_;
rtc::CopyOnWriteBuffer last_recv_packet1_;
rtc::CopyOnWriteBuffer last_recv_packet2_;
int sequence_number_ = 0;
};

10
pc/transportcontroller.cc
View File

@ -458,12 +458,16 @@ webrtc::DtlsSrtpTransport* TransportController::CreateDtlsSrtpTransport(
return existing_rtp_transport->dtls_srtp_transport;
}
auto new_dtls_srtp_transport =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(rtcp_mux_enabled);
auto new_srtp_transport =
rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_enabled);
#if defined(ENABLE_EXTERNAL_AUTH)
new_dtls_srtp_transport->EnableExternalAuth();
new_srtp_transport->EnableExternalAuth();
#endif
auto new_dtls_srtp_transport =
rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(new_srtp_transport));
auto rtp_dtls_transport = CreateDtlsTransport_n(
transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
auto rtcp_dtls_transport =