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Reland "sdp: parse and serialize b=TIAS"

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This reverts commit 20b701f3d79c499b0981f03fbf3a9b0fe531ac5d.

Reason for reland: Reverting did not affect the test regression.

Original change's description:
> Revert "sdp: parse and serialize b=TIAS"
>
> This reverts commit c6801d4522ab94f965e258e68259fde312023654.
>
> Reason for revert: Speculatively reverting since it possibly breaks downstream performance test.
>
> One issue I noticed is that the correct SDP won't be produced if set_bandwidth_type hasn't been called. Probably should default to b=AS in that case.
>
> Original change's description:
> > sdp: parse and serialize b=TIAS
> >
> > BUG=webrtc:5788
> >
> > Change-Id: I063c756004e4c224fffa36d2800603c7b7e50dce
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179223
> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31729}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org,minyue@webrtc.org,philipp.hancke@googlemail.com,jleconte@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:5788
> Change-Id: I2a3f676b4359834e511dffd5adedc9388e0ea0f8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179620
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31762}

TBR=nisse@webrtc.org

Bug: webrtc:5788
Change-Id: I5c0ef29d275bb2264d9b706b085f7933d59e2801
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179760
Commit-Queue: Taylor <deadbeef@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31816}
This commit is contained in:
Taylor Brandstetter
2020-07-27 15:52:02 -07:00
committed by Commit Bot
parent 003c9be817
commit ee8c246be7
5 changed files with 115 additions and 46 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
media/base/media_constants.cc
View File

@ -119,4 +119,8 @@ const int kDefaultVideoMaxFramerate = 60;
const size_t kConferenceMaxNumSpatialLayers = 3;
const size_t kConferenceMaxNumTemporalLayers = 3;
const size_t kConferenceDefaultNumTemporalLayers = 3;
// RFC 3556 and RFC 3890
const char kApplicationSpecificBandwidth[] = "AS";
const char kTransportSpecificBandwidth[] = "TIAS";
} // namespace cricket

3
media/base/media_constants.h
View File

@ -145,6 +145,9 @@ extern const int kDefaultVideoMaxFramerate;
extern const size_t kConferenceMaxNumSpatialLayers;
extern const size_t kConferenceMaxNumTemporalLayers;
extern const size_t kConferenceDefaultNumTemporalLayers;
extern const char kApplicationSpecificBandwidth[];
extern const char kTransportSpecificBandwidth[];
} // namespace cricket
#endif // MEDIA_BASE_MEDIA_CONSTANTS_H_

6
pc/session_description.h
View File

@ -26,6 +26,7 @@
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_interface.h"
#include "media/base/media_channel.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
@ -126,6 +127,10 @@ class MediaContentDescription {
virtual int bandwidth() const { return bandwidth_; }
virtual void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
virtual std::string bandwidth_type() const { return bandwidth_type_; }
virtual void set_bandwidth_type(std::string bandwidth_type) {
bandwidth_type_ = bandwidth_type;
}
virtual const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
virtual void AddCrypto(const CryptoParams& params) {
@ -251,6 +256,7 @@ class MediaContentDescription {
bool rtcp_reduced_size_ = false;
bool remote_estimate_ = false;
int bandwidth_ = kAutoBandwidth;
std::string bandwidth_type_ = kApplicationSpecificBandwidth;
std::string protocol_;
std::vector<CryptoParams> cryptos_;
std::vector<webrtc::RtpExtension> rtp_header_extensions_;

109
pc/webrtc_sdp.cc
View File

@ -55,9 +55,11 @@ using cricket::ContentInfo;
using cricket::CryptoParams;
using cricket::ICE_CANDIDATE_COMPONENT_RTCP;
using cricket::ICE_CANDIDATE_COMPONENT_RTP;
using cricket::kApplicationSpecificBandwidth;
using cricket::kCodecParamMaxPTime;
using cricket::kCodecParamMinPTime;
using cricket::kCodecParamPTime;
using cricket::kTransportSpecificBandwidth;
using cricket::MediaContentDescription;
using cricket::MediaProtocolType;
using cricket::MediaType;
@ -224,8 +226,6 @@ static const char kMediaPortRejected[] = "0";
// Use IPV4 per default.
static const char kDummyAddress[] = "0.0.0.0";
static const char kDummyPort[] = "9";
// RFC 3556
static const char kApplicationSpecificMaximum[] = "AS";
static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
@ -1436,10 +1436,18 @@ void BuildMediaDescription(const ContentInfo* content_info,
AddLine(os.str(), message);
// RFC 4566
// b=AS:<bandwidth>
if (media_desc->bandwidth() >= 1000) {
InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
// b=AS:<bandwidth> or
// b=TIAS:<bandwidth>
int bandwidth = media_desc->bandwidth();
std::string bandwidth_type = media_desc->bandwidth_type();
if (bandwidth_type == kApplicationSpecificBandwidth && bandwidth >= 1000) {
InitLine(kLineTypeSessionBandwidth, bandwidth_type, &os);
bandwidth /= 1000;
os << kSdpDelimiterColon << bandwidth;
AddLine(os.str(), message);
} else if (bandwidth_type == kTransportSpecificBandwidth && bandwidth > 0) {
InitLine(kLineTypeSessionBandwidth, bandwidth_type, &os);
os << kSdpDelimiterColon << bandwidth;
AddLine(os.str(), message);
}
@ -2983,46 +2991,61 @@ bool ParseContent(const std::string& message,
// b=* (zero or more bandwidth information lines)
if (IsLineType(line, kLineTypeSessionBandwidth)) {
std::string bandwidth;
if (HasAttribute(line, kApplicationSpecificMaximum)) {
if (!GetValue(line, kApplicationSpecificMaximum, &bandwidth, error)) {
std::string bandwidth_type;
if (HasAttribute(line, kApplicationSpecificBandwidth)) {
if (!GetValue(line, kApplicationSpecificBandwidth, &bandwidth, error)) {
return false;
} else {
int b = 0;
if (!GetValueFromString(line, bandwidth, &b, error)) {
return false;
}
// TODO(deadbeef): Historically, applications may be setting a value
// of -1 to mean "unset any previously set bandwidth limit", even
// though ommitting the "b=AS" entirely will do just that. Once we've
// transitioned applications to doing the right thing, it would be
// better to treat this as a hard error instead of just ignoring it.
if (b == -1) {
RTC_LOG(LS_WARNING)
<< "Ignoring \"b=AS:-1\"; will be treated as \"no "
"bandwidth limit\".";
continue;
}
if (b < 0) {
return ParseFailed(line, "b=AS value can't be negative.", error);
}
// We should never use more than the default bandwidth for RTP-based
// data channels. Don't allow SDP to set the bandwidth, because
// that would give JS the opportunity to "break the Internet".
// See: https://code.google.com/p/chromium/issues/detail?id=280726
if (media_type == cricket::MEDIA_TYPE_DATA &&
cricket::IsRtpProtocol(protocol) &&
b > cricket::kDataMaxBandwidth / 1000) {
rtc::StringBuilder description;
description << "RTP-based data channels may not send more than "
<< cricket::kDataMaxBandwidth / 1000 << "kbps.";
return ParseFailed(line, description.str(), error);
}
// Prevent integer overflow.
b = std::min(b, INT_MAX / 1000);
media_desc->set_bandwidth(b * 1000);
}
bandwidth_type = kApplicationSpecificBandwidth;
} else if (HasAttribute(line, kTransportSpecificBandwidth)) {
if (!GetValue(line, kTransportSpecificBandwidth, &bandwidth, error)) {
return false;
}
bandwidth_type = kTransportSpecificBandwidth;
} else {
continue;
}
continue;
int b = 0;
if (!GetValueFromString(line, bandwidth, &b, error)) {
return false;
}
// TODO(deadbeef): Historically, applications may be setting a value
// of -1 to mean "unset any previously set bandwidth limit", even
// though ommitting the "b=AS" entirely will do just that. Once we've
// transitioned applications to doing the right thing, it would be
// better to treat this as a hard error instead of just ignoring it.
if (bandwidth_type == kApplicationSpecificBandwidth && b == -1) {
RTC_LOG(LS_WARNING) << "Ignoring \"b=AS:-1\"; will be treated as \"no "
"bandwidth limit\".";
continue;
}
if (b < 0) {
return ParseFailed(
line, "b=" + bandwidth_type + " value can't be negative.", error);
}
// We should never use more than the default bandwidth for RTP-based
// data channels. Don't allow SDP to set the bandwidth, because
// that would give JS the opportunity to "break the Internet".
// See: https://code.google.com/p/chromium/issues/detail?id=280726
// Disallow TIAS since it shouldn't be generated for RTP data channels in
// the first place and provides another way to get around the limitation.
if (media_type == cricket::MEDIA_TYPE_DATA &&
cricket::IsRtpProtocol(protocol) &&
(b > cricket::kDataMaxBandwidth / 1000 ||
bandwidth_type == kTransportSpecificBandwidth)) {
rtc::StringBuilder description;
description << "RTP-based data channels may not send more than "
<< cricket::kDataMaxBandwidth / 1000 << "kbps.";
return ParseFailed(line, description.str(), error);
}
// Convert values. Prevent integer overflow.
if (bandwidth_type == kApplicationSpecificBandwidth) {
b = std::min(b, INT_MAX / 1000) * 1000;
} else {
b = std::min(b, INT_MAX);
}
media_desc->set_bandwidth(b);
media_desc->set_bandwidth_type(bandwidth_type);
}
// Parse the media level connection data.

39
pc/webrtc_sdp_unittest.cc
View File

@ -2189,16 +2189,31 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBundle) {
TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) {
VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
vcd->set_bandwidth(100 * 1000);
vcd->set_bandwidth(100 * 1000 + 755); // Integer division will drop the 755.
vcd->set_bandwidth_type("AS");
AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
acd->set_bandwidth(50 * 1000);
acd->set_bandwidth(555);
acd->set_bandwidth_type("TIAS");
ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
jdesc_.session_version()));
std::string message = webrtc::SdpSerialize(jdesc_);
std::string sdp_with_bandwidth = kSdpFullString;
InjectAfter("c=IN IP4 74.125.224.39\r\n", "b=AS:100\r\n",
&sdp_with_bandwidth);
InjectAfter("c=IN IP4 74.125.127.126\r\n", "b=AS:50\r\n",
InjectAfter("c=IN IP4 74.125.127.126\r\n", "b=TIAS:555\r\n",
&sdp_with_bandwidth);
EXPECT_EQ(sdp_with_bandwidth, message);
}
// Should default to b=AS if bandwidth_type isn't set.
TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithMissingBandwidthType) {
VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
vcd->set_bandwidth(100 * 1000);
ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
jdesc_.session_version()));
std::string message = webrtc::SdpSerialize(jdesc_);
std::string sdp_with_bandwidth = kSdpFullString;
InjectAfter("c=IN IP4 74.125.224.39\r\n", "b=AS:100\r\n",
&sdp_with_bandwidth);
EXPECT_EQ(sdp_with_bandwidth, message);
}
@ -2309,6 +2324,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
JsepSessionDescription jsep_desc(kDummyType);
AddRtpDataChannel();
data_desc_->set_bandwidth(100 * 1000);
data_desc_->set_bandwidth_type("AS");
MakeDescriptionWithoutCandidates(&jsep_desc);
std::string message = webrtc::SdpSerialize(jsep_desc);
@ -2612,6 +2628,23 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBandwidth) {
EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
}
TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithTiasBandwidth) {
JsepSessionDescription jdesc_with_bandwidth(kDummyType);
std::string sdp_with_bandwidth = kSdpFullString;
InjectAfter("a=mid:video_content_name\r\na=sendrecv\r\n", "b=TIAS:100000\r\n",
&sdp_with_bandwidth);
InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n", "b=TIAS:50000\r\n",
&sdp_with_bandwidth);
EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
vcd->set_bandwidth(100 * 1000);
AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
acd->set_bandwidth(50 * 1000);
ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
jdesc_.session_version()));
EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
}
TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithIceOptions) {
JsepSessionDescription jdesc_with_ice_options(kDummyType);
std::string sdp_with_ice_options = kSdpFullString;