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Revert "sdp: parse and serialize b=TIAS"
This reverts commit c6801d4522
.
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}
This commit is contained in:
parent
de90862cce
commit
20b701f3d7
3 changed files with 43 additions and 88 deletions
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@ -126,10 +126,6 @@ class MediaContentDescription {
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virtual int bandwidth() const { return bandwidth_; }
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virtual void set_bandwidth(int bandwidth) { bandwidth_ = bandwidth; }
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virtual std::string bandwidth_type() const { return bandwidth_type_; }
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virtual void set_bandwidth_type(std::string bandwidth_type) {
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bandwidth_type_ = bandwidth_type;
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}
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virtual const std::vector<CryptoParams>& cryptos() const { return cryptos_; }
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virtual void AddCrypto(const CryptoParams& params) {
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@ -255,7 +251,6 @@ class MediaContentDescription {
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bool rtcp_reduced_size_ = false;
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bool remote_estimate_ = false;
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int bandwidth_ = kAutoBandwidth;
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std::string bandwidth_type_;
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std::string protocol_;
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std::vector<CryptoParams> cryptos_;
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std::vector<webrtc::RtpExtension> rtp_header_extensions_;
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@ -225,8 +225,7 @@ static const char kMediaPortRejected[] = "0";
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static const char kDummyAddress[] = "0.0.0.0";
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static const char kDummyPort[] = "9";
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// RFC 3556
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static const char kApplicationSpecificBandwidth[] = "AS";
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static const char kTransportSpecificBandwidth[] = "TIAS";
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static const char kApplicationSpecificMaximum[] = "AS";
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static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
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@ -1438,14 +1437,9 @@ void BuildMediaDescription(const ContentInfo* content_info,
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// RFC 4566
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// b=AS:<bandwidth>
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int bandwidth = media_desc->bandwidth();
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if (bandwidth >= 1000) {
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std::string bandwidth_type = media_desc->bandwidth_type();
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InitLine(kLineTypeSessionBandwidth, bandwidth_type, &os);
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if (bandwidth_type == kApplicationSpecificBandwidth) {
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bandwidth /= 1000;
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}
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os << kSdpDelimiterColon << bandwidth;
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if (media_desc->bandwidth() >= 1000) {
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InitLine(kLineTypeSessionBandwidth, kApplicationSpecificMaximum, &os);
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os << kSdpDelimiterColon << (media_desc->bandwidth() / 1000);
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AddLine(os.str(), message);
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}
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@ -2989,20 +2983,10 @@ bool ParseContent(const std::string& message,
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// b=* (zero or more bandwidth information lines)
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if (IsLineType(line, kLineTypeSessionBandwidth)) {
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std::string bandwidth;
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std::string bandwidth_type;
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if (HasAttribute(line, kApplicationSpecificBandwidth)) {
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if (!GetValue(line, kApplicationSpecificBandwidth, &bandwidth, error)) {
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if (HasAttribute(line, kApplicationSpecificMaximum)) {
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if (!GetValue(line, kApplicationSpecificMaximum, &bandwidth, error)) {
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return false;
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}
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bandwidth_type = kApplicationSpecificBandwidth;
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} else if (HasAttribute(line, kTransportSpecificBandwidth)) {
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if (!GetValue(line, kTransportSpecificBandwidth, &bandwidth, error)) {
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return false;
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}
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bandwidth_type = kTransportSpecificBandwidth;
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} else {
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continue;
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}
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int b = 0;
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if (!GetValueFromString(line, bandwidth, &b, error)) {
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return false;
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@ -3012,37 +2996,33 @@ bool ParseContent(const std::string& message,
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// though ommitting the "b=AS" entirely will do just that. Once we've
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// transitioned applications to doing the right thing, it would be
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// better to treat this as a hard error instead of just ignoring it.
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if (bandwidth_type == kApplicationSpecificBandwidth && b == -1) {
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RTC_LOG(LS_WARNING) << "Ignoring \"b=AS:-1\"; will be treated as \"no "
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if (b == -1) {
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RTC_LOG(LS_WARNING)
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<< "Ignoring \"b=AS:-1\"; will be treated as \"no "
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"bandwidth limit\".";
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continue;
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}
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if (b < 0) {
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return ParseFailed(
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line, "b=" + bandwidth_type + " value can't be negative.", error);
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return ParseFailed(line, "b=AS value can't be negative.", error);
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}
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// We should never use more than the default bandwidth for RTP-based
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// data channels. Don't allow SDP to set the bandwidth, because
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// that would give JS the opportunity to "break the Internet".
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// See: https://code.google.com/p/chromium/issues/detail?id=280726
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// Disallow TIAS since that is not generated for this.
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if (media_type == cricket::MEDIA_TYPE_DATA &&
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cricket::IsRtpProtocol(protocol) &&
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(b > cricket::kDataMaxBandwidth / 1000 ||
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bandwidth_type == kTransportSpecificBandwidth)) {
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b > cricket::kDataMaxBandwidth / 1000) {
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rtc::StringBuilder description;
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description << "RTP-based data channels may not send more than "
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<< cricket::kDataMaxBandwidth / 1000 << "kbps.";
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return ParseFailed(line, description.str(), error);
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}
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// Convert values. Prevent integer overflow.
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if (bandwidth_type == kApplicationSpecificBandwidth) {
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b = std::min(b, INT_MAX / 1000) * 1000;
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} else {
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b = std::min(b, INT_MAX);
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// Prevent integer overflow.
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b = std::min(b, INT_MAX / 1000);
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media_desc->set_bandwidth(b * 1000);
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}
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media_desc->set_bandwidth(b);
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media_desc->set_bandwidth_type(bandwidth_type);
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}
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continue;
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}
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// Parse the media level connection data.
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@ -2189,18 +2189,16 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBundle) {
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TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) {
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VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
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vcd->set_bandwidth(100 * 1000 + 755); // Integer division will drop the 755.
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vcd->set_bandwidth_type("AS");
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vcd->set_bandwidth(100 * 1000);
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AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
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acd->set_bandwidth(50 * 1000);
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acd->set_bandwidth_type("TIAS");
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ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
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jdesc_.session_version()));
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std::string message = webrtc::SdpSerialize(jdesc_);
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std::string sdp_with_bandwidth = kSdpFullString;
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InjectAfter("c=IN IP4 74.125.224.39\r\n", "b=AS:100\r\n",
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&sdp_with_bandwidth);
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InjectAfter("c=IN IP4 74.125.127.126\r\n", "b=TIAS:50000\r\n",
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InjectAfter("c=IN IP4 74.125.127.126\r\n", "b=AS:50\r\n",
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&sdp_with_bandwidth);
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EXPECT_EQ(sdp_with_bandwidth, message);
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}
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@ -2311,7 +2309,6 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
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JsepSessionDescription jsep_desc(kDummyType);
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AddRtpDataChannel();
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data_desc_->set_bandwidth(100 * 1000);
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data_desc_->set_bandwidth_type("AS");
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MakeDescriptionWithoutCandidates(&jsep_desc);
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std::string message = webrtc::SdpSerialize(jsep_desc);
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@ -2615,23 +2612,6 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBandwidth) {
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EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
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}
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TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithTiasBandwidth) {
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JsepSessionDescription jdesc_with_bandwidth(kDummyType);
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std::string sdp_with_bandwidth = kSdpFullString;
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InjectAfter("a=mid:video_content_name\r\na=sendrecv\r\n", "b=TIAS:100000\r\n",
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&sdp_with_bandwidth);
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InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n", "b=TIAS:50000\r\n",
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&sdp_with_bandwidth);
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EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
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VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
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vcd->set_bandwidth(100 * 1000);
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AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
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acd->set_bandwidth(50 * 1000);
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ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
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jdesc_.session_version()));
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EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
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}
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TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithIceOptions) {
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JsepSessionDescription jdesc_with_ice_options(kDummyType);
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std::string sdp_with_ice_options = kSdpFullString;
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