Reland of: Adding error output param to SetConfiguration, using new RTCError type.
Most notably, will return "INVALID_MODIFICATION" if a field in the
configuration was modified and modification of that field isn't supported.
Also changing RTCError to a class that wraps an enum type, because it will
eventually need to hold other information (like SDP line number), to match
the RTCError that was recently added to the spec:
https://github.com/w3c/webrtc-pc/pull/850
BUG=webrtc:6916
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Original-Commit-Position: refs/heads/master@{#15777}
Committed: https://chromium.googlesource.com/external/webrtc/+/7a5fa6cd6173adbe32aedc1aedc872478121f5ed
Review-Url: https://codereview.webrtc.org/2587133004
Cr-Commit-Position: refs/heads/master@{#16016}
diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc
index 1d85e73..2f21e0c 100644
--- a/webrtc/api/peerconnection.cc
+++ b/webrtc/api/peerconnection.cc
@@ -49,6 +49,8 @@
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
+using webrtc::RTCError;
+using webrtc::RTCErrorType;
using webrtc::RtpSenderInternal;
using webrtc::RtpSenderInterface;
using webrtc::RtpSenderProxy;
@@ -208,10 +210,11 @@
// Adds a STUN or TURN server to the appropriate list,
// by parsing |url| and using the username/password in |server|.
-bool ParseIceServerUrl(const PeerConnectionInterface::IceServer& server,
- const std::string& url,
- cricket::ServerAddresses* stun_servers,
- std::vector<cricket::RelayServerConfig>* turn_servers) {
+RTCErrorType ParseIceServerUrl(
+ const PeerConnectionInterface::IceServer& server,
+ const std::string& url,
+ cricket::ServerAddresses* stun_servers,
+ std::vector<cricket::RelayServerConfig>* turn_servers) {
// draft-nandakumar-rtcweb-stun-uri-01
// stunURI = scheme ":" stun-host [ ":" stun-port ]
// scheme = "stun" / "stuns"
@@ -239,14 +242,14 @@
rtc::tokenize_with_empty_tokens(uri_transport_param, '=', &tokens);
if (tokens[0] != kTransport) {
LOG(LS_WARNING) << "Invalid transport parameter key.";
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
if (tokens.size() < 2 ||
!cricket::StringToProto(tokens[1].c_str(), &turn_transport_type) ||
(turn_transport_type != cricket::PROTO_UDP &&
turn_transport_type != cricket::PROTO_TCP)) {
LOG(LS_WARNING) << "Transport param should always be udp or tcp.";
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
}
@@ -256,7 +259,7 @@
&service_type,
&hoststring)) {
LOG(LS_WARNING) << "Invalid transport parameter in ICE URI: " << url;
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
// GetServiceTypeAndHostnameFromUri should never give an empty hoststring
@@ -269,12 +272,12 @@
std::string username(server.username);
if (tokens.size() > kTurnHostTokensNum) {
LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
if (tokens.size() == kTurnHostTokensNum) {
if (tokens[0].empty() || tokens[1].empty()) {
LOG(LS_WARNING) << "Invalid user@hostname format: " << hoststring;
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
username.assign(rtc::s_url_decode(tokens[0]));
hoststring = tokens[1];
@@ -291,12 +294,12 @@
std::string address;
if (!ParseHostnameAndPortFromString(hoststring, &address, &port)) {
LOG(WARNING) << "Invalid hostname format: " << uri_without_transport;
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
if (port <= 0 || port > 0xffff) {
LOG(WARNING) << "Invalid port: " << port;
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
switch (service_type) {
@@ -306,6 +309,11 @@
break;
case TURN:
case TURNS: {
+ if (username.empty() || server.password.empty()) {
+ // The WebRTC spec requires throwing an InvalidAccessError when username
+ // or credential are ommitted; this is the native equivalent.
+ return RTCErrorType::INVALID_PARAMETER;
+ }
cricket::RelayServerConfig config = cricket::RelayServerConfig(
address, port, username, server.password, turn_transport_type);
if (server.tls_cert_policy ==
@@ -316,12 +324,13 @@
turn_servers->push_back(config);
break;
}
- case INVALID:
default:
- LOG(WARNING) << "Configuration not supported: " << url;
- return false;
+ // We shouldn't get to this point with an invalid service_type, we should
+ // have returned an error already.
+ RTC_DCHECK(false) << "Unexpected service type";
+ return RTCErrorType::INTERNAL_ERROR;
}
- return true;
+ return RTCErrorType::NONE;
}
// Check if we can send |new_stream| on a PeerConnection.
@@ -433,11 +442,19 @@
}
}
+// Helper to set an error and return from a method.
+bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
+ if (error) {
+ error->set_type(type);
+ }
+ return type == webrtc::RTCErrorType::NONE;
+}
+
} // namespace
namespace webrtc {
-static const char* const kRtcErrorNames[] = {
+static const char* const kRTCErrorTypeNames[] = {
"NONE",
"UNSUPPORTED_PARAMETER",
"INVALID_PARAMETER",
@@ -448,12 +465,82 @@
"NETWORK_ERROR",
"INTERNAL_ERROR",
};
+static_assert(static_cast<int>(RTCErrorType::INTERNAL_ERROR) ==
+ (arraysize(kRTCErrorTypeNames) - 1),
+ "kRTCErrorTypeNames must have as many strings as RTCErrorType "
+ "has values.");
-std::ostream& operator<<(std::ostream& stream, RtcError error) {
+std::ostream& operator<<(std::ostream& stream, RTCErrorType error) {
int index = static_cast<int>(error);
- RTC_CHECK(index < static_cast<int>(sizeof(kRtcErrorNames) /
- sizeof(kRtcErrorNames[0])));
- return stream << kRtcErrorNames[index];
+ return stream << kRTCErrorTypeNames[index];
+}
+
+bool PeerConnectionInterface::RTCConfiguration::operator==(
+ const PeerConnectionInterface::RTCConfiguration& o) const {
+ // This static_assert prevents us from accidentally breaking operator==.
+ struct stuff_being_tested_for_equality {
+ IceTransportsType type;
+ IceServers servers;
+ BundlePolicy bundle_policy;
+ RtcpMuxPolicy rtcp_mux_policy;
+ TcpCandidatePolicy tcp_candidate_policy;
+ CandidateNetworkPolicy candidate_network_policy;
+ int audio_jitter_buffer_max_packets;
+ bool audio_jitter_buffer_fast_accelerate;
+ int ice_connection_receiving_timeout;
+ int ice_backup_candidate_pair_ping_interval;
+ ContinualGatheringPolicy continual_gathering_policy;
+ std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
+ bool prioritize_most_likely_ice_candidate_pairs;
+ struct cricket::MediaConfig media_config;
+ bool disable_ipv6;
+ bool enable_rtp_data_channel;
+ bool enable_quic;
+ rtc::Optional<int> screencast_min_bitrate;
+ rtc::Optional<bool> combined_audio_video_bwe;
+ rtc::Optional<bool> enable_dtls_srtp;
+ int ice_candidate_pool_size;
+ bool prune_turn_ports;
+ bool presume_writable_when_fully_relayed;
+ bool enable_ice_renomination;
+ bool redetermine_role_on_ice_restart;
+ };
+ static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
+ "Did you add something to RTCConfiguration and forget to "
+ "update operator==?");
+ return type == o.type && servers == o.servers &&
+ bundle_policy == o.bundle_policy &&
+ rtcp_mux_policy == o.rtcp_mux_policy &&
+ tcp_candidate_policy == o.tcp_candidate_policy &&
+ candidate_network_policy == o.candidate_network_policy &&
+ audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
+ audio_jitter_buffer_fast_accelerate ==
+ o.audio_jitter_buffer_fast_accelerate &&
+ ice_connection_receiving_timeout ==
+ o.ice_connection_receiving_timeout &&
+ ice_backup_candidate_pair_ping_interval ==
+ o.ice_backup_candidate_pair_ping_interval &&
+ continual_gathering_policy == o.continual_gathering_policy &&
+ certificates == o.certificates &&
+ prioritize_most_likely_ice_candidate_pairs ==
+ o.prioritize_most_likely_ice_candidate_pairs &&
+ media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
+ enable_rtp_data_channel == o.enable_rtp_data_channel &&
+ enable_quic == o.enable_quic &&
+ screencast_min_bitrate == o.screencast_min_bitrate &&
+ combined_audio_video_bwe == o.combined_audio_video_bwe &&
+ enable_dtls_srtp == o.enable_dtls_srtp &&
+ ice_candidate_pool_size == o.ice_candidate_pool_size &&
+ prune_turn_ports == o.prune_turn_ports &&
+ presume_writable_when_fully_relayed ==
+ o.presume_writable_when_fully_relayed &&
+ enable_ice_renomination == o.enable_ice_renomination &&
+ redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart;
+}
+
+bool PeerConnectionInterface::RTCConfiguration::operator!=(
+ const PeerConnectionInterface::RTCConfiguration& o) const {
+ return !(*this == o);
}
// Generate a RTCP CNAME when a PeerConnection is created.
@@ -555,28 +642,33 @@
return mandatory_constraints_satisfied == constraints->GetMandatory().size();
}
-bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
- cricket::ServerAddresses* stun_servers,
- std::vector<cricket::RelayServerConfig>* turn_servers) {
+RTCErrorType ParseIceServers(
+ const PeerConnectionInterface::IceServers& servers,
+ cricket::ServerAddresses* stun_servers,
+ std::vector<cricket::RelayServerConfig>* turn_servers) {
for (const webrtc::PeerConnectionInterface::IceServer& server : servers) {
if (!server.urls.empty()) {
for (const std::string& url : server.urls) {
if (url.empty()) {
LOG(LS_ERROR) << "Empty uri.";
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
- if (!ParseIceServerUrl(server, url, stun_servers, turn_servers)) {
- return false;
+ RTCErrorType err =
+ ParseIceServerUrl(server, url, stun_servers, turn_servers);
+ if (err != RTCErrorType::NONE) {
+ return err;
}
}
} else if (!server.uri.empty()) {
// Fallback to old .uri if new .urls isn't present.
- if (!ParseIceServerUrl(server, server.uri, stun_servers, turn_servers)) {
- return false;
+ RTCErrorType err =
+ ParseIceServerUrl(server, server.uri, stun_servers, turn_servers);
+ if (err != RTCErrorType::NONE) {
+ return err;
}
} else {
LOG(LS_ERROR) << "Empty uri.";
- return false;
+ return RTCErrorType::SYNTAX_ERROR;
}
}
// Candidates must have unique priorities, so that connectivity checks
@@ -586,7 +678,7 @@
// First in the list gets highest priority.
turn_server.priority = priority--;
}
- return true;
+ return RTCErrorType::NONE;
}
PeerConnection::PeerConnection(PeerConnectionFactory* factory)
@@ -633,12 +725,18 @@
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) {
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
- RTC_DCHECK(observer != nullptr);
+ if (!allocator) {
+ LOG(LS_ERROR) << "PeerConnection initialized without a PortAllocator? "
+ << "This shouldn't happen if using PeerConnectionFactory.";
+ return false;
+ }
if (!observer) {
+ // TODO(deadbeef): Why do we do this?
+ LOG(LS_ERROR) << "PeerConnection initialized without a "
+ << "PeerConnectionObserver";
return false;
}
observer_ = observer;
-
port_allocator_ = std::move(allocator);
// The port allocator lives on the network thread and should be initialized
@@ -1301,7 +1399,8 @@
return configuration_;
}
-bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration) {
+bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
+ RTCError* error) {
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
if (session_->local_description() &&
@@ -1309,32 +1408,61 @@
configuration_.ice_candidate_pool_size) {
LOG(LS_ERROR) << "Can't change candidate pool size after calling "
"SetLocalDescription.";
- return false;
- }
- // TODO(deadbeef): Return false and log an error if there are any unsupported
- // modifications.
- if (port_allocator_) {
- if (!network_thread()->Invoke<bool>(
- RTC_FROM_HERE,
- rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
- configuration))) {
- LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
- return false;
- }
+ return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
}
- // TODO(deadbeef): Shouldn't have to hop to the network thread twice...
- session_->SetIceConfig(session_->ParseIceConfig(configuration));
+ // The simplest (and most future-compatible) way to tell if the config was
+ // modified in an invalid way is to copy each property we do support
+ // modifying, then use operator==. There are far more properties we don't
+ // support modifying than those we do, and more could be added.
+ RTCConfiguration modified_config = configuration_;
+ modified_config.servers = configuration.servers;
+ modified_config.type = configuration.type;
+ modified_config.ice_candidate_pool_size =
+ configuration.ice_candidate_pool_size;
+ modified_config.prune_turn_ports = configuration.prune_turn_ports;
+ if (configuration != modified_config) {
+ LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
+ return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
+ }
+
+ // Note that this isn't possible through chromium, since it's an unsigned
+ // short in WebIDL.
+ if (configuration.ice_candidate_pool_size < 0 ||
+ configuration.ice_candidate_pool_size > UINT16_MAX) {
+ return SafeSetError(RTCErrorType::INVALID_RANGE, error);
+ }
+
+ // Parse ICE servers before hopping to network thread.
+ cricket::ServerAddresses stun_servers;
+ std::vector<cricket::RelayServerConfig> turn_servers;
+ RTCErrorType parse_error =
+ ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
+ if (parse_error != RTCErrorType::NONE) {
+ return SafeSetError(parse_error, error);
+ }
+
+ // In theory this shouldn't fail.
+ if (!network_thread()->Invoke<bool>(
+ RTC_FROM_HERE,
+ rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
+ stun_servers, turn_servers, modified_config.type,
+ modified_config.ice_candidate_pool_size,
+ modified_config.prune_turn_ports))) {
+ LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
+ return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
+ }
// As described in JSEP, calling setConfiguration with new ICE servers or
// candidate policy must set a "needs-ice-restart" bit so that the next offer
// triggers an ICE restart which will pick up the changes.
- if (configuration.servers != configuration_.servers ||
- configuration.type != configuration_.type) {
+ if (modified_config.servers != configuration_.servers ||
+ modified_config.type != configuration_.type ||
+ modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
session_->SetNeedsIceRestartFlag();
}
- configuration_ = configuration;
- return true;
+ configuration_ = modified_config;
+ return SafeSetError(RTCErrorType::NONE, error);
}
bool PeerConnection::AddIceCandidate(
@@ -1361,7 +1489,7 @@
}
// Send information about IPv4/IPv6 status.
- if (uma_observer_ && port_allocator_) {
+ if (uma_observer_) {
port_allocator_->SetMetricsObserver(uma_observer_);
if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
uma_observer_->IncrementEnumCounter(
@@ -2374,7 +2502,8 @@
const RTCConfiguration& configuration) {
cricket::ServerAddresses stun_servers;
std::vector<cricket::RelayServerConfig> turn_servers;
- if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
+ if (ParseIceServers(configuration.servers, &stun_servers, &turn_servers) !=
+ RTCErrorType::NONE) {
return false;
}
@@ -2420,19 +2549,17 @@
}
bool PeerConnection::ReconfigurePortAllocator_n(
- const RTCConfiguration& configuration) {
- cricket::ServerAddresses stun_servers;
- std::vector<cricket::RelayServerConfig> turn_servers;
- if (!ParseIceServers(configuration.servers, &stun_servers, &turn_servers)) {
- return false;
- }
+ const cricket::ServerAddresses& stun_servers,
+ const std::vector<cricket::RelayServerConfig>& turn_servers,
+ IceTransportsType type,
+ int candidate_pool_size,
+ bool prune_turn_ports) {
port_allocator_->set_candidate_filter(
- ConvertIceTransportTypeToCandidateFilter(configuration.type));
+ ConvertIceTransportTypeToCandidateFilter(type));
// Call this last since it may create pooled allocator sessions using the
// candidate filter set above.
return port_allocator_->SetConfiguration(
- stun_servers, turn_servers, configuration.ice_candidate_pool_size,
- configuration.prune_turn_ports);
+ stun_servers, turn_servers, candidate_pool_size, prune_turn_ports);
}
bool PeerConnection::StartRtcEventLog_w(rtc::PlatformFile file,
diff --git a/webrtc/api/peerconnection.h b/webrtc/api/peerconnection.h
index 5269e3a..82dc8ed 100644
--- a/webrtc/api/peerconnection.h
+++ b/webrtc/api/peerconnection.h
@@ -54,10 +54,12 @@
cricket::MediaSessionOptions* session_options);
// Parses the URLs for each server in |servers| to build |stun_servers| and
-// |turn_servers|.
-bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
- cricket::ServerAddresses* stun_servers,
- std::vector<cricket::RelayServerConfig>* turn_servers);
+// |turn_servers|. Can return SYNTAX_ERROR if the URL is malformed, or
+// INVALID_PARAMETER if a TURN server is missing |username| or |password|.
+RTCErrorType ParseIceServers(
+ const PeerConnectionInterface::IceServers& servers,
+ cricket::ServerAddresses* stun_servers,
+ std::vector<cricket::RelayServerConfig>* turn_servers);
// PeerConnection implements the PeerConnectionInterface interface.
// It uses WebRtcSession to implement the PeerConnection functionality.
@@ -137,7 +139,12 @@
SessionDescriptionInterface* desc) override;
PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
bool SetConfiguration(
- const PeerConnectionInterface::RTCConfiguration& configuration) override;
+ const PeerConnectionInterface::RTCConfiguration& configuration,
+ RTCError* error) override;
+ bool SetConfiguration(
+ const PeerConnectionInterface::RTCConfiguration& configuration) override {
+ return SetConfiguration(configuration, nullptr);
+ }
bool AddIceCandidate(const IceCandidateInterface* candidate) override;
bool RemoveIceCandidates(
const std::vector<cricket::Candidate>& candidates) override;
@@ -373,9 +380,14 @@
// Called when first configuring the port allocator.
bool InitializePortAllocator_n(const RTCConfiguration& configuration);
- // Called when SetConfiguration is called. Only a subset of the configuration
- // is applied.
- bool ReconfigurePortAllocator_n(const RTCConfiguration& configuration);
+ // Called when SetConfiguration is called to apply the supported subset
+ // of the configuration on the network thread.
+ bool ReconfigurePortAllocator_n(
+ const cricket::ServerAddresses& stun_servers,
+ const std::vector<cricket::RelayServerConfig>& turn_servers,
+ IceTransportsType type,
+ int candidate_pool_size,
+ bool prune_turn_ports);
// Starts recording an Rtc EventLog using the supplied platform file.
// This function should only be called from the worker thread.
diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc
index 4fe29f2..c27e53f 100644
--- a/webrtc/api/peerconnection_unittest.cc
+++ b/webrtc/api/peerconnection_unittest.cc
@@ -2621,6 +2621,10 @@
return ParseUrl(url, std::string(), std::string());
}
+ bool ParseTurnUrl(const std::string& url) {
+ return ParseUrl(url, "username", "password");
+ }
+
bool ParseUrl(const std::string& url,
const std::string& username,
const std::string& password) {
@@ -2640,7 +2644,8 @@
server.password = password;
server.tls_cert_policy = tls_certificate_policy;
servers.push_back(server);
- return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_);
+ return webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_) ==
+ webrtc::RTCErrorType::NONE;
}
protected:
@@ -2660,13 +2665,13 @@
EXPECT_EQ(0U, turn_servers_.size());
stun_servers_.clear();
- EXPECT_TRUE(ParseUrl("turn:hostname"));
+ EXPECT_TRUE(ParseTurnUrl("turn:hostname"));
EXPECT_EQ(0U, stun_servers_.size());
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
turn_servers_.clear();
- EXPECT_TRUE(ParseUrl("turns:hostname"));
+ EXPECT_TRUE(ParseTurnUrl("turns:hostname"));
EXPECT_EQ(0U, stun_servers_.size());
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto);
@@ -2675,7 +2680,7 @@
turn_servers_.clear();
EXPECT_TRUE(ParseUrl(
- "turns:hostname", "", "",
+ "turns:hostname", "username", "password",
PeerConnectionInterface::TlsCertPolicy::kTlsCertPolicyInsecureNoCheck));
EXPECT_EQ(0U, stun_servers_.size());
EXPECT_EQ(1U, turn_servers_.size());
@@ -2693,14 +2698,14 @@
TEST_F(IceServerParsingTest, VerifyDefaults) {
// TURNS defaults
- EXPECT_TRUE(ParseUrl("turns:hostname"));
+ EXPECT_TRUE(ParseTurnUrl("turns:hostname"));
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_EQ(5349, turn_servers_[0].ports[0].address.port());
EXPECT_EQ(cricket::PROTO_TLS, turn_servers_[0].ports[0].proto);
turn_servers_.clear();
// TURN defaults
- EXPECT_TRUE(ParseUrl("turn:hostname"));
+ EXPECT_TRUE(ParseTurnUrl("turn:hostname"));
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_EQ(3478, turn_servers_[0].ports[0].address.port());
EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
@@ -2765,33 +2770,33 @@
// Test parsing the "?transport=xxx" part of the URL.
TEST_F(IceServerParsingTest, ParseTransport) {
- EXPECT_TRUE(ParseUrl("turn:hostname:1234?transport=tcp"));
+ EXPECT_TRUE(ParseTurnUrl("turn:hostname:1234?transport=tcp"));
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_EQ(cricket::PROTO_TCP, turn_servers_[0].ports[0].proto);
turn_servers_.clear();
- EXPECT_TRUE(ParseUrl("turn:hostname?transport=udp"));
+ EXPECT_TRUE(ParseTurnUrl("turn:hostname?transport=udp"));
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_EQ(cricket::PROTO_UDP, turn_servers_[0].ports[0].proto);
turn_servers_.clear();
- EXPECT_FALSE(ParseUrl("turn:hostname?transport=invalid"));
- EXPECT_FALSE(ParseUrl("turn:hostname?transport="));
- EXPECT_FALSE(ParseUrl("turn:hostname?="));
- EXPECT_FALSE(ParseUrl("?"));
+ EXPECT_FALSE(ParseTurnUrl("turn:hostname?transport=invalid"));
+ EXPECT_FALSE(ParseTurnUrl("turn:hostname?transport="));
+ EXPECT_FALSE(ParseTurnUrl("turn:hostname?="));
+ EXPECT_FALSE(ParseTurnUrl("?"));
}
// Test parsing ICE username contained in URL.
TEST_F(IceServerParsingTest, ParseUsername) {
- EXPECT_TRUE(ParseUrl("turn:user@hostname"));
+ EXPECT_TRUE(ParseTurnUrl("turn:user@hostname"));
EXPECT_EQ(1U, turn_servers_.size());
EXPECT_EQ("user", turn_servers_[0].credentials.username);
turn_servers_.clear();
- EXPECT_FALSE(ParseUrl("turn:@hostname"));
- EXPECT_FALSE(ParseUrl("turn:username@"));
- EXPECT_FALSE(ParseUrl("turn:@"));
- EXPECT_FALSE(ParseUrl("turn:user@name@hostname"));
+ EXPECT_FALSE(ParseTurnUrl("turn:@hostname"));
+ EXPECT_FALSE(ParseTurnUrl("turn:username@"));
+ EXPECT_FALSE(ParseTurnUrl("turn:@"));
+ EXPECT_FALSE(ParseTurnUrl("turn:user@name@hostname"));
}
// Test that username and password from IceServer is copied into the resulting
@@ -2809,8 +2814,11 @@
PeerConnectionInterface::IceServer server;
server.urls.push_back("stun:hostname");
server.urls.push_back("turn:hostname");
+ server.username = "foo";
+ server.password = "bar";
servers.push_back(server);
- EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
+ EXPECT_EQ(webrtc::RTCErrorType::NONE,
+ webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
EXPECT_EQ(1U, stun_servers_.size());
EXPECT_EQ(1U, turn_servers_.size());
}
@@ -2822,8 +2830,11 @@
PeerConnectionInterface::IceServer server;
server.urls.push_back("turn:hostname");
server.urls.push_back("turn:hostname2");
+ server.username = "foo";
+ server.password = "bar";
servers.push_back(server);
- EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
+ EXPECT_EQ(webrtc::RTCErrorType::NONE,
+ webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
EXPECT_EQ(2U, turn_servers_.size());
EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
}
diff --git a/webrtc/api/peerconnectioninterface.h b/webrtc/api/peerconnectioninterface.h
index 5f5a58b4..5e1fd32 100644
--- a/webrtc/api/peerconnectioninterface.h
+++ b/webrtc/api/peerconnectioninterface.h
@@ -150,9 +150,10 @@
typedef MetricsObserverInterface UMAObserver;
// Enumeration to represent distinct classes of errors that an application
-// may wish to act upon differently. These roughly map to DOMExceptions in
-// the web API, as described in the comments below.
-enum class RtcError {
+// may wish to act upon differently. These roughly map to DOMExceptions or
+// RTCError "errorDetailEnum" values in the web API, as described in the
+// comments below.
+enum class RTCErrorType {
// No error.
NONE,
// A supplied parameter is valid, but currently unsupported.
@@ -183,9 +184,26 @@
INTERNAL_ERROR,
};
+// Roughly corresponds to RTCError in the web api. Holds an error type and
+// possibly additional information specific to that error.
+//
+// Doesn't contain anything beyond a type now, but will in the future as more
+// errors are implemented.
+class RTCError {
+ public:
+ RTCError() : type_(RTCErrorType::NONE) {}
+ explicit RTCError(RTCErrorType type) : type_(type) {}
+
+ RTCErrorType type() const { return type_; }
+ void set_type(RTCErrorType type) { type_ = type; }
+
+ private:
+ RTCErrorType type_;
+};
+
// Outputs the error as a friendly string.
// Update this method when adding a new error type.
-std::ostream& operator<<(std::ostream& stream, RtcError error);
+std::ostream& operator<<(std::ostream& stream, RTCErrorType error);
class PeerConnectionInterface : public rtc::RefCountInterface {
public:
@@ -319,6 +337,9 @@
}
}
+ bool operator==(const RTCConfiguration& o) const;
+ bool operator!=(const RTCConfiguration& o) const;
+
bool dscp() { return media_config.enable_dscp; }
void set_dscp(bool enable) { media_config.enable_dscp = enable; }
@@ -392,6 +413,9 @@
// If true, ICE role is redetermined when peerconnection sets a local
// transport description that indicates an ICE restart.
bool redetermine_role_on_ice_restart = true;
+ //
+ // Don't forget to update operator== if adding something.
+ //
};
struct RTCOfferAnswerOptions {
@@ -569,14 +593,38 @@
virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() {
return PeerConnectionInterface::RTCConfiguration();
}
+
// Sets the PeerConnection's global configuration to |config|.
+ //
+ // The members of |config| that may be changed are |type|, |servers|,
+ // |ice_candidate_pool_size| and |prune_turn_ports| (though the candidate
+ // pool size can't be changed after the first call to SetLocalDescription).
+ // Note that this means the BUNDLE and RTCP-multiplexing policies cannot be
+ // changed with this method.
+ //
// Any changes to STUN/TURN servers or ICE candidate policy will affect the
// next gathering phase, and cause the next call to createOffer to generate
- // new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
- // cannot be changed with this method.
+ // new ICE credentials, as described in JSEP. This also occurs when
+ // |prune_turn_ports| changes, for the same reasoning.
+ //
+ // If an error occurs, returns false and populates |error| if non-null:
+ // - INVALID_MODIFICATION if |config| contains a modified parameter other
+ // than one of the parameters listed above.
+ // - INVALID_RANGE if |ice_candidate_pool_size| is out of range.
+ // - SYNTAX_ERROR if parsing an ICE server URL failed.
+ // - INVALID_PARAMETER if a TURN server is missing |username| or |password|.
+ // - INTERNAL_ERROR if an unexpected error occurred.
+ //
// TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
// PeerConnectionInterface implement it.
virtual bool SetConfiguration(
+ const PeerConnectionInterface::RTCConfiguration& config,
+ RTCError* error) {
+ return false;
+ }
+ // Version without error output param for backwards compatibility.
+ // TODO(deadbeef): Remove once chromium is updated.
+ virtual bool SetConfiguration(
const PeerConnectionInterface::RTCConfiguration& config) {
return false;
}
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index d6da24e..2da3755 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -321,6 +321,8 @@
using webrtc::ObserverInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
+using webrtc::RTCError;
+using webrtc::RTCErrorType;
using webrtc::RtpReceiverInterface;
using webrtc::RtpSenderInterface;
using webrtc::SdpParseError;
@@ -694,6 +696,17 @@
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(), nullptr);
}
+ // DTLS does not work in a loopback call, so is disabled for most of the
+ // tests in this file.
+ void CreatePeerConnectionWithoutDtls() {
+ FakeConstraints no_dtls_constraints;
+ no_dtls_constraints.AddMandatory(
+ webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
+
+ CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
+ &no_dtls_constraints);
+ }
+
void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
constraints);
@@ -722,17 +735,6 @@
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
port_allocator_ = port_allocator.get();
- // DTLS does not work in a loopback call, so is disabled for most of the
- // tests in this file. We only create a FakeIdentityService if the test
- // explicitly sets the constraint.
- FakeConstraints default_constraints;
- if (!constraints) {
- constraints = &default_constraints;
-
- default_constraints.AddMandatory(
- webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
- }
-
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
bool dtls;
if (FindConstraint(constraints,
@@ -898,7 +900,7 @@
}
void InitiateCall() {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
// Create a local stream with audio&video tracks.
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
CreateOfferReceiveAnswer();
@@ -1105,7 +1107,7 @@
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVoiceStream(kStreamLabel1);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
@@ -1281,7 +1283,7 @@
}
TEST_F(PeerConnectionInterfaceTest, AddStreams) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamLabel1);
AddVoiceStream(kStreamLabel2);
ASSERT_EQ(2u, pc_->local_streams()->count());
@@ -1311,7 +1313,7 @@
// Test that the created offer includes streams we added.
TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
@@ -1355,7 +1357,7 @@
}
TEST_F(PeerConnectionInterfaceTest, RemoveStream) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamLabel1);
ASSERT_EQ(1u, pc_->local_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
@@ -1367,7 +1369,7 @@
// and that the RtpSenders are created correctly.
// Also tests that RemoveTrack removes the tracks from subsequent offers.
TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
// Create a dummy stream, so tracks share a stream label.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamLabel1));
@@ -1442,7 +1444,7 @@
// Test creating senders without a stream specified,
// expecting a random stream ID to be generated.
TEST_F(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
// Create a dummy stream, so tracks share a stream label.
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
@@ -1470,7 +1472,7 @@
}
TEST_F(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamLabel1);
CreateOfferAsLocalDescription();
std::string offer;
@@ -1480,7 +1482,7 @@
}
TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamLabel1);
CreateOfferAsRemoteDescription();
@@ -1490,7 +1492,7 @@
}
TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamLabel1);
CreateOfferAsRemoteDescription();
@@ -1513,7 +1515,7 @@
// Tests that after negotiating an audio only call, the respondent can perform a
// renegotiation that removes the audio stream.
TEST_F(PeerConnectionInterfaceTest, RenegotiateAudioOnly) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVoiceStream(kStreamLabel1);
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
@@ -1526,7 +1528,7 @@
// Test that candidates are generated and that we can parse our own candidates.
TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
// SetRemoteDescription takes ownership of offer.
@@ -1549,7 +1551,7 @@
// Test that CreateOffer and CreateAnswer will fail if the track labels are
// not unique.
TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
// Create a regular offer for the CreateAnswer test later.
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
@@ -1570,7 +1572,7 @@
// Test that we will get different SSRCs for each tracks in the offer and answer
// we created.
TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
// Create a local stream with audio&video tracks having different labels.
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
@@ -1602,7 +1604,7 @@
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
// Create audio stream and add to PeerConnection.
AddVoiceStream(kStreamLabel1);
MediaStreamInterface* stream = pc_->local_streams()->at(0);
@@ -1626,7 +1628,7 @@
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
// Create audio/video stream and add to PeerConnection.
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
MediaStreamInterface* stream = pc_->local_streams()->at(0);
@@ -1645,7 +1647,7 @@
// Test creating a sender with a stream ID, and ensure the ID is populated
// in the offer.
TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
pc_->CreateSender("video", kStreamLabel1);
std::unique_ptr<SessionDescriptionInterface> offer;
@@ -2075,7 +2077,7 @@
// limited set of audio codecs and receive an updated offer with more audio
// codecs, where the added codecs are not supported.
TEST_F(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddVoiceStream("audio_label");
CreateOfferAsLocalDescription();
@@ -2183,6 +2185,17 @@
EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
}
+TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
+ PeerConnectionInterface::RTCConfiguration config;
+ config.prune_turn_ports = false;
+ CreatePeerConnection(config, nullptr);
+ EXPECT_FALSE(port_allocator_->prune_turn_ports());
+
+ config.prune_turn_ports = true;
+ EXPECT_TRUE(pc_->SetConfiguration(config));
+ EXPECT_TRUE(port_allocator_->prune_turn_ports());
+}
+
// Test that when SetConfiguration changes both the pool size and other
// attributes, the pooled session is created with the updated attributes.
TEST_F(PeerConnectionInterfaceTest,
@@ -2203,7 +2216,8 @@
EXPECT_EQ(1UL, session->stun_servers().size());
}
-// Test that after SetLocalDescription, changing the pool size is not allowed.
+// Test that after SetLocalDescription, changing the pool size is not allowed,
+// and an invalid modification error is returned.
TEST_F(PeerConnectionInterfaceTest,
CantChangePoolSizeAfterSetLocalDescription) {
CreatePeerConnection();
@@ -2220,7 +2234,91 @@
// Set local answer; now it's too late.
CreateAnswerAsLocalDescription();
config.ice_candidate_pool_size = 3;
- EXPECT_FALSE(pc_->SetConfiguration(config));
+ RTCError error;
+ EXPECT_FALSE(pc_->SetConfiguration(config, &error));
+ EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
+}
+
+// Test that SetConfiguration returns an invalid modification error if
+// modifying a field in the configuration that isn't allowed to be modified.
+TEST_F(PeerConnectionInterfaceTest,
+ SetConfigurationReturnsInvalidModificationError) {
+ PeerConnectionInterface::RTCConfiguration config;
+ config.bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced;
+ config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
+ config.continual_gathering_policy = PeerConnectionInterface::GATHER_ONCE;
+ CreatePeerConnection(config, nullptr);
+
+ PeerConnectionInterface::RTCConfiguration modified_config = config;
+ modified_config.bundle_policy =
+ PeerConnectionInterface::kBundlePolicyMaxBundle;
+ RTCError error;
+ EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
+ EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
+
+ modified_config = config;
+ modified_config.rtcp_mux_policy =
+ PeerConnectionInterface::kRtcpMuxPolicyRequire;
+ error.set_type(RTCErrorType::NONE);
+ EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
+ EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
+
+ modified_config = config;
+ modified_config.continual_gathering_policy =
+ PeerConnectionInterface::GATHER_CONTINUALLY;
+ error.set_type(RTCErrorType::NONE);
+ EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
+ EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
+}
+
+// Test that SetConfiguration returns a range error if the candidate pool size
+// is negative or larger than allowed by the spec.
+TEST_F(PeerConnectionInterfaceTest,
+ SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) {
+ PeerConnectionInterface::RTCConfiguration config;
+ CreatePeerConnection(config, nullptr);
+
+ config.ice_candidate_pool_size = -1;
+ RTCError error;
+ EXPECT_FALSE(pc_->SetConfiguration(config, &error));
+ EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
+
+ config.ice_candidate_pool_size = INT_MAX;
+ error.set_type(RTCErrorType::NONE);
+ EXPECT_FALSE(pc_->SetConfiguration(config, &error));
+ EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
+}
+
+// Test that SetConfiguration returns a syntax error if parsing an ICE server
+// URL failed.
+TEST_F(PeerConnectionInterfaceTest,
+ SetConfigurationReturnsSyntaxErrorFromBadIceUrls) {
+ PeerConnectionInterface::RTCConfiguration config;
+ CreatePeerConnection(config, nullptr);
+
+ PeerConnectionInterface::IceServer bad_server;
+ bad_server.uri = "stunn:www.example.com";
+ config.servers.push_back(bad_server);
+ RTCError error;
+ EXPECT_FALSE(pc_->SetConfiguration(config, &error));
+ EXPECT_EQ(RTCErrorType::SYNTAX_ERROR, error.type());
+}
+
+// Test that SetConfiguration returns an invalid parameter error if a TURN
+// IceServer is missing a username or password.
+TEST_F(PeerConnectionInterfaceTest,
+ SetConfigurationReturnsInvalidParameterIfCredentialsMissing) {
+ PeerConnectionInterface::RTCConfiguration config;
+ CreatePeerConnection(config, nullptr);
+
+ PeerConnectionInterface::IceServer bad_server;
+ bad_server.uri = "turn:www.example.com";
+ // Missing password.
+ bad_server.username = "foo";
+ config.servers.push_back(bad_server);
+ RTCError error;
+ EXPECT_FALSE(pc_->SetConfiguration(config, &error));
+ EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type());
}
// Test that PeerConnection::Close changes the states to closed and all remote
@@ -2255,7 +2353,7 @@
// Test that PeerConnection methods fails gracefully after
// PeerConnection::Close has been called.
TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
- CreatePeerConnection();
+ CreatePeerConnectionWithoutDtls();
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
@@ -3227,10 +3325,41 @@
EXPECT_TRUE(updated_answer_options.has_video());
}
-TEST(RtcErrorTest, OstreamOperator) {
+TEST(RTCErrorTypeTest, OstreamOperator) {
std::ostringstream oss;
- oss << webrtc::RtcError::NONE << ' '
- << webrtc::RtcError::INVALID_PARAMETER << ' '
- << webrtc::RtcError::INTERNAL_ERROR;
+ oss << webrtc::RTCErrorType::NONE << ' '
+ << webrtc::RTCErrorType::INVALID_PARAMETER << ' '
+ << webrtc::RTCErrorType::INTERNAL_ERROR;
EXPECT_EQ("NONE INVALID_PARAMETER INTERNAL_ERROR", oss.str());
}
+
+// Tests a few random fields being different.
+TEST(RTCConfigurationTest, ComparisonOperators) {
+ PeerConnectionInterface::RTCConfiguration a;
+ PeerConnectionInterface::RTCConfiguration b;
+ EXPECT_EQ(a, b);
+
+ PeerConnectionInterface::RTCConfiguration c;
+ c.servers.push_back(PeerConnectionInterface::IceServer());
+ EXPECT_NE(a, c);
+
+ PeerConnectionInterface::RTCConfiguration d;
+ d.type = PeerConnectionInterface::kRelay;
+ EXPECT_NE(a, d);
+
+ PeerConnectionInterface::RTCConfiguration e;
+ e.audio_jitter_buffer_max_packets = 5;
+ EXPECT_NE(a, e);
+
+ PeerConnectionInterface::RTCConfiguration f;
+ f.ice_connection_receiving_timeout = 1337;
+ EXPECT_NE(a, f);
+
+ PeerConnectionInterface::RTCConfiguration g;
+ g.disable_ipv6 = true;
+ EXPECT_NE(a, g);
+
+ PeerConnectionInterface::RTCConfiguration h(
+ PeerConnectionInterface::RTCConfigurationType::kAggressive);
+ EXPECT_NE(a, h);
+}
diff --git a/webrtc/api/peerconnectionproxy.h b/webrtc/api/peerconnectionproxy.h
index 2110ee2..1609a75 100644
--- a/webrtc/api/peerconnectionproxy.h
+++ b/webrtc/api/peerconnectionproxy.h
@@ -72,6 +72,10 @@
PROXY_METHOD2(void, SetRemoteDescription, SetSessionDescriptionObserver*,
SessionDescriptionInterface*)
PROXY_METHOD0(PeerConnectionInterface::RTCConfiguration, GetConfiguration);
+ PROXY_METHOD2(bool,
+ SetConfiguration,
+ const PeerConnectionInterface::RTCConfiguration&,
+ RTCError*);
PROXY_METHOD1(bool,
SetConfiguration,
const PeerConnectionInterface::RTCConfiguration&);