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/*
* Copyright 2015 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.
*/
#import <Foundation/Foundation.h>
#import <XCTest/XCTest.h>
#include <vector>
#include "rtc_base/gunit.h"
#import "api/peerconnection/RTCConfiguration+Private.h"
#import "api/peerconnection/RTCConfiguration.h"
#import "api/peerconnection/RTCIceServer.h"
#import "helpers/NSString+StdString.h"
@interface RTCConfigurationTest : XCTestCase
@end
@implementation RTCConfigurationTest
- (void)testConversionToNativeConfiguration {
NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
RTC_OBJC_TYPE(RTCIceServer) *server =
[[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
config.iceServers = @[ server ];
config.iceTransportPolicy = RTCIceTransportPolicyRelay;
config.bundlePolicy = RTCBundlePolicyMaxBundle;
config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
const int maxPackets = 60;
const int timeout = 1;
const int interval = 2;
config.audioJitterBufferMaxPackets = maxPackets;
config.audioJitterBufferFastAccelerate = YES;
config.iceConnectionReceivingTimeout = timeout;
config.iceBackupCandidatePairPingInterval = interval;
config.continualGatheringPolicy =
RTCContinualGatheringPolicyGatherContinually;
config.shouldPruneTurnPorts = YES;
config.cryptoOptions =
[[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES
srtpEnableAes128Sha1_32CryptoCipher:YES
srtpEnableEncryptedRtpHeaderExtensions:YES
sframeRequireFrameEncryption:YES];
config.rtcpAudioReportIntervalMs = 2500;
config.rtcpVideoReportIntervalMs = 3750;
std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
nativeConfig([config createNativeConfiguration]);
EXPECT_TRUE(nativeConfig.get());
EXPECT_EQ(1u, nativeConfig->servers.size());
webrtc::PeerConnectionInterface::IceServer nativeServer =
nativeConfig->servers.front();
EXPECT_EQ(1u, nativeServer.urls.size());
EXPECT_EQ("stun:stun1.example.net", nativeServer.urls.front());
EXPECT_EQ(webrtc::PeerConnectionInterface::kRelay, nativeConfig->type);
EXPECT_EQ(webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle,
nativeConfig->bundle_policy);
EXPECT_EQ(webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate,
nativeConfig->rtcp_mux_policy);
EXPECT_EQ(webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled,
nativeConfig->tcp_candidate_policy);
EXPECT_EQ(webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost,
nativeConfig->candidate_network_policy);
EXPECT_EQ(maxPackets, nativeConfig->audio_jitter_buffer_max_packets);
EXPECT_EQ(true, nativeConfig->audio_jitter_buffer_fast_accelerate);
EXPECT_EQ(timeout, nativeConfig->ice_connection_receiving_timeout);
EXPECT_EQ(interval, nativeConfig->ice_backup_candidate_pair_ping_interval);
EXPECT_EQ(webrtc::PeerConnectionInterface::GATHER_CONTINUALLY,
nativeConfig->continual_gathering_policy);
EXPECT_EQ(true, nativeConfig->prune_turn_ports);
EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_gcm_crypto_suites);
EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_aes128_sha1_32_crypto_cipher);
EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_encrypted_rtp_header_extensions);
EXPECT_EQ(true, nativeConfig->crypto_options->sframe.require_frame_encryption);
EXPECT_EQ(2500, nativeConfig->audio_rtcp_report_interval_ms());
EXPECT_EQ(3750, nativeConfig->video_rtcp_report_interval_ms());
}
- (void)testNativeConversionToConfiguration {
NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
RTC_OBJC_TYPE(RTCIceServer) *server =
[[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
config.iceServers = @[ server ];
config.iceTransportPolicy = RTCIceTransportPolicyRelay;
config.bundlePolicy = RTCBundlePolicyMaxBundle;
config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
const int maxPackets = 60;
const int timeout = 1;
const int interval = 2;
config.audioJitterBufferMaxPackets = maxPackets;
config.audioJitterBufferFastAccelerate = YES;
config.iceConnectionReceivingTimeout = timeout;
config.iceBackupCandidatePairPingInterval = interval;
config.continualGatheringPolicy =
RTCContinualGatheringPolicyGatherContinually;
config.shouldPruneTurnPorts = YES;
config.cryptoOptions =
[[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES
srtpEnableAes128Sha1_32CryptoCipher:NO
srtpEnableEncryptedRtpHeaderExtensions:NO
sframeRequireFrameEncryption:NO];
config.rtcpAudioReportIntervalMs = 1500;
config.rtcpVideoReportIntervalMs = 2150;
webrtc::PeerConnectionInterface::RTCConfiguration *nativeConfig =
[config createNativeConfiguration];
RTC_OBJC_TYPE(RTCConfiguration) *newConfig =
[[RTC_OBJC_TYPE(RTCConfiguration) alloc] initWithNativeConfiguration:*nativeConfig];
EXPECT_EQ([config.iceServers count], newConfig.iceServers.count);
RTC_OBJC_TYPE(RTCIceServer) *newServer = newConfig.iceServers[0];
RTC_OBJC_TYPE(RTCIceServer) *origServer = config.iceServers[0];
EXPECT_EQ(origServer.urlStrings.count, server.urlStrings.count);
std::string origUrl = origServer.urlStrings.firstObject.UTF8String;
std::string url = newServer.urlStrings.firstObject.UTF8String;
EXPECT_EQ(origUrl, url);
EXPECT_EQ(config.iceTransportPolicy, newConfig.iceTransportPolicy);
EXPECT_EQ(config.bundlePolicy, newConfig.bundlePolicy);
EXPECT_EQ(config.rtcpMuxPolicy, newConfig.rtcpMuxPolicy);
EXPECT_EQ(config.tcpCandidatePolicy, newConfig.tcpCandidatePolicy);
EXPECT_EQ(config.candidateNetworkPolicy, newConfig.candidateNetworkPolicy);
EXPECT_EQ(config.audioJitterBufferMaxPackets, newConfig.audioJitterBufferMaxPackets);
EXPECT_EQ(config.audioJitterBufferFastAccelerate, newConfig.audioJitterBufferFastAccelerate);
EXPECT_EQ(config.iceConnectionReceivingTimeout, newConfig.iceConnectionReceivingTimeout);
EXPECT_EQ(config.iceBackupCandidatePairPingInterval,
newConfig.iceBackupCandidatePairPingInterval);
EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
EXPECT_EQ(config.cryptoOptions.srtpEnableGcmCryptoSuites,
newConfig.cryptoOptions.srtpEnableGcmCryptoSuites);
EXPECT_EQ(config.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher,
newConfig.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher);
EXPECT_EQ(config.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions,
newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
newConfig.cryptoOptions.sframeRequireFrameEncryption);
EXPECT_EQ(config.rtcpAudioReportIntervalMs, newConfig.rtcpAudioReportIntervalMs);
EXPECT_EQ(config.rtcpVideoReportIntervalMs, newConfig.rtcpVideoReportIntervalMs);
}
- (void)testDefaultValues {
RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
EXPECT_EQ(config.cryptoOptions, nil);
}
@end
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