RTCConfigurationTest.mm

/*

 *  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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