webrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm
Benjamin Wright 8c27ccac75 Promotoing webrtc::CryptoOptions to RTCConfiguration.
With the expanding use cases for webrtc::CryptoOptions it makes more sense for
it to be be available per peer connection instead of only as a factory option.

To support backwards compatability for now this code will support the factory
method of setting crypto options by default. However it will completely
overwrite these settings if an RTCConfiguration.crypto_options is provided.

Got LGTM offline from Sami, adding him to TBR if he has any further comments.

TBR=sakal@webrtc.org

Bug: webrtc:9891
Change-Id: I86914cab69284ad82afd7285fd84ec5f4f2c4986
Reviewed-on: https://webrtc-review.googlesource.com/c/107029
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25375}
2018-10-25 17:59:48 +00:00

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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>
#include <vector>
#include "rtc_base/gunit.h"
#import "api/peerconnection/RTCConfiguration+Private.h"
#import "api/peerconnection/RTCConfiguration.h"
#import "api/peerconnection/RTCCryptoOptions.h"
#import "api/peerconnection/RTCIceServer.h"
#import "api/peerconnection/RTCMediaConstraints.h"
#import "api/peerconnection/RTCPeerConnection.h"
#import "api/peerconnection/RTCPeerConnectionFactory.h"
#import "helpers/NSString+StdString.h"
@interface RTCPeerConnectionTest : NSObject
- (void)testConfigurationGetter;
@end
@implementation RTCPeerConnectionTest
- (void)testConfigurationGetter {
NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
RTCConfiguration *config = [[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 = 1500;
const int interval = 2000;
config.audioJitterBufferMaxPackets = maxPackets;
config.audioJitterBufferFastAccelerate = YES;
config.iceConnectionReceivingTimeout = timeout;
config.iceBackupCandidatePairPingInterval = interval;
config.continualGatheringPolicy =
RTCContinualGatheringPolicyGatherContinually;
config.shouldPruneTurnPorts = YES;
config.activeResetSrtpParams = YES;
config.cryptoOptions = [[RTCCryptoOptions alloc] initWithSrtpEnableGcmCryptoSuites:YES
srtpEnableAes128Sha1_32CryptoCipher:YES
srtpEnableEncryptedRtpHeaderExtensions:NO
sframeRequireFrameEncryption:NO];
RTCMediaConstraints *contraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{}
optionalConstraints:nil];
RTCPeerConnectionFactory *factory = [[RTCPeerConnectionFactory alloc] init];
RTCConfiguration *newConfig;
@autoreleasepool {
RTCPeerConnection *peerConnection =
[factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
newConfig = peerConnection.configuration;
EXPECT_TRUE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:100000]
currentBitrateBps:[NSNumber numberWithInt:5000000]
maxBitrateBps:[NSNumber numberWithInt:500000000]]);
EXPECT_FALSE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:2]
currentBitrateBps:[NSNumber numberWithInt:1]
maxBitrateBps:nil]);
}
EXPECT_EQ([config.iceServers count], [newConfig.iceServers count]);
RTCIceServer *newServer = newConfig.iceServers[0];
RTCIceServer *origServer = config.iceServers[0];
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.activeResetSrtpParams, newConfig.activeResetSrtpParams);
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);
}
@end
TEST(RTCPeerConnectionTest, ConfigurationGetterTest) {
@autoreleasepool {
RTCPeerConnectionTest *test = [[RTCPeerConnectionTest alloc] init];
[test testConfigurationGetter];
}
}