talk/app/webrtc/objc/RTCPeerConnectionInterface.mm

/*
 * libjingle
 * Copyright 2015 Google Inc.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions are met:
 *
 *  1. Redistributions of source code must retain the above copyright notice,
 *     this list of conditions and the following disclaimer.
 *  2. Redistributions in binary form must reproduce the above copyright notice,
 *     this list of conditions and the following disclaimer in the documentation
 *     and/or other materials provided with the distribution.
 *  3. The name of the author may not be used to endorse or promote products
 *     derived from this software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */

#import "talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h"

#import "talk/app/webrtc/objc/RTCEnumConverter.h"
#import "talk/app/webrtc/objc/RTCICEServer+Internal.h"
#import "talk/app/webrtc/objc/public/RTCLogging.h"

@implementation RTCConfiguration

@synthesize iceTransportsType = _iceTransportsType;
@synthesize iceServers = _iceServers;
@synthesize bundlePolicy = _bundlePolicy;
@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
@synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
@synthesize iceBackupCandidatePairPingInterval = _iceBackupCandidatePairPingInterval;
@synthesize keyType = _keyType;

- (instancetype)init {
  if (self = [super init]) {
    // Copy defaults.
    webrtc::PeerConnectionInterface::RTCConfiguration config;
    _iceTransportsType = [RTCEnumConverter iceTransportsTypeForNativeEnum:config.type];
    _bundlePolicy = [RTCEnumConverter bundlePolicyForNativeEnum:config.bundle_policy];
    _rtcpMuxPolicy = [RTCEnumConverter rtcpMuxPolicyForNativeEnum:config.rtcp_mux_policy];
    _tcpCandidatePolicy =
        [RTCEnumConverter tcpCandidatePolicyForNativeEnum:config.tcp_candidate_policy];
    _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
    _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
    _iceBackupCandidatePairPingInterval = config.ice_backup_candidate_pair_ping_interval;
    _keyType = kRTCEncryptionKeyTypeECDSA;
  }
  return self;
}

- (instancetype)initWithIceTransportsType:(RTCIceTransportsType)iceTransportsType
                             bundlePolicy:(RTCBundlePolicy)bundlePolicy
                            rtcpMuxPolicy:(RTCRtcpMuxPolicy)rtcpMuxPolicy
                       tcpCandidatePolicy:(RTCTcpCandidatePolicy)tcpCandidatePolicy
              audioJitterBufferMaxPackets:(int)audioJitterBufferMaxPackets
            iceConnectionReceivingTimeout:(int)iceConnectionReceivingTimeout
       iceBackupCandidatePairPingInterval:(int)iceBackupCandidatePairPingInterval {
  if (self = [super init]) {
    _iceTransportsType = iceTransportsType;
    _bundlePolicy = bundlePolicy;
    _rtcpMuxPolicy = rtcpMuxPolicy;
    _tcpCandidatePolicy = tcpCandidatePolicy;
    _audioJitterBufferMaxPackets = audioJitterBufferMaxPackets;
    _iceConnectionReceivingTimeout = iceConnectionReceivingTimeout;
    _iceBackupCandidatePairPingInterval = iceBackupCandidatePairPingInterval;
  }
  return self;
}

#pragma mark - Private

- (webrtc::PeerConnectionInterface::RTCConfiguration)nativeConfiguration {
  webrtc::PeerConnectionInterface::RTCConfiguration nativeConfig;
  nativeConfig.type = [RTCEnumConverter nativeEnumForIceTransportsType:_iceTransportsType];
  for (RTCICEServer *iceServer : _iceServers) {
    nativeConfig.servers.push_back(iceServer.iceServer);
  }
  nativeConfig.bundle_policy = [RTCEnumConverter nativeEnumForBundlePolicy:_bundlePolicy];
  nativeConfig.rtcp_mux_policy = [RTCEnumConverter nativeEnumForRtcpMuxPolicy:_rtcpMuxPolicy];
  nativeConfig.tcp_candidate_policy =
      [RTCEnumConverter nativeEnumForTcpCandidatePolicy:_tcpCandidatePolicy];
  nativeConfig.audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
  nativeConfig.ice_connection_receiving_timeout = _iceConnectionReceivingTimeout;
  nativeConfig.ice_backup_candidate_pair_ping_interval = _iceBackupCandidatePairPingInterval;
  if (_keyType == kRTCEncryptionKeyTypeECDSA) {
    rtc::scoped_ptr<rtc::SSLIdentity> identity(
        rtc::SSLIdentity::Generate(webrtc::kIdentityName, rtc::KT_ECDSA));
    if (identity) {
      nativeConfig.certificates.push_back(
          rtc::RTCCertificate::Create(std::move(identity)));
    } else {
      RTCLogWarning(@"Failed to generate ECDSA identity. RSA will be used.");
    }
  }
  return nativeConfig;
}

@end