title: WebRTC iOS Native app audio problems with latest revisions
body: <p>i'm trying to build iOS native audio/video chat and stucked on audio. The sound is lagging, missing some parts and distorting. I tried this with r9919 and the latest one (r10184) builded by pristine build scripts. But when I tried to use older versions (r8444, r8926, r9132 and r9137) taken from PerchRTC Demo project (<a href="https://github.com/perchco/perchrtc" rel="nofollow">https://github.com/perchco/perchrtc</a> - fat lib and public headers) everything seems ok (except little echo). What preactions should I perform to achieve at least the same streaming quality as (r8444, r8926, r9132 and r9137) provide? 
  I also tried to use different audio codecs and different media constraints, still have no luck. 
  I create audio connection this way</p>
  
  <pre><code>RTCMediaStream *localStream = [_pcFactory mediaStreamWithLabel:@"ARDAMS"];
  
  RTCAudioTrack *localAudioTrack = [_pcFactory audioTrackWithID:@"ARDAMSa0"];
  
  localAudioTrack.delegate = self;
  
  [localStream addAudioTrack:localAudioTrack];
  
  [self.peerConnection addStream:localStream];
  
  [self.peerConnection createOfferWithDelegate:self constraints:_constraints];
  </code></pre>
  
  <p>using this constraints (tried different combinations)</p>
  
  <p>mandatoryConstraints = @[</p>
  
  <pre><code>                                 [[RTCPair alloc] initWithKey:@"OfferToReceiveAudio" value:@"true"]
  
                                   ];
  </code></pre>
  
  <p>optionalConstraints = @[</p>
  
  <pre><code>                                [[RTCPair alloc] initWithKey:@"internalSctpDataChannels" value:@"true"],
  
                                  [[RTCPair alloc] initWithKey:@"DtlsSrtpKeyAgreement" value:@"true"]
  
                                  ];
  </code></pre>
  
  <p>Also I tried to different manipulations with AVAudioSession before obtaining audioTrack with no luck:</p>
  
  <pre><code>AVAudioSession *audioSession = [AVAudioSession sharedInstance];
  
  [audioSession setCategory:AVAudioSessionCategoryPlayAndRecord error:&amp;categoryError];
  
  [audioSession setMode:AVAudioSessionModeVoiceChat error:&amp;modeError];
  
  [audioSession overrideOutputAudioPort:AVAudioSessionPortOverrideNone error:&amp;overrideError];
  
  [audioSession setActive:YES withOptions:AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation error:&amp;activeError];
  </code></pre>
  
  <p>SDP descriptions look similar in lagging and not lagging builds (this one is offer - a=setup:actpass,answer has a=setup:active):</p>
  
  <pre><code>sdp = "v=0
  
  o=- 7772121714021031999 2 IN IP4 127.0.0.1
  
  s=-
  
  t=0 0
  
  a=group:BUNDLE audio
  
  a=msid-semantic: WMS ARDAMS
  
  m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 102 0 8 106 105 13 127 126
  
  c=IN IP4 0.0.0.0
  
  a=rtcp:9 IN IP4 0.0.0.0
  
  a=ice-ufrag:XLpFvB+JEaSN7tww
  
  a=ice-pwd:9hoMfb7AJ9jC6Weej7qqTWkT
  
  a=fingerprint:sha-256 AE:73:33:DD:31:CA:84:5A:96:4D:68:27:A0:23:82:3C:08:3B:7F:7B:A2:FE:91:1D:A7:3A:1F:2A:58:4B:FF:A2
  
  a=setup:actpass
  
  a=mid:audio
  
  a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  
  a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
  
  a=sendrecv
  
  a=rtcp-mux
  
  a=rtpmap:111 opus/48000/2
  
  a=fmtp:111 minptime=10; useinbandfec=1
  
  a=rtpmap:103 ISAC/16000
  
  a=rtpmap:104 ISAC/32000
  
  a=rtpmap:9 G722/8000
  
  a=rtpmap:102 ILBC/8000
  
  a=rtpmap:0 PCMU/8000
  
  a=rtpmap:8 PCMA/8000
  
  a=rtpmap:106 CN/32000
  
  a=rtpmap:105 CN/16000
  
  a=rtpmap:13 CN/8000
  
  a=rtpmap:127 red/8000
  
  a=rtpmap:126 telephone-event/8000
  
  a=maxptime:60
  
  a=ssrc:2797474154 cname:43zbAmj6VvYHT31F
  
  a=ssrc:2797474154 msid:ARDAMS ARDAMSa0
  
  a=ssrc:2797474154 mslabel:ARDAMS
  
  a=ssrc:2797474154 label:ARDAMSa0
  </code></pre>
  
  <p>Any suggestions?</p>