webRTC sdp b=AS:600 x-google-min-bitrate 带宽码率那些事

发送端两个sdp片段：SetLocalDescription，SetremoteDescription;

SetLocalDescription:
b=AS:600
a=rtcp:9 IN IP4 0.0.0.0
a=ice-ufrag:gY3K
a=ice-pwd:1HSgJcvP5SkTGcJ8uiixrdqe
a=ice-options:trickle
a=fingerprint:sha-256 BC:7A:18:73:C5:BF:A3:9D:06:BC:13:A6:AC:99:0C:6B:BE:C8:6A:B5:DF:78:2D:C9:CE:3C:22:0E:2B:02:53:20
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendonly
a=msid:aG6LXeJdnm3w9XiAXNENCH2AsDWvZKmUrzot 3f2908cd-12b0-4158-a9fc-9cf7c285993f

SetremoteDescription:
b=AS:600
a=rtpmap:125 H264/90000
a=rtpmap:107 rtx/90000
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;x-google-min-bitrate=400;x-google-max-bitrate=600;profile-level-id=42e01f
a=fmtp:107 apt=125
a=rtcp-fb:125 ccm fir
a=rtcp-fb:125 nack
a=rtcp-fb:125 nack pli
a=rtcp-fb:125 goog-remb
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=setup:active
a=mid:1
a=recvonly
a=ice-ufrag:4689494443441386ohoc
a=ice-pwd:orts7pigqcfgxy51906986jdz5wtizq2
a=ice-options:renomination
a=rtcp-mux
a=rtcp-rsize

webrtc::VideoEncoderConfig
WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
    const VideoCodec& codec) const {
  RTC_DCHECK_RUN_ON(&thread_checker_);
  webrtc::VideoEncoderConfig encoder_config;
  encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name);
  encoder_config.video_format =
      webrtc::SdpVideoFormat(codec.name, codec.params);

  bool is_screencast = parameters_.options.is_screencast.value_or(false);
  if (is_screencast) {
    encoder_config.min_transmit_bitrate_bps =
        1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0);
    encoder_config.content_type =
        webrtc::VideoEncoderConfig::ContentType::kScreen;
  } else {
    encoder_config.min_transmit_bitrate_bps = 0;
    encoder_config.content_type =
        webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;
  }

  // By default, the stream count for the codec configuration should match the
  // number of negotiated ssrcs. But if the codec is blacklisted for simulcast
  // or a screencast (and not in simulcast screenshare experiment), only
  // configur