Some comments//Re: New version of editor draft for webrtc

Hi, all
I am new to WebRTC, and begin with initial study of WebRTC 1.0 specificationhttp://dev.w3.org/2011/webrtc/editor/webrtc.html.

There are some problems I encountered while reading this spec. I would appreciate it if you could reply.

1．“discard” means that codecs can abandon some channels? Or maybe it should be “decode” by mistake?
Location: “4.1 Introduction”, the third paragraph and last sentence. “All of the channels that a codec needs to encode jointly MUST be in the same MediaStreamTrack and the codecs SHOULD be able to encode, or discard, all the channels in the track.”
2．I saw (1) and (3) in this specification, but I didn’t find any link or reference. Besides, there is no (2), but many (3) following” RTCPeerConnection readiness state<http://dev.w3.org/2011/webrtc/editor/webrtc.html> is closed”.
Location1: “4.2.2 MediaStreamTrack”, the third paragraph. “A track in a MediaStream , received with a RTCPeerConnection , MUST have its readyState attribute [GETUSERMEDIA] set to muted (1) until media data arrives.”
Location2: “5. Peer-to-peer connections” under”NOTE” step 4. “If the connection’s RTCPeerConnection readiness state is closed (3), abort these steps”
3. Step 12 of “Simple Call Flow” is addStream(data), but as I know the parameter of addStream(MediaStream stream) is for audio/video not data. By the way, is it possible to add some descriptions for steps of the flow charts?
Location: “9. Call Flow Browser to Browser”, “10. Call Flow Browser to MCU”

In addition, some revisions for editorial errors are attached for your information. Thank you for checking it.

Best Regards,

Yahui
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Yahui Wang
Device Standards & Patents Dept
Huawei Technologies Co., Ltd.
Addr: Huawei Building, No.156 Beiqing Rd. Z-park, Shi-Chuang-Ke-Ji-Shi-Fan-Yuan,
Hai-Dian District, Beijing 100095, P.R. China
Phone: +86 10 60613252
Mobile：+86 13522157152
Email:yahui.wang@huawei.com
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