I have reviewed this document as part of the Operational directorate's ongoing effort to review all IETF documents being processed by the IESG.  These comments were written with the intent of improving the operational aspects of the IETF drafts. Comments that are not addressed in last call may be included in AD reviews during the IESG review.  Document editors and WG chairs should treat these comments just like any other last call comments.     Status:  Almost ready to go, 2 minor concerns, NIT   General comment:  Document and concept are generally clear.  Thank you for providing a simple solution to this problem.     Caveat:  My expertise is at lower end of the stack.  I cross referenced all the WebRTC documentation, but I’ve missed how implementations provide feedback that this protocols is up and working.  Therefore, I’ve indicated the operational issues as a set of questions for the authors to consider.   Minor concern: 1)       Error handling: Is it possible that the msid-value, msid-id, and msid-appdata can be inserted, and then received with values that are not valid  (1*64token-char]? a.         If so, an error sequence is necessary.  b.       If not, it is important to explain why not Add section to 3.2 2)       Operational  issues:  A few questions for your to consider to provide the link to operational issues.   Normal operational:  How does the person who is utilizing this protocol in WebRTC situation check the status of the protocols?  Is it part of the WebRTC status information that the implementations provide?  If so, is there any common management parameters that you can suggest?  Is this in another document in IETF or W3C?   Error operations:  If you can have errors, how does the person who utilizes this protocol in WebRTC  find out the error rate.  Again, is it part of the WebRTC status information on errors?  Is it in another document W3C?       Editorial NITS: Page 7, section 3.1   Paragraph 2:  double “,” in the section highlighted makes this sentence’s meaning unclear. Are these two sub-thoughts? If not two sub-thoughts, then perhaps the /new/ suggested text.      When MSID is used, the only time this can happen is when, at a time    subsequent to the initial negotiation, a negotiation is performed    where the answerer adds a MediaStreamTrack t o an already established    connection and starts sending data before the answer is received by    the offerer.  For initial negotiation, packets won't flow until the    ICE candidates and fingerprints have been exchanged, so this is not    an issue.   /new suggested/ When MSID is used, the only time this can happen is at a time    subsequent to the initial negotiation, /   Paragraph 3   Pagination makes the following text difficult.  Repagination in /new/ may help.  Or it may highlight where I was confused by your document.   /old/ The recipient of those packets will perform the following steps:      o  When RTP packets are initially received, it will create an       appropriate MediaStreamTrack based on the type of the media       (carried in PayloadType), and use the MID RTP header extension       [ I-D.ietf-mmusic-sdp-bundle-negotiation ] (if present) to associate       the RTP packets with a specific media section.  If the connection       is not in the RTCSignalingState "stable", it will wait at this       point.      o  When the connection is in the RTCSignalingState "stable", it will       look at the relevant media section to find the msid attribute.      o  If there is an msid attribute, it will use that attribute to       populate the "id" field of the MediaStreamTrack and associated       MediaStreams, as described above.      o  If there is no msid attribute, the identifier of the       MediaStreamTrack will be set to a randomly generated string, and       it will be signalled as being part of a MediaStream with the       WebIDL "label" attribute set to "Non-WebRTC stream".      o  After deciding on the "id" field to be applied to the       MediaStreamTrack, the track will be signalled to the user. /   /new/ The recipient of those packets will perform the following steps:      o  When RTP packets are initially received, it will create an       appropriate MediaStreamTrack based on the type of the media       (carried in PayloadType), and use the MID RTP header extension       [ I-D.ietf-mmusic-sdp-bundle-negotiation ] (if present) to associate       the RTP packets with a specific media section.  -     If the connection is not in the RTCSignalingState "stable", it will wait at this point. -     When the connection is in the RTCSignalingState "stable", it will look at the relevant media section to find the msid attribute.   ·          Looking a Media section: o     If there is an msid attribute, it will use that attribute to populate the "id" field of the MediaStreamTrack and associated MediaStreams, as described above. o     If there is no msid attribute, the identifier of the MediaStreamTrack will be set to a randomly generated string, and it will be signalled as being part of a MediaStream with the  WebIDL "label" attribute set to "Non-WebRTC stream".      o  After deciding on the "id" field to be applied to the       MediaStreamTrack, the track will be signalled to the user. /