[Tsv-art] Tsvart last call review of draft-ietf-mmusic-t140-usage-data-channel-11

[Michael Scharf via Datatracker <[e-mail]>]

Reviewer: Michael Scharf
Review result: Ready with Nits

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This document specifies how a WebRTC data channel can transport ITU-T T.140
text conversation.

Disclaimer: I am really not familiar with this area and this particular use
case. For instance, I have never heard of ITU-T T.140 before reviewing this
document.

The draft is pretty short. In general, the basic protocol mechanisms are well
described. However, the document is rather vague when it comes to some details.
Thus, I have some comments (well, actually questions). Maybe some of this is
obvious to the domain experts - but not to me.

1/ I wonder why Section 4.2.1 does not include any normative statements on how
to handle the maximum character transmission rate ('cps' attribute). RFC 4103
states that "In receipt of this parameter, devices MUST adhere to the request
by transmitting characters at a rate at or below the specified <integer>
value." Isn't a similar statement needed in this document?

2/ Also, it is not really clear from the document what would happen if a peer
exceeds this maximum character transmission rate (or the rate allowed by
congestion/flow control). What happens if the sender types faster than the
'cps' attribute (say, an automated chat bot)? I guess characters would be
dropped at the sender? In that case, no missing text markers would be displayed
in the receiver, right?

3/ Section 5.3. "Data Buffering" includes the following statement: "As
described in [T140], buffering can be used to reduce overhead, with the maximum
buffering time being 500 ms.  It can also be used for staying within the
maximum character transmission rate (Section 4.2), if such has been provided by
the peer." I don't understand the second sentence. At first sight, enforcing
the 'cps' attribute does not only require a buffer, but also some sort of rate
shaper/policer (e.g., token bucket or the like). Do I miss something?

4/ Also in Section 5.3 is written: "An implementation needs to take the user
requirements for smooth flow and low latency in real-time text conversation
into consideration when assigning a buffer time.  It is RECOMMENDED to use the
default transmission interval of 300 milliseconds [RFC4103], or lower, for
T.140 data channels". What is meant here by "or lower"? Does the document want
to recommend values much smaller than 300 ms, say, 1 ms? As explained in RFC
4103, this could increase the overhead and bitrate, right? The absolute rate
values are relatively small for large parts of today's Internet, but couldn't
this text conversation be particularly useful in scenarios with very small
capacity of links (i.e., kbps range)?

5/ Section 5.4 mandates: "Retransmission of already successfully transmitted
T140blocks MUST be avoided, and missing text markers [T140ad1] SHOULD be
inserted in the received data stream where loss is detected or suspected." I
believe a better wording for the MUST would be "... sucessfully received
T140blocks ...", albeit the document does not detail how an implementation can
indeed fulfill this MUST. Regarding the SHOULD, I assume that "loss suspected"
could be deterrmined by a heuristic. Could such a heuristic fail and result in
spurious missing text markers? If so, would a SHOULD be reasonable for that?

I don't know whether interoperable running code would need a specification with
that level of detail. Thus, I flag these questions only as "nits".