Last Call Review of draft-ietf-sipcore-sip-websocket-08

Request Review of draft-ietf-sipcore-sip-websocket
Requested rev. no specific revision (document currently at 10)
Type Last Call Review
Team General Area Review Team (Gen-ART) (genart)
Deadline 2013-04-15
Requested 2013-04-04
Authors Inaki Castillo, Jose Millan, Victor Pascual
Draft last updated 2013-04-29
Completed reviews Genart Last Call review of -08 by Suresh Krishnan (diff)
Genart Telechat review of -08 by Suresh Krishnan (diff)
Secdir Last Call review of -08 by Yaron Sheffer (diff)
Assignment Reviewer Suresh Krishnan 
State Completed
Review review-ietf-sipcore-sip-websocket-08-genart-lc-krishnan-2013-04-29
Reviewed rev. 08 (document currently at 10)
Review result Ready with Issues
Review completed: 2013-04-29


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Please resolve these comments along with any other Last Call comments
you may receive.

Summary: This draft is almost ready for publication as a Proposed
Standard, but has some minor issues that need to be fixed.

* Section 4.1

There is no error handling specified. i.e. What happens if the server
does not send a 101 reply with sip in the Sec-WebSocket-Protocol header?

* Section 5.2

The updated ABNF provided in section 5.2.1. is confusing. e.g. RFC4168
specifies the entire transport rule as follows

transport         =  "UDP" / "TCP" / "TLS" / "SCTP" / "TLS-SCTP"
                        / other-transport

while this document just uses an incremental alternative and calls it an
updated rule. I would have expected an *updated* rule to say

transport         =  "UDP" / "TCP" / "TLS" / "SCTP" / "TLS-SCTP" / "WS"
		        / "WSS" / other-transport

* Section 5.2.2

RFC4168 does not update transport-param. Is this a copy and paste error?

* Section 5.2.4

Maybe it is just me, but the way I read it, this section leaves the
client without any mandatory to implement transport protocol for SIP
since it relaxes the need to implement UDP and TCP, as well as allowing
the implementation of the websocket transport using a MAY.