Telechat Review of draft-ietf-sipcore-sip-websocket-08
review-ietf-sipcore-sip-websocket-08-genart-telechat-krishnan-2013-06-24-00

Request Review of draft-ietf-sipcore-sip-websocket
Requested rev. no specific revision (document currently at 10)
Type Telechat Review
Team General Area Review Team (Gen-ART) (genart)
Deadline 2013-06-11
Requested 2013-06-06
Draft last updated 2013-06-24
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-telechat-krishnan-2013-06-24
Reviewed rev. 08 (document currently at 10)
Review result Ready
Review completed: 2013-06-24

Review
review-ietf-sipcore-sip-websocket-08-genart-telechat-krishnan-2013-06-24

I have been selected as the General Area Review Team (Gen-ART)
reviewer for this draft (for background on Gen-ART, please see


http://www.alvestrand.no/ietf/gen/art/gen-art-FAQ.html

).

Please wait for direction from your document shepherd
or AD before posting a new version of the draft.

Document:  draft-ietf-sipcore-sip-websocket-08.txt
Reviewer: Suresh Krishnan
Review Date: 2013/06/10
IESG Telechat date: 2013/06/13

Summary: This draft is almost ready for publication as a Proposed
Standard, but has some minor issues that need to be fixed as identified
in my last call review dated 2013/04/16. One of the issues I raised was
clarified as common practice in the RAI wgs and I have removed it from
the list. The authors had agreed to fix the following issues but I have
not seen an updated draft yet.

Minor
=====

* 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 5.2.2

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.

Thanks
Suresh