Time Division Multiplexing over IP (TDMoIP)
draft-ietf-pwe3-tdmoip-06
Yes
(Mark Townsley)
No Objection
(Bill Fenner)
(David Kessens)
(Jari Arkko)
(Lisa Dusseault)
(Ross Callon)
(Russ Housley)
Note: This ballot was opened for revision 06 and is now closed.
Mark Townsley Former IESG member
Yes
Yes
()
Unknown
Bill Fenner Former IESG member
No Objection
No Objection
()
Unknown
Brian Carpenter Former IESG member
(was Discuss)
No Objection
No Objection
(2006-10-09)
Unknown
From discussion between Gen-ART reviewer Joel Halpern and Yakov Stein: The introduction does not tell me what this document is about. A few sentences along the lines of the technical summary in the tracker would help. [YJS] The description is in the Abstract and hinted at in the first paragraph of the Introducton. I will add a few sentences to the first paragraph to make this more clear. ----------- The paragraph on SAToP in section 2.seems to characterize a loss rate of one fifth of one percent as an "extremely reliable and overprovisioned PSN." While I do not want to get in to an argument about what is acceptable, it is to be noted that even TCP will have performance problems with a link with a 0.2% loss rate. ISPs with loss rates much lower than that are still not usually referred to as "extremely reliable." I would recommend simply removing the sentence. [YJS] This number come up from two sources. 1) documents presented to the ITU question that worked in parallel to PWE3 showing that this is indeed a kind of dividing line between well-engineered networks and best-effort ones. 2) empirical results that the user experience of circuit emulation with AIS substitiution drops at this level (see draft-stein-pwe3-tdm-packetloss-01.txt now expired). In any case, we are continually asked to characterize when to stop using SAToP and go over to one of the structure-aware methods. This is as good a break-point as any. For these reasons I would suggest keeping this sentence. BC: I disagree. It's almost as though people working in the ITU are wishing to characterize the Internet as horribly unreliable, but that can't be true can it? Better to simply drop the specific percentage, I think.
Cullen Jennings Former IESG member
(was Discuss)
No Objection
No Objection
(2006-10-11)
Unknown
The statement about customary practice is to operate jitter buffer half full would lead to much larger voice latency that would be needed in most cases. I would like to talk to authors on what they mean by this - perhaps we are just talking past each other but as it is, it does not seem correct.
Dan Romascanu Former IESG member
No Objection
No Objection
(2006-10-12)
Unknown
This document should be consistent with the requirements and architecture defined in RFC3916 and RFC4197. RFC3916 includes in Section 6 a statement that 'Each PWE3 approach SHOULD provide some mechanisms for network operators to manage the emulated service' and then goes into more detailed requirements about how such mechanisms could be implemented. I am surprised about the lack of any managemement or operational considerations section in this document in order to show how the requirements in Section 6 of RFC3916 are supposed to be addressed.
David Kessens Former IESG member
No Objection
No Objection
()
Unknown
Jari Arkko Former IESG member
No Objection
No Objection
()
Unknown
Lars Eggert Former IESG member
No Objection
No Objection
(2006-10-11)
Unknown
Lisa Dusseault Former IESG member
No Objection
No Objection
()
Unknown
Magnus Westerlund Former IESG member
(was Discuss)
No Objection
No Objection
(2006-10-12)
Unknown
Section 4.1: UDP Checksum (16 bits) is the checksum of UDP/IP header and data. If not computed it must be set to zero. Does there exist any other error checking of the packet? If not I would strongly recommend against setting the checksum to 0.
Ross Callon Former IESG member
No Objection
No Objection
()
Unknown
Russ Housley Former IESG member
No Objection
No Objection
()
Unknown