
From Even.roni@huawei.com  Thu Sep  1 02:20:32 2011
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Date: Thu, 01 Sep 2011 12:20:57 +0300
From: Roni Even <Even.roni@huawei.com>
In-reply-to: <4E5DE6D1.3070702@glycon.org>
To: =?iso-8859-1?Q?'st=E9phane_bryant'?= <stephane@glycon.org>, p2psip@ietf.org
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Subject: Re: [P2PSIP] structure definition issue in draft-zong-p2psip-drr-00
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Hi Stephane,
Yes you are right I thought this meant the number of elements in the
structure but now I see from the base draft that it is the length of the
structure. 
Since destination may have 1 or 2 elements I think that the following will
be enough. 

struct {
           RouteMode               routemode;
           OverlayLink             transport;
           IpAddressPort          ipaddressport;
           Destination             destination<1..2^8-1>;
   } ExtensiveRoutingModeOption


Regards
Roni


> -----Original Message-----
> From: p2psip-bounces@ietf.org [mailto:p2psip-bounces@ietf.org] On
> Behalf Of st?phane bryant
> Sent: Wednesday, August 31, 2011 10:46 AM
> To: p2psip@ietf.org
> Subject: [P2PSIP] structure definition issue in draft-zong-p2psip-drr-
> 00
> 
> Hello,
> 
> In draft-zong-p2psip-drr-00, section 6.2.2. Extensive Routing Mode,
> the ExtensiveRoutingModeOption structure is defined thus:
> 
>  struct {
>            RouteMode               routemode;
>            OverlayLink             transport;
>            IpAddressPort          ipaddressport;
>            Destination             destination<1..2>;
>    } ExtensiveRoutingModeOption;
> 
> however the <x..y> convention for variable length
> fields is designed to represent the length in bytes
> of the field, the range allowing to also derive
> the size of the length field. <1..2> does not work.
> My take is that it should be <0...2^16>.
> 
> I am currently adding new parsing capabilities to
> the wireshark dissector: as soon as this is defined
> i will add capabilities for this draft.
> 
> regards
> stephane bryant
> _______________________________________________
> P2PSIP mailing list
> P2PSIP@ietf.org
> https://www.ietf.org/mailman/listinfo/p2psip


From michaelc@IDSSOFTWARE.COM  Thu Sep  1 19:52:04 2011
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From: "Michael Chen" <michaelc@idssoftware.com>
To: p2psip@ietf.org
Date: Thu, 01 Sep 2011 19:53:36 -0700
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Subject: [P2PSIP] Base draft 9.7.4.1 title does not match its content
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<html><body><span style=3D"font-family:Courier New; color:#000000; font-siz=
e:10pt;"><div>Hi,<br><br>I surprise this has slipped by everyone. The title=
 of the base draft section 9.7.4.1. is<br><br> &nbsp;"9.7.4.1. &nbsp;Updati=
ng <u><b>neighbor</b></u> table"<br><br>Its first sentence says,<br><br> &n=
bsp;"A peer MUST periodically send an Update request to every peer in its<b=
r> &nbsp; <u><b>Connection</b></u> Table."</div><div><br></div><div>Which o=
ne is the intended collection of peer for periodic update? The first bullet=
 of 9.7.4 seems to confirm that it should be "neighbor" table.<br></div><di=
v><br></div><div>Thanks</div><div><br></div><div>Michael Chen</div><div><br=
></div><div><br></div></span></body></html>

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From michaelc@IDSSOFTWARE.COM  Sat Sep  3 20:51:01 2011
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Subject: [P2PSIP] Question about base draft 9.7.4.4 Detecting partitioning
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Hi,=0A=0ABase draft section "9.7.4.4. Detecting partitioning" says,=0A=0A  =
"P should then send a Ping for its own Node-ID routed through B."=0A=0ASay =
you have a ring looks like this: ...-> X -> B -> Y -> P -> Z ->...=0Awhere =
B is P's bootstrap node.  If the overlay is healthy, wouldn't a=0APing to P=
's own Node-ID sent to B be routed back to P itself?  In that=0Acase, P wil=
l not be able to discover its possible new successor Z.=0A=0AAny thoughts? =
=0A=0A--Michael=0A

From michaelc@IDSSOFTWARE.COM  Sat Sep  3 22:07:56 2011
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Subject: [P2PSIP] Base draft 9.5 bullet 2 needs to be revised
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Hi,=0A=0ABase draft section 5.5.1 second paragraph states:=0A=0A  "As descr=
ibed in Section 5.1, an Attach may be routed to either a=0A   Node-ID or to=
 a Resource-ID.  An Attach routed to a specific Node-ID=0A   will fail if t=
hat node is not reached."=0A=0AThe second sentence above invalidates sectio=
n 9.5 bullet 2, which says=0A=0A  "2.  JP SHOULD send an Attach request to =
the admitting peer (AP) for=0A       Node-ID n.  The "send_update" flag sho=
uld be used to acquire the=0A       routing table for AP."=0A=0ASince JP's =
Node-ID is not yet valid, AP should fail the request as=0Ainstructed by 5.5=
.1.=0A=0AI believe this should be changed to (similar to how to discover=0A=
successor in later paragraph):=0A=0A  "2.  JP SHOULD send an Attach request=
 to the admitting peer (AP) for=0A       Resource-ID n+1.  The "send_update=
" flag should be used to=0A       acquire the routing table for AP."=0A=0AT=
hanks=0A=0A--Michael=0A=0A=0A

From stephane.ml.bryant@gmail.com  Sun Sep  4 23:21:25 2011
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Subject: [P2PSIP] typos in draft-ietf-p2psip-diagnostics-06
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Hello,

In draft-ietf-p2psip-diagnostics-06, the structures
DiagnosticsRequest and DiagnosticsResponse (Diagnostics
with an s) are at times referred to as DiagnosticRequest
and DiagnosticResponse (no s. In particular in the
definition of the PathTrackReq and PathTrackAns structures).

regards
stephane bryant

From haibin.song@huawei.com  Mon Sep  5 00:17:57 2011
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Date: Mon, 05 Sep 2011 07:18:55 +0000
From: Songhaibin <haibin.song@huawei.com>
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Thank you very much, Stephane. The authors will correct them in the next re=
vision.

BR,
-Haibin

> -----Original Message-----
> From: p2psip-bounces@ietf.org [mailto:p2psip-bounces@ietf.org] On Behalf =
Of
> st=E9phane bryant
> Sent: Monday, September 05, 2011 2:23 PM
> To: p2psip@ietf.org
> Subject: [P2PSIP] typos in draft-ietf-p2psip-diagnostics-06
>=20
> Hello,
>=20
> In draft-ietf-p2psip-diagnostics-06, the structures
> DiagnosticsRequest and DiagnosticsResponse (Diagnostics
> with an s) are at times referred to as DiagnosticRequest
> and DiagnosticResponse (no s. In particular in the
> definition of the PathTrackReq and PathTrackAns structures).
>=20
> regards
> stephane bryant
> _______________________________________________
> P2PSIP mailing list
> P2PSIP@ietf.org
> https://www.ietf.org/mailman/listinfo/p2psip

From Michael.Scharf@alcatel-lucent.com  Sun Sep  4 05:48:36 2011
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Subject: [P2PSIP] tsv-dir review of draft-ietf-p2psip-base-18
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Hi,

I've reviewed this document as part of the transport area directorate's
ongoing effort to review key IETF documents. These comments were written
primarily for the transport area directors, but are copied to the
document's authors for their information and to allow them to address
any issues raised. The authors should consider this review together with
any other last-call comments they receive. Please always CC
tsv-dir@ietf.org if you reply to or forward this review.=20


This draft is on the right track but has open issues, described in the
review. Given the length and complexity of the document, I only comment
on transport issues. However, for what it is worth, my personal
impression is that other sections not addressed by this review seem to
be somehow experimental.

My main concern is the Forwarding and Link Management Layer. It is
designed to run over different transports, most notably TCP and UDP.
Having such a generic transport including built-in NAT traversal support
is very valuable. But I find parts of the specification confusing, as
explained below. Also, building a simple, generic transport mechanism on
top of UDP and TCP (or, DTLS and TLS) is actually something that is
hardly specific to RELOAD.


Main concerns:

Section 1.2.5: "This layer also utilizes a framing header to encapsulate
messages as they are forwarding along each hop.  This header aids
reliability congestion control, flow control, etc.  It has meaning only
in the context of that individual link." (BTW: Note that a comma is
missing after reliability.)

=3D> This already well explains my concern that is further detailed =
below:
If TCP (or SCTP) transport is used, reliability, congestion control,
flow control are already solved by the transport protocol. Does the
Forwarding and Link Management Layer duplicate these functions then?
This and potential interactions are not well described in the document.


Section 5.2.1: "Because messages may be lost in transit through the
overlay, RELOAD incorporates an end-to-end reliability mechanism.  When
an originating node transmits a request it MUST set a 3 second timer. If
a response has not been received when the timer fires, the request is
retransmitted with the same transaction identifier. The request MAY be
retransmitted up to 4 times (for a total of 5 messages).  After the
timer for the fifth transmission fires, the message SHALL be considered
to have failed."

=3D> It may make sense to exponentially backoff this timer. Or, at least
it would be useful to explain why not to backoff. 3 sec is the initial
retransmission timeout duration of older TCP stacks, i. e., in worst
case scenarios even the transfer over a single overlap hop can result in
3 sec delay.


Section 5.5.1.6: "Highest priority is assigned to protocols that offer
well-understood congestion and flow control without head of line
blocking. For example, SCTP without message ordering, DCCP, or those
protocols encapsulated using UDP. [...] Second highest priority is
assigned to protocols that offer well-understood congestion and flow
control but have head of line blocking such as TCP."

=3D> While SCTP may be better suited for signaling applications than =
TCP,
I wonder why DCCP should have a high priority than TCP. AFAIK, DCCP
offers no reliability and is not build for signaling transport. Any
application-level retransmission mechanism on top of DCCP is likely to
be slower than transport protocol mechanism offered by SCTP and TCP.
Maybe I miss something?

(As a side note: Research results have shown that head-of-line blocking
only has a significant impact for high packet loss rates (> 1%). In that
case, the link is seriously congested and congestion control will
significantly throttle the data transfer. Head-of-line blocking might
then not be the most important delay component. Personally, I don't
think that avoiding head-of-line blocking is the most important
advantage of SCTP over TCP, and thus I don't think that it has to be
stressed in that section. But that is my personal view and it might not
be shared by the whole research community.)


Section 5.6: "The Framing Header (FH) is used to frame messages and
provide timing when used on a reliable stream-based transport protocol.
Simple Reliability (SR) makes use of the FH to provide congestion
control and semi-reliability when using unreliable message-oriented
transport protocols."=20

=3D> In this and in followup sections, the operation on top of a =
reliable
congestion-controlled protocol (TCP, possibly SCTP in future) should be
better separated from datagram transport (UDP). The required functions
in the Overlay Link Layer seem to be quite different.=20


Sections 5.6.1.1, Section 5.6.1.3, Section 5.6.1.4

=3D> These cases are apparently not covered by this spec. Move to
appendix?


Section 5.6.2: "The same header is used for both reliable and unreliable
transports for simplicity of implementation."

=3D> True, one can do that. But as both endpoints anyway have to agree =
to
on a transport, it would not be overly complex to use headers that are
better aligned with the features of the underlying transport, right? I.
e., one could use a more complex header if UDP transport is needed.
RELOAD has a similar flexibility concerning data structures in many
other parts of the protocol.


Section 5.6.2: "When the receiver receives a message, it SHOULD
immediately send an ACK message.  The receiver MUST keep track of the 32
most recent sequence numbers received on this association in order to
generate the appropriate ack."

=3D> Maybe I am completely lost here: Why is this MUST needed for =
TCP/TLS
links? In that case, messages arrive in order on a link, i. e., acking
the most recent one is sufficient. All previous ones must already have
been received.


Section 5.6.2: " received  A bitmask indicating if each of the previous
32 sequence numbers before this packet has been among the 32 packets
most recently received on this connection."

=3D> I had a hard time in understanding this sentence, and it
implications. I guess that this bitmask doesn't ensure full reliability,
at least in corner cases (e. g. 33 messages being lost in sequence). In
other words, this is a partial reliability scheme. I wonder why not to
use a cumulative ack with a bitmask acking out-of-order data, similar to
TCP's SACK. Also, the requirement of only acking the "32 packets most
recently received" may have undesired effects (e. g., what happens in
case of packet duplication?). In fact, instead of this customized
scheme, running a small subset of TCP's mechanisms in the user space
might just be fine, and I don't understand why this would be more
complex than implementing this scheme.


Section 5.6.2: "The received field bits in the ACK provide a high degree
of redundancy so that the sender can figure out which packets the
receiver has received and can then estimate packet loss rates. If the
sender also keeps track of the time at which recent sequence numbers
have been sent, the RTT can be estimated."

=3D> There is a lot of redunancy, indeed. But it would help to have a
short paragraph that explains how the sender indeed estimates the packet
loss rate. And note that the wording is slightly misleading: IMHO it is
not the *redundancy* that enables the sender to estimate the packet loss
rate. Other acknowlegdement schemes with less redundancy would work as
well.


Section 5.6.3.1: "In general, senders MAY implement any rate control
scheme of their choice, provided that it is REQUIRED to be no more
aggressive then TFRC[RFC5348]. The following section describes a simple,
inefficient scheme that complies with this requirement."

=3D> Maybe I again miss something, but as far as I can see the following
sections doesn't fully explain how TFRC is implemented here. For
example, TFRC requires information about the segment size; this is not
considered here. In general, it is not clear to me why a simple baseline
implementation does not follow the TFRC protocol (RFC 5348) more closely
(or a light-weight user-space TCP-like protocol).


Section 5.6.3.1.1: "In each retransmission, the sequence number is
incremented."

=3D> It would be worth to spend a sentence about the rational and
consequences of this. For instance, whether this implies that a receiver
may process the original message and the retransmission twice.


Section 5.6.3.1.1: "Implementations that use a dynamic estimate to
compute the RTO MUST use the algorithm described in RFC 6298[RFC6298],
with the exception that the value of RTO SHOULD NOT be rounded up to the
nearest second but instead rounded up to the nearest millisecond."

=3D> This sentence doesn't cite RFC 6298 correctly. RFC 6298 doesn't
mandate to round up to the nearest second. It mandates a minimum RTO of
1 second. And I strongly suggest to use such a minimum RTO as well
(maybe 500ms as minimum would be OK as well).


Section 5.6.3.1.1: "Once an ACK has been received for a message, the
next message can be sent, but the peer SHOULD ensure that there is at
least 10 ms between sending any two messages.  The only time a value
less than 10 ms can be used is when it is known that all nodes are on a
network that can support retransmissions faster than 10 ms with no
congestion issues."

=3D> This last requirement cannot be achieved in practice, IMHO. It is
impossible to know in advance congestion situations unless the overlay
operates in a fully controlled environment without any risk of link
failures etc. Thus, this paragraph effectively seems to limit the peak
load per link to 100 messages/s. I could imagine that this is not
sufficient in very large overlays. Would it make sense to put this
parameter in the configuration file instead?


Section 5.6.5: "Because the TCP layer's application-level timeout is too
slow to be useful for overlay routing, the Overlay Link implementation
MUST use the framing header to measure the RTT of the connection and
calculate an RTO as specified in Section 2 of [RFC6298]. The resulting
RTO is not used for retransmissions, but as a timeout to indicate when
the link SHOULD be removed from the routing table."

=3D> Again, instead of such implicit statements, I somehow miss in the
document an explicit statement that explains that relability, congestion
control, flow control etc. does not have to be provided by the overlay
link protocol when running over TCP.


Editorial issues:

Section 5.3.2.1: "Because the sequence number may in principle wrap,
greater than or less than are interpreted by modulo arithmetic as in
TCP."

=3D> Maybe better "... as sequence numbers in TCP"?


Section 5.5.1: "A node sends an Attach request when it wishes to
establish a direct TCP or UDP connection to another node for the purpose
of sending RELOAD messages."

=3D> As correctly noted later in Section 5.6, UDP does not properly have
"connections". The rationale for still using that term should be moved
to Section 5.5.1.


Section 5.5.1.8 and following sections: "An agent MUST skip the
verification procedures in Section 5.1 and 6.1 of ICE."

=3D> Less confusing would be "... of the ICE specification".


All Sections: Please ensure that concepts and acronyms are explained
when they are first used.


I apolologize if I should have missed something.

Michael

From haibin.song@huawei.com  Sun Sep  4 23:33:11 2011
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From: Songhaibin <haibin.song@huawei.com>
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Thank you very much, Stephane. The authors will correct them in the next re=
vision.

BR,
-Haibin

> -----Original Message-----
> From: p2psip-bounces@ietf.org [mailto:p2psip-bounces@ietf.org] On Behalf =
Of
> st=E9phane bryant
> Sent: Monday, September 05, 2011 2:23 PM
> To: p2psip@ietf.org
> Subject: [P2PSIP] typos in draft-ietf-p2psip-diagnostics-06
>=20
> Hello,
>=20
> In draft-ietf-p2psip-diagnostics-06, the structures
> DiagnosticsRequest and DiagnosticsResponse (Diagnostics
> with an s) are at times referred to as DiagnosticRequest
> and DiagnosticResponse (no s. In particular in the
> definition of the PathTrackReq and PathTrackAns structures).
>=20
> regards
> stephane bryant
> _______________________________________________
> P2PSIP mailing list
> P2PSIP@ietf.org
> https://www.ietf.org/mailman/listinfo/p2psip

From bbl@lowekamp.net  Mon Sep  5 12:03:28 2011
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Thanks for the review.  Responses inline

Bruce

On Sun, Sep 4, 2011 at 8:50 AM, SCHARF, Michael
<Michael.Scharf@alcatel-lucent.com> wrote:
> Hi,
>
> I've reviewed this document as part of the transport area directorate's
> ongoing effort to review key IETF documents. These comments were written
> primarily for the transport area directors, but are copied to the
> document's authors for their information and to allow them to address
> any issues raised. The authors should consider this review together with
> any other last-call comments they receive. Please always CC
> tsv-dir@ietf.org if you reply to or forward this review.
>
>
> This draft is on the right track but has open issues, described in the
> review. Given the length and complexity of the document, I only comment
> on transport issues. However, for what it is worth, my personal
> impression is that other sections not addressed by this review seem to
> be somehow experimental.
>
> My main concern is the Forwarding and Link Management Layer. It is
> designed to run over different transports, most notably TCP and UDP.
> Having such a generic transport including built-in NAT traversal support
> is very valuable. But I find parts of the specification confusing, as
> explained below. Also, building a simple, generic transport mechanism on
> top of UDP and TCP (or, DTLS and TLS) is actually something that is
> hardly specific to RELOAD.

Totally agree.  One of the reasons we left room for extension points
for new transports is in the hope that a more general solution will
become mature enough to be dropped in at a later date.   In line with
that idea, we tried to go for overly simple to avoid spending too much
time specifying complexity that we hope would be replaced with a
better solution.

>
>
> Main concerns:
>
> Section 1.2.5: "This layer also utilizes a framing header to encapsulate
> messages as they are forwarding along each hop. =C2=A0This header aids
> reliability congestion control, flow control, etc. =C2=A0It has meaning o=
nly
> in the context of that individual link." (BTW: Note that a comma is
> missing after reliability.)
>
> =3D> This already well explains my concern that is further detailed below=
:
> If TCP (or SCTP) transport is used, reliability, congestion control,
> flow control are already solved by the transport protocol. Does the
> Forwarding and Link Management Layer duplicate these functions then?
> This and potential interactions are not well described in the document.

Within the role of the overlay network, TCP/SCTP are used as
link-layer protocols for hop-by-hop connections (part of forwarding
and link management, as you observe).  However, e2e control is still
needed and provided by the Message Transport layer.  While we may get
some benefit from TCP/SCTP over the individual hops, they don't
address the end-to-end transport issues.

>
>
> Section 5.2.1: "Because messages may be lost in transit through the
> overlay, RELOAD incorporates an end-to-end reliability mechanism. =C2=A0W=
hen
> an originating node transmits a request it MUST set a 3 second timer. If
> a response has not been received when the timer fires, the request is
> retransmitted with the same transaction identifier. The request MAY be
> retransmitted up to 4 times (for a total of 5 messages). =C2=A0After the
> timer for the fifth transmission fires, the message SHALL be considered
> to have failed."
>
> =3D> It may make sense to exponentially backoff this timer. Or, at least
> it would be useful to explain why not to backoff. 3 sec is the initial
> retransmission timeout duration of older TCP stacks, i. e., in worst
> case scenarios even the transfer over a single overlap hop can result in
> 3 sec delay.

I think we had a discussion about this timer at some point.  I would
hope we don't need to worry about such ancient TCP stacks, but a more
realistic problem is the RTT for the multi-hop path across a large
overlay exceeding 3 seconds.  I admit I thought we had turned this
into a configurable parameter, at least.

There might be an argument that this should actually be adaptive,
applying something like 6298's RTO estimator across all e2e RTTs on
the overlay network and using that as an initial timeout for each new
message.  However, given the high variance in RTTs, it's not clear the
effect this would have on performance, and it would add a lot of
complexity to the implementation.  Given that the underlying
hop-by-hop transport has congestion control already, we decided to
leave it as a simple constant as network stability shouldn't be a
factor here.

>
>
> Section 5.5.1.6: "Highest priority is assigned to protocols that offer
> well-understood congestion and flow control without head of line
> blocking. For example, SCTP without message ordering, DCCP, or those
> protocols encapsulated using UDP. [...] Second highest priority is
> assigned to protocols that offer well-understood congestion and flow
> control but have head of line blocking such as TCP."
>
> =3D> While SCTP may be better suited for signaling applications than TCP,
> I wonder why DCCP should have a high priority than TCP. AFAIK, DCCP
> offers no reliability and is not build for signaling transport. Any
> application-level retransmission mechanism on top of DCCP is likely to
> be slower than transport protocol mechanism offered by SCTP and TCP.
> Maybe I miss something?

We don't care about hop-by-hop reliability here---so a datagram-based
protocol with congestion control seems ideal.

>
> (As a side note: Research results have shown that head-of-line blocking
> only has a significant impact for high packet loss rates (> 1%). In that
> case, the link is seriously congested and congestion control will
> significantly throttle the data transfer. Head-of-line blocking might
> then not be the most important delay component. Personally, I don't
> think that avoiding head-of-line blocking is the most important
> advantage of SCTP over TCP, and thus I don't think that it has to be
> stressed in that section. But that is my personal view and it might not
> be shared by the whole research community.)

Without being completely sure what results you're referring to, I'm
going to assume that the analysis is whether HoL blocking affects the
overall throughput of a TCP link, which I agree it shouldn't given
modern SACK/retransmission etc.  However, as we're using TCP here as
the hop-by-hop link layer underlying a higher-level e2e protocol in
the overlay network, the overall throughput of the hop isn't
necessarily as important as eliminating the latency imposed by HoL
blocking, which is there whenever there's a loss that requires
retransmission and a delay in delivering subsequent messages to the
application layer, and would greatly increase the e2e latency (and
variance in latency) of the messages.


>
>
> Section 5.6: "The Framing Header (FH) is used to frame messages and
> provide timing when used on a reliable stream-based transport protocol.
> Simple Reliability (SR) makes use of the FH to provide congestion
> control and semi-reliability when using unreliable message-oriented
> transport protocols."
>
> =3D> In this and in followup sections, the operation on top of a reliable
> congestion-controlled protocol (TCP, possibly SCTP in future) should be
> better separated from datagram transport (UDP). The required functions
> in the Overlay Link Layer seem to be quite different.
>

why?  e2e issues seem the same---even between consecutive TCP hops,
the peers may experience congestion on the outgoing link and be forced
to drop messages.

>
> Sections 5.6.1.1, Section 5.6.1.3, Section 5.6.1.4
>
> =3D> These cases are apparently not covered by this spec. Move to
> appendix?
>

We agreed a long time ago to leave things in 5.6.1.x in as callouts to
future transports that were hoped to be added.  I have no particular
opinion whether it should be there or in an appendix.

>
> Section 5.6.2: "The same header is used for both reliable and unreliable
> transports for simplicity of implementation."
>
> =3D> True, one can do that. But as both endpoints anyway have to agree to
> on a transport, it would not be overly complex to use headers that are
> better aligned with the features of the underlying transport, right? I.
> e., one could use a more complex header if UDP transport is needed.
> RELOAD has a similar flexibility concerning data structures in many
> other parts of the protocol.
>
>
> Section 5.6.2: "When the receiver receives a message, it SHOULD
> immediately send an ACK message. =C2=A0The receiver MUST keep track of th=
e 32
> most recent sequence numbers received on this association in order to
> generate the appropriate ack."
>
> =3D> Maybe I am completely lost here: Why is this MUST needed for TCP/TLS
> links? In that case, messages arrive in order on a link, i. e., acking
> the most recent one is sufficient. All previous ones must already have
> been received.

Given the already significant length and complexity of the draft and
protocol, I think "simplicity of implementation" is a significant
concern.  I don't see a negative impact here.

>
>
> Section 5.6.2: " received =C2=A0A bitmask indicating if each of the previ=
ous
> 32 sequence numbers before this packet has been among the 32 packets
> most recently received on this connection."
>
> =3D> I had a hard time in understanding this sentence, and it
> implications. I guess that this bitmask doesn't ensure full reliability,
> at least in corner cases (e. g. 33 messages being lost in sequence). In
> other words, this is a partial reliability scheme. I wonder why not to
> use a cumulative ack with a bitmask acking out-of-order data, similar to
> TCP's SACK. Also, the requirement of only acking the "32 packets most
> recently received" may have undesired effects (e. g., what happens in
> case of packet duplication?). In fact, instead of this customized
> scheme, running a small subset of TCP's mechanisms in the user space
> might just be fine, and I don't understand why this would be more
> complex than implementing this scheme.
>

below

>
> Section 5.6.2: "The received field bits in the ACK provide a high degree
> of redundancy so that the sender can figure out which packets the
> receiver has received and can then estimate packet loss rates. If the
> sender also keeps track of the time at which recent sequence numbers
> have been sent, the RTT can be estimated."
>
> =3D> There is a lot of redunancy, indeed. But it would help to have a
> short paragraph that explains how the sender indeed estimates the packet
> loss rate. And note that the wording is slightly misleading: IMHO it is
> not the *redundancy* that enables the sender to estimate the packet loss
> rate. Other acknowlegdement schemes with less redundancy would work as
> well.
>

The discussion about what protocol to use here was one of the most
controversial issues we had to address (both among the authors and
within the wg).  While I personally wanted to include a more tcp-like
layer here, I was outvoted on the issue on the ground that what we
have now is simpler (at least to those not sufficiently familiar with
TCP), safe for network stability, and has sufficient performance for
most overlays.  We hope that more work will come out of TSV on
appropriate user-layer/tunneled over UDP protocols that can be used to
replace what's here in the future, but we decided to leave such work
out of this draft as it was already sufficiently complexed.  (and at
the time I had hoped to spend time working with folks in TSV on such
issues, but have had to curtail my IETF time recently as my current
position doesn't support such work)

>
> Section 5.6.3.1: "In general, senders MAY implement any rate control
> scheme of their choice, provided that it is REQUIRED to be no more
> aggressive then TFRC[RFC5348]. The following section describes a simple,
> inefficient scheme that complies with this requirement."
>
> =3D> Maybe I again miss something, but as far as I can see the following
> sections doesn't fully explain how TFRC is implemented here. For
> example, TFRC requires information about the segment size; this is not
> considered here. In general, it is not clear to me why a simple baseline
> implementation does not follow the TFRC protocol (RFC 5348) more closely
> (or a light-weight user-space TCP-like protocol).
>

again, simplicity won out.  We believe what's there is a simpler
protocol (the simplest we could think of) that follows the requirement
of being less aggressive than TFRC.

>
> Section 5.6.3.1.1: "In each retransmission, the sequence number is
> incremented."
>
> =3D> It would be worth to spend a sentence about the rational and
> consequences of this. For instance, whether this implies that a receiver
> may process the original message and the retransmission twice.
>

Happy to spend a sentence on this, but just to be clear, this is
designed to allow the world's dumbest stop-and-wait protocol.
Specifically, the goal of 5.6.3.1.1 is to specify a ludicrously
simple, totally safe algorithm that is sufficient for most envisioned
deployments of P2PSIP for implementors who don't want to spend the
time implementing TFRC.  I would hope most do spend the time, but it
seemed unnecessary to specify how TFRC works in this document.


As far as the separate sequence numbers, they can be used because
there's no issue with processing here as with TCP sequence numbers
(this is the link-level hop-by-hop protocol), and if you want to do
something fancier than stop-and-wait, it avoids ambiguity in
calculating RTO (i.e. you don't need Karn's algorithm).  Essentially,
the receiver should is designed to be compliant with a much more
intelligent sender than the stop-and-wait specified in this section,
similar to the simplest TCP receivers.

>
> Section 5.6.3.1.1: "Implementations that use a dynamic estimate to
> compute the RTO MUST use the algorithm described in RFC 6298[RFC6298],
> with the exception that the value of RTO SHOULD NOT be rounded up to the
> nearest second but instead rounded up to the nearest millisecond."
>
> =3D> This sentence doesn't cite RFC 6298 correctly. RFC 6298 doesn't
> mandate to round up to the nearest second. It mandates a minimum RTO of
> 1 second. And I strongly suggest to use such a minimum RTO as well
> (maybe 500ms as minimum would be OK as well).
>

I think you refer to where 6298 says:

         Until a round-trip time (RTT) measurement has been made for a
         segment sent between the sender and receiver, the sender SHOULD
         set RTO <- 1 second, though the "backing off" on repeated
         retransmission discussed in (5.5) still applies.

there's no need for that as ICE has already completed, so we have an
RTT measurement.

>
> Section 5.6.3.1.1: "Once an ACK has been received for a message, the
> next message can be sent, but the peer SHOULD ensure that there is at
> least 10 ms between sending any two messages. =C2=A0The only time a value
> less than 10 ms can be used is when it is known that all nodes are on a
> network that can support retransmissions faster than 10 ms with no
> congestion issues."
>
> =3D> This last requirement cannot be achieved in practice, IMHO. It is
> impossible to know in advance congestion situations unless the overlay
> operates in a fully controlled environment without any risk of link
> failures etc.

The operating environment referred to would be a LAN or similar.  (more bel=
ow)


> Thus, this paragraph effectively seems to limit the peak
> load per link to 100 messages/s. I could imagine that this is not
> sufficient in very large overlays.

Agreed.  This protocol isn't intended to be operated in such an
environment (would need TFRC, etc for that), but for many of the
envisioned deployments of P2PSIP, it should be fine. (or some future
emerging option)

> Would it make sense to put this
> parameter in the configuration file instead?

I guess it's a valid point that you can't determine if you're in a
network where you can do better without a configuration option.  I
hate to spend the time overthinking this option, given its deliberate
limited applicability.  Maybe we should just remove that.

>
>
> Section 5.6.5: "Because the TCP layer's application-level timeout is too
> slow to be useful for overlay routing, the Overlay Link implementation
> MUST use the framing header to measure the RTT of the connection and
> calculate an RTO as specified in Section 2 of [RFC6298]. The resulting
> RTO is not used for retransmissions, but as a timeout to indicate when
> the link SHOULD be removed from the routing table."
>
> =3D> Again, instead of such implicit statements, I somehow miss in the
> document an explicit statement that explains that relability, congestion
> control, flow control etc. does not have to be provided by the overlay
> link protocol when running over TCP.

Can add something to the beginning of 5.6 to this effect.

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Sorry to follow up on my on post.  Meant to change a comment before I sent.=
..

On Mon, Sep 5, 2011 at 3:05 PM, Bruce Lowekamp <bbl@lowekamp.net> wrote:
> On Sun, Sep 4, 2011 at 8:50 AM, SCHARF, Michael
> <Michael.Scharf@alcatel-lucent.com> wrote:
>>
>> Section 5.6: "The Framing Header (FH) is used to frame messages and
>> provide timing when used on a reliable stream-based transport protocol.
>> Simple Reliability (SR) makes use of the FH to provide congestion
>> control and semi-reliability when using unreliable message-oriented
>> transport protocols."
>>
>> =3D> In this and in followup sections, the operation on top of a reliabl=
e
>> congestion-controlled protocol (TCP, possibly SCTP in future) should be
>> better separated from datagram transport (UDP). The required functions
>> in the Overlay Link Layer seem to be quite different.
>>
>
> why? =C2=A0e2e issues seem the same---even between consecutive TCP hops,
> the peers may experience congestion on the outgoing link and be forced
> to drop messages.

Ignore that comment, please.

Functionally, since TCP doesn't provide fast enough timeouts to be
useful as an overlay link protocol, an application-layer protocol is
required regardless of its underlying guarantees.
Implementation-wise, I think simplicity is the overriding concern.

Bruce

From Michael.Scharf@alcatel-lucent.com  Tue Sep  6 10:18:54 2011
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Some remarks on selected aspects inline.=20

> > This draft is on the right track but has open issues,=20
> described in the=20
> > review. Given the length and complexity of the document, I only=20
> > comment on transport issues. However, for what it is worth, my=20
> > personal impression is that other sections not addressed by this=20
> > review seem to be somehow experimental.
> >
> > My main concern is the Forwarding and Link Management Layer. It is=20
> > designed to run over different transports, most notably TCP and UDP.
> > Having such a generic transport including built-in NAT traversal=20
> > support is very valuable. But I find parts of the specification=20
> > confusing, as explained below. Also, building a simple, generic=20
> > transport mechanism on top of UDP and TCP (or, DTLS and TLS) is=20
> > actually something that is hardly specific to RELOAD.
>=20
> Totally agree.  One of the reasons we left room for extension=20
> points for new transports is in the hope that a more general=20
> solution will
> become mature enough to be dropped in at a later date.   In line with
> that idea, we tried to go for overly simple to avoid spending=20
> too much time specifying complexity that we hope would be=20
> replaced with a better solution.

Then we should really avoid that someone read this spec and thinks that
it describes a good one-fits-all solution.


> > Section 5.6.3.1: "In general, senders MAY implement any=20
> rate control=20
> > scheme of their choice, provided that it is REQUIRED to be no more=20
> > aggressive then TFRC[RFC5348]. The following section describes a=20
> > simple, inefficient scheme that complies with this requirement."
> >
> > =3D> Maybe I again miss something, but as far as I can see=20
> the following=20
> > sections doesn't fully explain how TFRC is implemented here. For=20
> > example, TFRC requires information about the segment size;=20
> this is not=20
> > considered here. In general, it is not clear to me why a simple=20
> > baseline implementation does not follow the TFRC protocol=20
> (RFC 5348)=20
> > more closely (or a light-weight user-space TCP-like protocol).
> >
>=20
> again, simplicity won out.  We believe what's there is a=20
> simpler protocol (the simplest we could think of) that=20
> follows the requirement of being less aggressive than TFRC.

Just to repeat what I said: The draft doesn't explain how the
requirement of being less aggressive than TFRC is indeed achieved.


> > Section 5.6.3.1.1: "In each retransmission, the sequence number is=20
> > incremented."
> >
> > =3D> It would be worth to spend a sentence about the rational and=20
> > consequences of this. For instance, whether this implies that a=20
> > receiver may process the original message and the=20
> retransmission twice.
> >
>=20
> Happy to spend a sentence on this, but just to be clear, this=20
> is designed to allow the world's dumbest stop-and-wait protocol.

Well, this sentence prevents even the alternating bit protocol... In
fact, this solution is closer to a timestamp than a sequence number.


> Specifically, the goal of 5.6.3.1.1 is to specify a=20
> ludicrously simple, totally safe algorithm that is sufficient=20
> for most envisioned deployments of P2PSIP for implementors=20
> who don't want to spend the time implementing TFRC.  I would=20
> hope most do spend the time, but it seemed unnecessary to=20
> specify how TFRC works in this document.

AFAIK, section 5.6.3.1.1 is *not* save as it is, and it does *not*
guarantee that it is less aggressive than TFRC.

The specified protocol is probably save *if and only if* the sender
ensures that only very few messages are sent. Then, section 3.1.2. of
RFC 5405 applies ("Low Data-Volume Applications").=20

For instance, I guess that the protocol would indeed be OK if the spec
did mandate that only a single message MUST be outstanding on a link.
But, unless I miss something, at the moment there is only an upper limit
to 100 messages/sec. IMHO this is orders of magnitude above the low
data-volume applications discussed in RFC 5405.


> > Section 5.6.3.1.1: "Implementations that use a dynamic estimate to=20
> > compute the RTO MUST use the algorithm described in RFC=20
> 6298[RFC6298],=20
> > with the exception that the value of RTO SHOULD NOT be=20
> rounded up to=20
> > the nearest second but instead rounded up to the nearest=20
> millisecond."
> >
> > =3D> This sentence doesn't cite RFC 6298 correctly. RFC 6298 doesn't =

> > mandate to round up to the nearest second. It mandates a=20
> minimum RTO=20
> > of
> > 1 second. And I strongly suggest to use such a minimum RTO as well=20
> > (maybe 500ms as minimum would be OK as well).
> >
>=20
> I think you refer to where 6298 says:
>=20
>          Until a round-trip time (RTT) measurement has been made for a
>          segment sent between the sender and receiver, the=20
> sender SHOULD
>          set RTO <- 1 second, though the "backing off" on repeated
>          retransmission discussed in (5.5) still applies.
>=20
> there's no need for that as ICE has already completed, so we=20
> have an RTT measurement.

My understanding of this sentence is completely different. In my
opinion, it suggests not to use this part 6298:

"  (2.4) Whenever RTO is computed, if it is less than 1 second, then the
         RTO SHOULD be rounded up to 1 second."

In other words, Section 5.6.3.1.1 argues that the minumum RTO should be
1ms. IMHO, this value is orders of magnitude too small.

Maybe this is just a phrasing issue. But, again, I think that you should
use a minimum RTO of the order of 500ms independent of how you measure
the RTT. Otherwise, the protocol is very vulnerable to changes of the
path RTT etc.


Michael


From bbl@lowekamp.net  Tue Sep  6 19:20:37 2011
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On Tue, Sep 6, 2011 at 1:19 PM, SCHARF, Michael
<Michael.Scharf@alcatel-lucent.com> wrote:
> Some remarks on selected aspects inline.
>
>> > This draft is on the right track but has open issues,
>> described in the
>> > review. Given the length and complexity of the document, I only
>> > comment on transport issues. However, for what it is worth, my
>> > personal impression is that other sections not addressed by this
>> > review seem to be somehow experimental.
>> >
>> > My main concern is the Forwarding and Link Management Layer. It is
>> > designed to run over different transports, most notably TCP and UDP.
>> > Having such a generic transport including built-in NAT traversal
>> > support is very valuable. But I find parts of the specification
>> > confusing, as explained below. Also, building a simple, generic
>> > transport mechanism on top of UDP and TCP (or, DTLS and TLS) is
>> > actually something that is hardly specific to RELOAD.
>>
>> Totally agree. =C2=A0One of the reasons we left room for extension
>> points for new transports is in the hope that a more general
>> solution will
>> become mature enough to be dropped in at a later date. =C2=A0 In line wi=
th
>> that idea, we tried to go for overly simple to avoid spending
>> too much time specifying complexity that we hope would be
>> replaced with a better solution.
>
> Then we should really avoid that someone read this spec and thinks that
> it describes a good one-fits-all solution.
>

I think one-size-fits-all is a dangerous claim for anything.  I looked
for and was surprised not to find a disclaimer at the beginning of
5.6.3 (or at least 5.6.3.1.1).  And certainly we took this into
account in the design---for example in 5.5.1.6, DTLS/UDP with Simple
Reliability (as specified in the document) is the lowest priority
option.  But we can be a bit clearer in 5.6 as to the limitations of
what's there and the reasons.

>
>> > Section 5.6.3.1: "In general, senders MAY implement any
>> rate control
>> > scheme of their choice, provided that it is REQUIRED to be no more
>> > aggressive then TFRC[RFC5348]. The following section describes a
>> > simple, inefficient scheme that complies with this requirement."
>> >
>> > =3D> Maybe I again miss something, but as far as I can see
>> the following
>> > sections doesn't fully explain how TFRC is implemented here. For
>> > example, TFRC requires information about the segment size;
>> this is not
>> > considered here. In general, it is not clear to me why a simple
>> > baseline implementation does not follow the TFRC protocol
>> (RFC 5348)
>> > more closely (or a light-weight user-space TCP-like protocol).
>> >
>>
>> again, simplicity won out. =C2=A0We believe what's there is a
>> simpler protocol (the simplest we could think of) that
>> follows the requirement of being less aggressive than TFRC.
>
> Just to repeat what I said: The draft doesn't explain how the
> requirement of being less aggressive than TFRC is indeed achieved.
>

Regarding this and following discussion, maybe it would be better to
back up a level and make sure we're talking about the same things.
5.6.3.1.1, which is the only actual retransmission scheme defined in
the document says:

   The node MUST
   double the time to wait after each retransmission.
   ...
   Once an ACK has been received for a message, the next message can be
   sent, but the peer SHOULD ensure that there is at least 10 ms between
   sending any two messages.

so this is a stop and wait algorithm with exponential backoff on
retransmissions.  Furthermore,

   If all
   retransmissions for a message fail, then the sending node SHOULD
   close the connection routing the message.

So it actually abandons a link if retransmissions don't work, rather
than continuing to put traffic into it.

Is there any actual question that this is a safe (albeit
non-performant) algorithm?


>
>> > Section 5.6.3.1.1: "In each retransmission, the sequence number is
>> > incremented."
>> >
>> > =3D> It would be worth to spend a sentence about the rational and
>> > consequences of this. For instance, whether this implies that a
>> > receiver may process the original message and the
>> retransmission twice.
>> >
>>
>> Happy to spend a sentence on this, but just to be clear, this
>> is designed to allow the world's dumbest stop-and-wait protocol.
>
> Well, this sentence prevents even the alternating bit protocol... In
> fact, this solution is closer to a timestamp than a sequence number.
>

It's a sequence number, it's just not conflating the index of data in
a stream with the retransmission of fragments, as we are all used to
with TCP.  Being able to calculate RTTs from every ACK is a really
useful feature.

>
>> Specifically, the goal of 5.6.3.1.1 is to specify a
>> ludicrously simple, totally safe algorithm that is sufficient
>> for most envisioned deployments of P2PSIP for implementors
>> who don't want to spend the time implementing TFRC. =C2=A0I would
>> hope most do spend the time, but it seemed unnecessary to
>> specify how TFRC works in this document.
>
> AFAIK, section 5.6.3.1.1 is *not* save as it is, and it does *not*
> guarantee that it is less aggressive than TFRC.
>
> The specified protocol is probably save *if and only if* the sender
> ensures that only very few messages are sent. Then, section 3.1.2. of
> RFC 5405 applies ("Low Data-Volume Applications").
>
> For instance, I guess that the protocol would indeed be OK if the spec
> did mandate that only a single message MUST be outstanding on a link.
> But, unless I miss something, at the moment there is only an upper limit
> to 100 messages/sec. IMHO this is orders of magnitude above the low
> data-volume applications discussed in RFC 5405.
>
>
>> > Section 5.6.3.1.1: "Implementations that use a dynamic estimate to
>> > compute the RTO MUST use the algorithm described in RFC
>> 6298[RFC6298],
>> > with the exception that the value of RTO SHOULD NOT be
>> rounded up to
>> > the nearest second but instead rounded up to the nearest
>> millisecond."
>> >
>> > =3D> This sentence doesn't cite RFC 6298 correctly. RFC 6298 doesn't
>> > mandate to round up to the nearest second. It mandates a
>> minimum RTO
>> > of
>> > 1 second. And I strongly suggest to use such a minimum RTO as well
>> > (maybe 500ms as minimum would be OK as well).
>> >
>>
>> I think you refer to where 6298 says:
>>
>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Until a round-trip time (RTT) measurem=
ent has been made for a
>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0segment sent between the sender and re=
ceiver, the
>> sender SHOULD
>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0set RTO <- 1 second, though the "backi=
ng off" on repeated
>> =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0retransmission discussed in (5.5) stil=
l applies.
>>
>> there's no need for that as ICE has already completed, so we
>> have an RTT measurement.
>
> My understanding of this sentence is completely different. In my
> opinion, it suggests not to use this part 6298:
>
> " =C2=A0(2.4) Whenever RTO is computed, if it is less than 1 second, then=
 the
> =C2=A0 =C2=A0 =C2=A0 =C2=A0 RTO SHOULD be rounded up to 1 second."
>
> In other words, Section 5.6.3.1.1 argues that the minumum RTO should be
> 1ms. IMHO, this value is orders of magnitude too small.
>
> Maybe this is just a phrasing issue. But, again, I think that you should
> use a minimum RTO of the order of 500ms independent of how you measure
> the RTT. Otherwise, the protocol is very vulnerable to changes of the
> path RTT etc.
>
>
> Michael
>
>

From brian.rosen@neustar.biz  Wed Sep  7 05:41:43 2011
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
To: P2PSIP WG <p2psip@ietf.org>
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--_000_8CC519847A2A42DB84D1B3C828873BEEneustarbiz_
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For some reason, the mailer discarded this message.  Doesn't appear to have=
 hit the list any other way, so I'm forwarding it

Begin forwarded message:

From: "SCHARF, Michael" <Michael.Scharf@alcatel-lucent.com<mailto:Michael.S=
charf@alcatel-lucent.com>>
Subject: RE: tsv-dir review of draft-ietf-p2psip-base-18
Date: September 7, 2011 7:59:11 AM EDT
To: "Bruce Lowekamp" <bbl@lowekamp.net<mailto:bbl@lowekamp.net>>
Cc: <iesg@ietf.org<mailto:iesg@ietf.org>>, <tsv-dir@ietf.org<mailto:tsv-dir=
@ietf.org>>, <tsv-area@ietf.org<mailto:tsv-area@ietf.org>>, <p2psip@ietf.or=
g<mailto:p2psip@ietf.org>>, <draft-ietf-p2psip-base@tools.ietf.org<mailto:d=
raft-ietf-p2psip-base@tools.ietf.org>>


I'll just comment on one detail below, as it is indeed important

Section 5.6.3.1: "In general, senders MAY implement any
rate control
scheme of their choice, provided that it is REQUIRED to
be no more
aggressive then TFRC[RFC5348]. The following section describes a
simple, inefficient scheme that complies with this requirement."

=3D> Maybe I again miss something, but as far as I can see
the following
sections doesn't fully explain how TFRC is implemented here. For
example, TFRC requires information about the segment size;
this is not
considered here. In general, it is not clear to me why a simple
baseline implementation does not follow the TFRC protocol
(RFC 5348)
more closely (or a light-weight user-space TCP-like protocol).


again, simplicity won out.  We believe what's there is a simpler
protocol (the simplest we could think of) that follows the
requirement of being less aggressive than TFRC.

Just to repeat what I said: The draft doesn't explain how the
requirement of being less aggressive than TFRC is indeed achieved.


Regarding this and following discussion, maybe it would be
better to back up a level and make sure we're talking about
the same things.
5.6.3.1.1, which is the only actual retransmission scheme
defined in the document says:

  The node MUST
  double the time to wait after each retransmission.
  ...
  Once an ACK has been received for a message, the next
message can be
  sent, but the peer SHOULD ensure that there is at least 10
ms between
  sending any two messages.

so this is a stop and wait algorithm with exponential backoff
on retransmissions.  Furthermore,

  If all
  retransmissions for a message fail, then the sending node SHOULD
  close the connection routing the message.

So it actually abandons a link if retransmissions don't work,
rather than continuing to put traffic into it.

Is there any actual question that this is a safe (albeit
non-performant) algorithm?

I don't think that the current text in section 5.6.3.1.1 actually describes=
 a stop-and-wait protocol. In my reading, the sentence "Once an ACK has bee=
n received for a message, the next message can be sent" does not prevent th=
e sender from transmitting several messages in parallel (but well, I am not=
 a native speaker). To me, the text describes a windowing protocol that is =
allowed to send one message per 10ms, i. e., in summary 100 messages per se=
cond.

If this section is about a stop-and-wait protocol, then please mention that=
 and use appropriate normative language. For instance, stop-and-wait would =
result in something like "The next message MUST only be sent after the ACK =
of the previous message has been received (or the retransmission finally ti=
mes out)".

One reason why this section may be confusing is the "received" field, as al=
ready noted in my review. A stop-and-wait protocol doesn't need it; it only=
 makes sense if there is more than one message outstanding, i. e., if there=
 is a send window. If the intention is to specify a stop-and-wait protocol =
only, then I suggest to remove the "received" field from the spec. Anyway, =
other semantics for this field might make sense if a better transport mecha=
nism will be defined in future, as already noted in my review.

A stop-and-wait protocol with a single outstanding message (not too large, =
i. e., not significantly exceeding 3-5 KB) and adaptive RTO including back-=
off would be rather safe to use and sufficiently TCP friendly, according to=
 RFC 5405. A corresponding rephrasing of these sections might address most =
of my concerns regarding congestion control.

Of course, such a stop-and-wait doesn't make sense for any larger overlay, =
but apparently we agree on that.

Michael




--_000_8CC519847A2A42DB84D1B3C828873BEEneustarbiz_
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Content-Transfer-Encoding: quoted-printable

<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode:=
 space; -webkit-line-break: after-white-space; ">For some reason, the maile=
r discarded this message. &nbsp;Doesn't appear to have hit the list any oth=
er way, so I'm forwarding it<br><div><br><div>Begin forwarded message:</div=
><br class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div sty=
le=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: =
0px;"><span style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(=
127, 127, 127, 1.0);"><b>From: </b></span><span style=3D"font-family:'Helve=
tica'; font-size:medium;">"SCHARF, Michael" &lt;<a href=3D"mailto:Michael.S=
charf@alcatel-lucent.com">Michael.Scharf@alcatel-lucent.com</a>&gt;</span><=
/div><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; font-size:medium=
; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span><span style=3D"fo=
nt-family:'Helvetica'; font-size:medium;"><b>RE: tsv-dir review of draft-ie=
tf-p2psip-base-18</b><br></span></div><div style=3D"margin-top: 0px; margin=
-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style=3D"font-fam=
ily:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Date=
: </b></span><span style=3D"font-family:'Helvetica'; font-size:medium;">Sep=
tember 7, 2011 7:59:11 AM EDT<br></span></div><div style=3D"margin-top: 0px=
; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style=3D"=
font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);"=
><b>To: </b></span><span style=3D"font-family:'Helvetica'; font-size:medium=
;">"Bruce Lowekamp" &lt;<a href=3D"mailto:bbl@lowekamp.net">bbl@lowekamp.ne=
t</a>&gt;<br></span></div><div style=3D"margin-top: 0px; margin-right: 0px;=
 margin-bottom: 0px; margin-left: 0px;"><span style=3D"font-family:'Helveti=
ca'; font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Cc: </b></span><=
span style=3D"font-family:'Helvetica'; font-size:medium;">&lt;<a href=3D"ma=
ilto:iesg@ietf.org">iesg@ietf.org</a>&gt;, &lt;<a href=3D"mailto:tsv-dir@ie=
tf.org">tsv-dir@ietf.org</a>&gt;, &lt;<a href=3D"mailto:tsv-area@ietf.org">=
tsv-area@ietf.org</a>&gt;, &lt;<a href=3D"mailto:p2psip@ietf.org">p2psip@ie=
tf.org</a>&gt;, &lt;<a href=3D"mailto:draft-ietf-p2psip-base@tools.ietf.org=
">draft-ietf-p2psip-base@tools.ietf.org</a>&gt;<br></span></div><br><br>I'l=
l just comment on one detail below, as it is indeed important<br><br><block=
quote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite"><bl=
ockquote type=3D"cite">Section 5.6.3.1: "In general, senders MAY implement =
any<br></blockquote></blockquote></blockquote></blockquote><blockquote type=
=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite">rate control<=
br></blockquote></blockquote></blockquote><blockquote type=3D"cite"><blockq=
uote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite">sche=
me of their choice, provided that it is REQUIRED to <br></blockquote></bloc=
kquote></blockquote></blockquote><blockquote type=3D"cite">be no more <br><=
/blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote=
 type=3D"cite"><blockquote type=3D"cite">aggressive then TFRC[RFC5348]. The=
 following section describes a <br></blockquote></blockquote></blockquote><=
/blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote=
 type=3D"cite"><blockquote type=3D"cite">simple, inefficient scheme that co=
mplies with this requirement."<br></blockquote></blockquote></blockquote></=
blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote =
type=3D"cite"><blockquote type=3D"cite"><br></blockquote></blockquote></blo=
ckquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><b=
lockquote type=3D"cite"><blockquote type=3D"cite">=3D&gt; Maybe I again mis=
s something, but as far as I can see<br></blockquote></blockquote></blockqu=
ote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><block=
quote type=3D"cite">the following<br></blockquote></blockquote></blockquote=
><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"ci=
te"><blockquote type=3D"cite">sections doesn't fully explain how TFRC is im=
plemented here. For <br></blockquote></blockquote></blockquote></blockquote=
><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"ci=
te"><blockquote type=3D"cite">example, TFRC requires information about the =
segment size;<br></blockquote></blockquote></blockquote></blockquote><block=
quote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite">thi=
s is not<br></blockquote></blockquote></blockquote><blockquote type=3D"cite=
"><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"c=
ite">considered here. In general, it is not clear to me why a simple <br></=
blockquote></blockquote></blockquote></blockquote><blockquote type=3D"cite"=
><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"ci=
te">baseline implementation does not follow the TFRC protocol<br></blockquo=
te></blockquote></blockquote></blockquote><blockquote type=3D"cite"><blockq=
uote type=3D"cite"><blockquote type=3D"cite">(RFC 5348)<br></blockquote></b=
lockquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite">=
<blockquote type=3D"cite"><blockquote type=3D"cite">more closely (or a ligh=
t-weight user-space TCP-like protocol).<br></blockquote></blockquote></bloc=
kquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><bl=
ockquote type=3D"cite"><blockquote type=3D"cite"><br></blockquote></blockqu=
ote></blockquote></blockquote><blockquote type=3D"cite"><blockquote type=3D=
"cite"><blockquote type=3D"cite"><br></blockquote></blockquote></blockquote=
><blockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"ci=
te">again, simplicity won out. &nbsp;We believe what's there is a simpler <=
br></blockquote></blockquote></blockquote><blockquote type=3D"cite"><blockq=
uote type=3D"cite"><blockquote type=3D"cite">protocol (the simplest we coul=
d think of) that follows the <br></blockquote></blockquote></blockquote><bl=
ockquote type=3D"cite"><blockquote type=3D"cite"><blockquote type=3D"cite">=
requirement of being less aggressive than TFRC.<br></blockquote></blockquot=
e></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><br></bl=
ockquote></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite">J=
ust to repeat what I said: The draft doesn't explain how the <br></blockquo=
te></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite">require=
ment of being less aggressive than TFRC is indeed achieved.<br></blockquote=
></blockquote><blockquote type=3D"cite"><blockquote type=3D"cite"><br></blo=
ckquote></blockquote><blockquote type=3D"cite"><br></blockquote><blockquote=
 type=3D"cite">Regarding this and following discussion, maybe it would be <=
br></blockquote><blockquote type=3D"cite">better to back up a level and mak=
e sure we're talking about <br></blockquote><blockquote type=3D"cite">the s=
ame things.<br></blockquote><blockquote type=3D"cite">5.6.3.1.1, which is t=
he only actual retransmission scheme <br></blockquote><blockquote type=3D"c=
ite">defined in the document says:<br></blockquote><blockquote type=3D"cite=
"><br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;The node MUST<br>=
</blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;double the time to wait=
 after each retransmission.<br></blockquote><blockquote type=3D"cite"> &nbs=
p;&nbsp;...<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;Once an =
ACK has been received for a message, the next <br></blockquote><blockquote =
type=3D"cite">message can be<br></blockquote><blockquote type=3D"cite"> &nb=
sp;&nbsp;sent, but the peer SHOULD ensure that there is at least 10 <br></b=
lockquote><blockquote type=3D"cite">ms between<br></blockquote><blockquote =
type=3D"cite"> &nbsp;&nbsp;sending any two messages.<br></blockquote><block=
quote type=3D"cite"><br></blockquote><blockquote type=3D"cite">so this is a=
 stop and wait algorithm with exponential backoff <br></blockquote><blockqu=
ote type=3D"cite">on retransmissions. &nbsp;Furthermore,<br></blockquote><b=
lockquote type=3D"cite"><br></blockquote><blockquote type=3D"cite"> &nbsp;&=
nbsp;If all<br></blockquote><blockquote type=3D"cite"> &nbsp;&nbsp;retransm=
issions for a message fail, then the sending node SHOULD<br></blockquote><b=
lockquote type=3D"cite"> &nbsp;&nbsp;close the connection routing the messa=
ge.<br></blockquote><blockquote type=3D"cite"><br></blockquote><blockquote =
type=3D"cite">So it actually abandons a link if retransmissions don't work,=
 <br></blockquote><blockquote type=3D"cite">rather than continuing to put t=
raffic into it.<br></blockquote><blockquote type=3D"cite"><br></blockquote>=
<blockquote type=3D"cite">Is there any actual question that this is a safe =
(albeit<br></blockquote><blockquote type=3D"cite">non-performant) algorithm=
?<br></blockquote><br>I don't think that the current text in section 5.6.3.=
1.1 actually describes a stop-and-wait protocol. In my reading, the sentenc=
e "Once an ACK has been received for a message, the next message can be sen=
t" does not prevent the sender from transmitting several messages in parall=
el (but well, I am not a native speaker). To me, the text describes a windo=
wing protocol that is allowed to send one message per 10ms, i. e., in summa=
ry 100 messages per second.<br><br>If this section is about a stop-and-wait=
 protocol, then please mention that and use appropriate normative language.=
 For instance, stop-and-wait would result in something like "The next messa=
ge MUST only be sent after the ACK of the previous message has been receive=
d (or the retransmission finally times out)".<br><br>One reason why this se=
ction may be confusing is the "received" field, as already noted in my revi=
ew. A stop-and-wait protocol doesn't need it; it only makes sense if there =
is more than one message outstanding, i. e., if there is a send window. If =
the intention is to specify a stop-and-wait protocol only, then I suggest t=
o remove the "received" field from the spec. Anyway, other semantics for th=
is field might make sense if a better transport mechanism will be defined i=
n future, as already noted in my review.<br><br>A stop-and-wait protocol wi=
th a single outstanding message (not too large, i. e., not significantly ex=
ceeding 3-5 KB) and adaptive RTO including back-off would be rather safe to=
 use and sufficiently TCP friendly, according to RFC 5405. A corresponding =
rephrasing of these sections might address most of my concerns regarding co=
ngestion control.<br><br>Of course, such a stop-and-wait doesn't make sense=
 for any larger overlay, but apparently we agree on that.<br><br>Michael<br=
><br><br></blockquote></div><br></body></html>=

--_000_8CC519847A2A42DB84D1B3C828873BEEneustarbiz_--

From bbl@lowekamp.net  Wed Sep  7 20:10:30 2011
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In-Reply-To: <133D9897FB9C5E4E9DF2779DC91E947C06683549@SLFSNX.rcs.alcatel-research.de>
References: <133D9897FB9C5E4E9DF2779DC91E947C066833EF@SLFSNX.rcs.alcatel-research.de> <CAEOK=o=1Bd+K9=ieTVgMO=5k6sA6Qb+UaNOs1vXP2p5aGiDHUA@mail.gmail.com> <133D9897FB9C5E4E9DF2779DC91E947C066834EF@SLFSNX.rcs.alcatel-research.de> <CAEOK=o=dEDtn_t+L3BiZTkFS5R1SdpyDfu8rb=61xdijP0OK-Q@mail.gmail.com> <133D9897FB9C5E4E9DF2779DC91E947C06683549@SLFSNX.rcs.alcatel-research.de>
Date: Wed, 7 Sep 2011 23:12:02 -0400
Message-ID: <CAEOK=o=kEAXMvBMtbVvuh=ZPjQtB5k2EweqHBVnQ2_ke00AxDQ@mail.gmail.com>
From: Bruce Lowekamp <bbl@lowekamp.net>
To: "SCHARF, Michael" <Michael.Scharf@alcatel-lucent.com>
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Subject: Re: [P2PSIP] tsv-dir review of draft-ietf-p2psip-base-18
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On Wed, Sep 7, 2011 at 7:59 AM, SCHARF, Michael
<Michael.Scharf@alcatel-lucent.com> wrote:
> I'll just comment on one detail below, as it is indeed important
>
>> >> > Section 5.6.3.1: "In general, senders MAY implement any
>> >> rate control
>> >> > scheme of their choice, provided that it is REQUIRED to
>> be no more
>> >> > aggressive then TFRC[RFC5348]. The following section describes a
>> >> > simple, inefficient scheme that complies with this requirement."
>> >> >
>> >> > =3D> Maybe I again miss something, but as far as I can see
>> >> the following
>> >> > sections doesn't fully explain how TFRC is implemented here. For
>> >> > example, TFRC requires information about the segment size;
>> >> this is not
>> >> > considered here. In general, it is not clear to me why a simple
>> >> > baseline implementation does not follow the TFRC protocol
>> >> (RFC 5348)
>> >> > more closely (or a light-weight user-space TCP-like protocol).
>> >> >
>> >>
>> >> again, simplicity won out. =C2=A0We believe what's there is a simpler
>> >> protocol (the simplest we could think of) that follows the
>> >> requirement of being less aggressive than TFRC.
>> >
>> > Just to repeat what I said: The draft doesn't explain how the
>> > requirement of being less aggressive than TFRC is indeed achieved.
>> >
>>
>> Regarding this and following discussion, maybe it would be
>> better to back up a level and make sure we're talking about
>> the same things.
>> 5.6.3.1.1, which is the only actual retransmission scheme
>> defined in the document says:
>>
>> =C2=A0 =C2=A0The node MUST
>> =C2=A0 =C2=A0double the time to wait after each retransmission.
>> =C2=A0 =C2=A0...
>> =C2=A0 =C2=A0Once an ACK has been received for a message, the next
>> message can be
>> =C2=A0 =C2=A0sent, but the peer SHOULD ensure that there is at least 10
>> ms between
>> =C2=A0 =C2=A0sending any two messages.
>>
>> so this is a stop and wait algorithm with exponential backoff
>> on retransmissions. =C2=A0Furthermore,
>>
>> =C2=A0 =C2=A0If all
>> =C2=A0 =C2=A0retransmissions for a message fail, then the sending node S=
HOULD
>> =C2=A0 =C2=A0close the connection routing the message.
>>
>> So it actually abandons a link if retransmissions don't work,
>> rather than continuing to put traffic into it.
>>
>> Is there any actual question that this is a safe (albeit
>> non-performant) algorithm?
>
> I don't think that the current text in section 5.6.3.1.1 actually describ=
es a stop-and-wait protocol. In my reading, the sentence "Once an ACK has b=
een received for a message, the next message can be sent" does not prevent =
the sender from transmitting several messages in parallel (but well, I am n=
ot a native speaker). To me, the text describes a windowing protocol that i=
s allowed to send one message per 10ms, i. e., in summary 100 messages per =
second.
>
> If this section is about a stop-and-wait protocol, then please mention th=
at and use appropriate normative language. For instance, stop-and-wait woul=
d result in something like "The next message MUST only be sent after the AC=
K of the previous message has been received (or the retransmission finally =
times out)".
>

Agree.

> One reason why this section may be confusing is the "received" field, as =
already noted in my review. A stop-and-wait protocol doesn't need it; it on=
ly makes sense if there is more than one message outstanding, i. e., if the=
re is a send window. If the intention is to specify a stop-and-wait protoco=
l only, then I suggest to remove the "received" field from the spec. Anyway=
, other semantics for this field might make sense if a better transport mec=
hanism will be defined in future, as already noted in my review.
>

At one point there was an additional more TCP-like sender algorithm,
which we removed at some point due to concerns over complexity and the
hope that a suitable alternative would arrive via TSV.  But we wanted
to leave the receiver algorithm in there as-is with the potential of
allowing an improved sender algorithm to interact with it if such a
thing evolved.

> A stop-and-wait protocol with a single outstanding message (not too large=
, i. e., not significantly exceeding 3-5 KB) and adaptive RTO including bac=
k-off would be rather safe to use and sufficiently TCP friendly, according =
to RFC 5405. A corresponding rephrasing of these sections might address mos=
t of my concerns regarding congestion control.
>

On that issue, would it be better to explicitly state section 3.1.1 of
5405 for other non stop-and-wait senders rather than TFRC itself?  I'm
not quite sure why we were pointed at TFRC---it may have actually
predated 5405, but that once you go beyond stop-and-wait, 3.1.1 is the
section that applies (as 3.1.3 case 2 points back to 3.1.1).

Thanks,
Bruce


> Of course, such a stop-and-wait doesn't make sense for any larger overlay=
, but apparently we agree on that.
>
> Michael
>

From brian.rosen@neustar.biz  Thu Sep  8 09:41:11 2011
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From: "Rosen, Brian" <Brian.Rosen@neustar.biz>
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From: "SCHARF, Michael" <Michael.Scharf@alcatel-lucent.com<mailto:Michael.S=
charf@alcatel-lucent.com>>
Subject: RE: tsv-dir review of draft-ietf-p2psip-base-18
Date: September 8, 2011 3:36:20 AM EDT
To: "Bruce Lowekamp" <bbl@lowekamp.net<mailto:bbl@lowekamp.net>>
Cc: <iesg@ietf.org<mailto:iesg@ietf.org>>, <tsv-dir@ietf.org<mailto:tsv-dir=
@ietf.org>>, <tsv-area@ietf.org<mailto:tsv-area@ietf.org>>, <p2psip@ietf.or=
g<mailto:p2psip@ietf.org>>, <draft-ietf-p2psip-base@tools.ietf.org<mailto:d=
raft-ietf-p2psip-base@tools.ietf.org>>


A stop-and-wait protocol with a single outstanding message
(not too large, i. e., not significantly exceeding 3-5 KB)
and adaptive RTO including back-off would be rather safe to
use and sufficiently TCP friendly, according to RFC 5405. A
corresponding rephrasing of these sections might address most
of my concerns regarding congestion control.


On that issue, would it be better to explicitly state section 3.1.1 of
5405 for other non stop-and-wait senders rather than TFRC
itself?  I'm not quite sure why we were pointed at TFRC---it
may have actually predated 5405, but that once you go beyond
stop-and-wait, 3.1.1 is the section that applies (as 3.1.3
case 2 points back to 3.1.1).

The document should probably reference RFC 5405, as the guidelines are
obviously relevant for RELOAD in general.

The first sentence of Section 3.1.1 in RFC 5405 states that TFRC or
something equivalent SHOULD be used once going beyond stop-and-wait. So,
finally it comes down to TFRC whatever reference you use.

IMHO the simplest solution for moving this forward would be just to
specify stop-and-wait and to postpone considerations regarding a
TCP-friendly transport mechanism with better performance to a future
document.

Michael




--_000_DB3583A8EC87432E9AF70EA1D4B2B7A1neustarbiz_
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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode:=
 space; -webkit-line-break: after-white-space; "><div><blockquote type=3D"c=
ite"><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px;"><span style=3D"font-family:'Helvetica'; font-size:medium=
; color:rgba(127, 127, 127, 1.0);"><b>From: </b></span><span style=3D"font-=
family:'Helvetica'; font-size:medium;">"SCHARF, Michael" &lt;<a href=3D"mai=
lto:Michael.Scharf@alcatel-lucent.com">Michael.Scharf@alcatel-lucent.com</a=
>&gt;<br></span></div><div style=3D"margin-top: 0px; margin-right: 0px; mar=
gin-bottom: 0px; margin-left: 0px;"><span style=3D"font-family:'Helvetica';=
 font-size:medium; color:rgba(127, 127, 127, 1.0);"><b>Subject: </b></span>=
<span style=3D"font-family:'Helvetica'; font-size:medium;"><b>RE: tsv-dir r=
eview of draft-ietf-p2psip-base-18</b><br></span></div><div style=3D"margin=
-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 12=
7, 1.0);"><b>Date: </b></span><span style=3D"font-family:'Helvetica'; font-=
size:medium;">September 8, 2011 3:36:20 AM EDT<br></span></div><div style=
=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0p=
x;"><span style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(12=
7, 127, 127, 1.0);"><b>To: </b></span><span style=3D"font-family:'Helvetica=
'; font-size:medium;">"Bruce Lowekamp" &lt;<a href=3D"mailto:bbl@lowekamp.n=
et">bbl@lowekamp.net</a>&gt;<br></span></div><div style=3D"margin-top: 0px;=
 margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span style=3D"f=
ont-family:'Helvetica'; font-size:medium; color:rgba(127, 127, 127, 1.0);">=
<b>Cc: </b></span><span style=3D"font-family:'Helvetica'; font-size:medium;=
">&lt;<a href=3D"mailto:iesg@ietf.org">iesg@ietf.org</a>&gt;, &lt;<a href=
=3D"mailto:tsv-dir@ietf.org">tsv-dir@ietf.org</a>&gt;, &lt;<a href=3D"mailt=
o:tsv-area@ietf.org">tsv-area@ietf.org</a>&gt;, &lt;<a href=3D"mailto:p2psi=
p@ietf.org">p2psip@ietf.org</a>&gt;, &lt;<a href=3D"mailto:draft-ietf-p2psi=
p-base@tools.ietf.org">draft-ietf-p2psip-base@tools.ietf.org</a>&gt;<br></s=
pan></div><br><br><blockquote type=3D"cite"><blockquote type=3D"cite">A sto=
p-and-wait protocol with a single outstanding message <br></blockquote></bl=
ockquote><blockquote type=3D"cite">(not too large, i. e., not significantly=
 exceeding 3-5 KB) <br></blockquote><blockquote type=3D"cite">and adaptive =
RTO including back-off would be rather safe to <br></blockquote><blockquote=
 type=3D"cite">use and sufficiently TCP friendly, according to RFC 5405. A =
<br></blockquote><blockquote type=3D"cite">corresponding rephrasing of thes=
e sections might address most <br></blockquote><blockquote type=3D"cite">of=
 my concerns regarding congestion control.<br></blockquote><blockquote type=
=3D"cite"><blockquote type=3D"cite"><br></blockquote></blockquote><blockquo=
te type=3D"cite"><br></blockquote><blockquote type=3D"cite">On that issue, =
would it be better to explicitly state section 3.1.1 of<br></blockquote><bl=
ockquote type=3D"cite">5405 for other non stop-and-wait senders rather than=
 TFRC <br></blockquote><blockquote type=3D"cite">itself? &nbsp;I'm not quit=
e sure why we were pointed at TFRC---it <br></blockquote><blockquote type=
=3D"cite">may have actually predated 5405, but that once you go beyond <br>=
</blockquote><blockquote type=3D"cite">stop-and-wait, 3.1.1 is the section =
that applies (as 3.1.3 <br></blockquote><blockquote type=3D"cite">case 2 po=
ints back to 3.1.1).<br></blockquote><br>The document should probably refer=
ence RFC 5405, as the guidelines are<br>obviously relevant for RELOAD in ge=
neral.<br><br>The first sentence of Section 3.1.1 in RFC 5405 states that T=
FRC or<br>something equivalent SHOULD be used once going beyond stop-and-wa=
it. So,<br>finally it comes down to TFRC whatever reference you use.<br><br=
>IMHO the simplest solution for moving this forward would be just to<br>spe=
cify stop-and-wait and to postpone considerations regarding a<br>TCP-friend=
ly transport mechanism with better performance to a future<br>document.<br>=
<br>Michael<br><br><br></blockquote></div><br></body></html>=

--_000_DB3583A8EC87432E9AF70EA1D4B2B7A1neustarbiz_--

From gonzalo.camarillo@ericsson.com  Mon Sep 12 00:34:31 2011
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Folks,

we have created the RELOAD implementers mailing list:

  <reload@implementers.org>

If you are implementing, have implemented, or intend to implement
RELOAD, please join this list. We hope to build up a community of RELOAD
implementers around this forum.

You can find information on this list under the following link:

  http://implementers.org/mailman/listinfo/reload

"This list is for discussing implementation issues for the current
version of RELOAD, including questions on current protocol features and
extensions. Protocol development issues are discussed on the
p2psip@ietf.org list.

The first posting for a new member is always moderated, so it can take
up to 24 hours for this post to appear.

Commercial advertisements of any form (for products, software, jobs) are
inappropriate. Announcements related to RELOAD interoperability test
events or FOSS software are welcome. Product names may be mentioned if
necessary ("Software X, Y and Z does it that way."), while disparaging
or general comments ("Software W sucks rocks.") are inappropriate.
Because there is no way to correctly process them, emails containing
legal boilerplate are inappropriate.

Never cross-post a message to both the RELOAD list (p2psip@ietf.org) and
this list."

Cheers,

Gonzalo


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From zongning@huawei.com  Mon Sep 19 01:43:10 2011
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From petithug@acm.org  Mon Sep 19 08:39:00 2011
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I thought these drafts were adopted as WG document in Quebec City.

On 09/19/2011 01:43 AM, ZongNing wrote:
> Hi,
> 
> We have submitted a new version of Direct Response Routing draft as following link:
> http://tools.ietf.org/id/draft-zong-p2psip-drr-01.txt
> 
> Main changes are:
> 1) Change flag value of Ignore-State-Keeping to 0x8
> 2) Change extensive_routing_mode type value to 0x2 in IANA section
> 3) Update the definition of destination in structure ExtensiveRoutingModeOption
> 4) Update references
> 
> Review and comments are welcome.
> 
> BR,
> Ning Zong
> 
> -----邮件原件-----
> 发件人: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org] 
> 发送时间: 2011年9月19日 15:32
> 收件人: ZongNing
> 抄送: zhangyunfei@chinamobile.com; Johnson Jiang(Xingfeng); even.roni@huawei.com; ZongNing
> 主题: New Version Notification for draft-zong-p2psip-drr-01.txt
> 
> A new version of I-D, draft-zong-p2psip-drr-01.txt has been successfully submitted by Ning Zong and posted to the IETF repository.
> 
> Filename:	 draft-zong-p2psip-drr
> Revision:	 01
> Title:		 An extension to RELOAD to support Direct Response Routing
> Creation date:	 2011-09-19
> WG ID:		 Individual Submission
> Number of pages: 18
> 
> Abstract:
>    This document proposes an optional extension to RELOAD to support
>    direct response routing mode.  RELOAD recommends symmetric recursive
>    routing for routing messages.  The new optional extension provides a
>    shorter route for responses reducing the overhead on intermediary
>    peers and describes the potential cases where this extension can be
>    used.
> 

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Mon Sep 19 13:55:57 2011
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In section 5.7:

"Each fragment MUST contain a full copy of the via and
 destination list and MUST contain at least 256 bytes of the message
 body.  If the via and destination list are so large that this is not
 possible, RELOAD fragmentation is not performed and IP-layer
 fragmentation is allowed to occur."

I think that the text should say this instead:

"Each fragment MUST contain a full copy of the via list,
 destination list and options and MUST contain at least 256 bytes of
 the message body.  If the via list, destination list and options are
 so large that this is not possible, RELOAD fragmentation is not
 performed and IP-layer fragmentation is allowed to occur."

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From Even.roni@huawei.com  Mon Sep 19 14:28:07 2011
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Date: Tue, 20 Sep 2011 00:28:54 +0300
From: Roni Even <Even.roni@huawei.com>
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To: 'Marc Petit-Huguenin' <petithug@acm.org>, 'ZongNing' <zongning@huawei.com>
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Subject: Re: [P2PSIP] =?utf-8?b?6L2s5Y+ROiBOZXcgVmVyc2lvbiBOb3RpZmljYXRpb24g?= =?utf-8?q?for_draft-zong-p2psip-drr-01=2Etxt?=
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These is from the notes from Quebec sent by David to the list

"Additionally, the chairs would like to solicit the list to confirm the =
following consensus reached by hum at the IETF-81 meeting.

     1) Hum that the RELOAD base draft (-17) is ready for publication =
(hum indicated strong consensus)

     2) Hum that draft-zong-p2psip-drr-00 and draft-zong-p2psip-rpr-00 =
be adopted as WG items to address milestones for direct and recursive =
routing (hum indicated consensus)

Again, please send any comments to the list."



> -----Original Message-----
> From: p2psip-bounces@ietf.org [mailto:p2psip-bounces@ietf.org] On
> Behalf Of Marc Petit-Huguenin
> Sent: Monday, September 19, 2011 6:41 PM
> To: ZongNing
> Cc: p2psip@ietf.org
> Subject: Re: [P2PSIP] =E8=BD=AC=E5=8F=91: New Version Notification for =
draft-zong-
> p2psip-drr-01.txt
>=20
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>=20
> I thought these drafts were adopted as WG document in Quebec City.
>=20
> On 09/19/2011 01:43 AM, ZongNing wrote:
> > Hi,
> >
> > We have submitted a new version of Direct Response Routing draft as
> following link:
> > http://tools.ietf.org/id/draft-zong-p2psip-drr-01.txt
> >
> > Main changes are:
> > 1) Change flag value of Ignore-State-Keeping to 0x8
> > 2) Change extensive_routing_mode type value to 0x2 in IANA section
> > 3) Update the definition of destination in structure
> ExtensiveRoutingModeOption
> > 4) Update references
> >
> > Review and comments are welcome.
> >
> > BR,
> > Ning Zong
> >
> > -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> > =E5=8F=91=E4=BB=B6=E4=BA=BA: internet-drafts@ietf.org =
[mailto:internet-drafts@ietf.org]
> > =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: =
2011=E5=B9=B49=E6=9C=8819=E6=97=A5 15:32
> > =E6=94=B6=E4=BB=B6=E4=BA=BA: ZongNing
> > =E6=8A=84=E9=80=81: zhangyunfei@chinamobile.com; Johnson =
Jiang(Xingfeng);
> even.roni@huawei.com; ZongNing
> > =E4=B8=BB=E9=A2=98: New Version Notification for =
draft-zong-p2psip-drr-01.txt
> >
> > A new version of I-D, draft-zong-p2psip-drr-01.txt has been
> successfully submitted by Ning Zong and posted to the IETF repository.
> >
> > Filename:	 draft-zong-p2psip-drr
> > Revision:	 01
> > Title:		 An extension to RELOAD to support Direct Response
> Routing
> > Creation date:	 2011-09-19
> > WG ID:		 Individual Submission
> > Number of pages: 18
> >
> > Abstract:
> >    This document proposes an optional extension to RELOAD to support
> >    direct response routing mode.  RELOAD recommends symmetric
> recursive
> >    routing for routing messages.  The new optional extension =
provides
> a
> >    shorter route for responses reducing the overhead on intermediary
> >    peers and describes the potential cases where this extension can
> be
> >    used.
> >
>=20
> - --
> Marc Petit-Huguenin
> Personal email: marc@petit-huguenin.org
> Professional email: petithug@acm.org
> Blog: http://blog.marc.petit-huguenin.org
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> nCEAn1ybcoEAHZITmEr3uMwxBA0MZmPM
> =3Dhkhi
> -----END PGP SIGNATURE-----
> _______________________________________________
> P2PSIP mailing list
> P2PSIP@ietf.org
> https://www.ietf.org/mailman/listinfo/p2psip


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-----BEGIN PGP SIGNED MESSAGE-----
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On 09/19/2011 02:28 PM, Roni Even wrote:
> These is from the notes from Quebec sent by David to the list
> 
> "Additionally, the chairs would like to solicit the list to confirm the following consensus reached by hum at the IETF-81 meeting.
> 
>      1) Hum that the RELOAD base draft (-17) is ready for publication (hum indicated strong consensus)
> 
>      2) Hum that draft-zong-p2psip-drr-00 and draft-zong-p2psip-rpr-00 be adopted as WG items to address milestones for direct and recursive routing (hum indicated consensus)
> 
> Again, please send any comments to the list."

I support the adoption of draft-zong-p2psip-drr and draft-zong-p2psip-rpr as WG
documents.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From michaelc@IDSSOFTWARE.COM  Wed Sep 21 17:43:10 2011
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From: "Michael Chen" <michaelc@idssoftware.com>
To: p2psip@ietf.org
Date: Wed, 21 Sep 2011 17:45:37 -0700
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Subject: [P2PSIP] Base section 10.4 clarification
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Hi,=0A=0AA discussion with Marc Petit-Huguenin brings up the issue of "nake=
d=0APing" described in section 10.4 of the base draft:=0A=0A   If no cached=
 bootstrap nodes are available and the configuration file=0A   has an multi=
cast-bootstrap element, then the node SHOULD send a Ping=0A   request over =
UDP to the address and port found to each multicast-=0A   bootstrap element=
 found in the configuration document.  This MAY be a=0A   multicast, broadc=
ast, or anycast address.  The Ping should use the=0A   wildcard Node-ID as =
the destination Node-ID.=0A=0AIt should be clarified that this Ping message=
, wrapped the Frame Header=0Ais sent via UDP without DTLS, thus the term na=
ked Ping. Further=0Aimplication is that a RELOAD application that supports =
UDP must=0Amultiplex among three protocols: STUN, DTLS and framed_naked_Pin=
g.=0A=0AMarc also raised the issue of anycast, and I'll let him jump in fro=
m=0Ahere.=0A=0AThanks=0A=0AMichael Chen=0A

From petithug@acm.org  Thu Sep 22 09:29:25 2011
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-----BEGIN PGP SIGNED MESSAGE-----
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On 09/21/2011 05:45 PM, Michael Chen wrote:
> Hi,
> 
> A discussion with Marc Petit-Huguenin brings up the issue of "naked
> Ping" described in section 10.4 of the base draft:
> 
>    If no cached bootstrap nodes are available and the configuration file
>    has an multicast-bootstrap element, then the node SHOULD send a Ping
>    request over UDP to the address and port found to each multicast-
>    bootstrap element found in the configuration document.  This MAY be a
>    multicast, broadcast, or anycast address.  The Ping should use the
>    wildcard Node-ID as the destination Node-ID.
> 
> It should be clarified that this Ping message, wrapped the Frame Header
> is sent via UDP without DTLS, thus the term naked Ping. Further
> implication is that a RELOAD application that supports UDP must
> multiplex among three protocols: STUN, DTLS and framed_naked_Ping.
> 
> Marc also raised the issue of anycast, and I'll let him jump in from
> here.

The problem is that it is a bad idea to establish a DTLS connection to an
anycast address, as there is no guarantee that the subsequent UDP packets will
reach the same host.  It is even no guarantee that the ACK for the Ping answer
will go to the same host, which is why I also think that Framing should not be
used for sending a Ping to an anycast address.

But the problem in the case of anycast is NAT traversal.  We cannot use the
source IP/port of the Ping answer (or Ping request ACK) as an indication of the
unicast address to use for subsequent transactions, because the packet will be
dropped by symmetrical NATs.  The reasonable thing to do would have been to add
an IpAddressPort field in the Ping answer, field that contains the IP
address/port of a unicast bootstrap server (as if retrieved in the configuration
file), but the authors sent a clear message that breaking compatibility is out
of the equation.  So what do the authors propose to fix this problem?

Also I think that the spec should clearly state that a unicast bootstrap server
MUST support both DTLS-UDP-SR-NO-ICE and TLS-TCP-FH-NO-ICE on its public IP
address/port.

- -- 
Marc Petit-Huguenin
Personal email: marc@petit-huguenin.org
Professional email: petithug@acm.org
Blog: http://blog.marc.petit-huguenin.org
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From petithug@acm.org  Wed Sep 28 14:38:11 2011
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Message-ID: <4E839468.8000500@acm.org>
Date: Wed, 28 Sep 2011 14:40:56 -0700
From: Marc Petit-Huguenin <petithug@acm.org>
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Cc: "vipr@ietf.org" <vipr@ietf.org>, P2PSIP Mailing List <p2psip@ietf.org>
Subject: [P2PSIP] Fwd: I-D Action: draft-petithuguenin-dispatch-unique-overlay-00.txt
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I just published a short I-D describing a special kind of peer-to-peer overlay,
and listing some of the requirements for a mechanism to find the bootstrap
servers for these overlays.

The goal is to start a discussion on how to best solve this.

Please follow-up in the dispatch mailing-list.

Thanks.


- -------- Original Message --------
Subject: I-D Action: draft-petithuguenin-dispatch-unique-overlay-00.txt
Date: Wed, 28 Sep 2011 14:33:59 -0700
From: internet-drafts@ietf.org
Reply-To: internet-drafts@ietf.org
To: i-d-announce@ietf.org

A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : Infrastructure Overlay
	Author(s)       : Marc Petit-Huguenin
	Filename        : draft-petithuguenin-dispatch-unique-overlay-00.txt
	Pages           : 5
	Date            : 2011-09-28

   This document provides requirements for infrastructure overlays, a
   special kind of peer-to-peer overlay whose main purpose would be
   defeated if more than one instance would exist on the Internet.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-petithuguenin-dispatch-unique-overlay-00.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-petithuguenin-dispatch-unique-overlay-00.txt

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