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From: Mark Allman <mallman@icir.org>
Subject: tsv-dir review of draft-ietf-mile-rfc4046-bis-05
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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 basically ready for publication, but has nits that should
be fixed before publication. 

In reading this document I flagged two issues.  Neither of them huge,
but I'd think perhaps they should be addressed before publication.

(1) The language is a little wonky in my opinion.  The document is
    laying out a 'transport protocol'.  But, while this protocol does
    transport bits of data this is absolutely not a layer 4 protocol and
    so I was initially a bit confused.  This draft lays out an
    application layer protocol (a slightly tweaked version of HTTP to be
    exact).  It would seem useful to me to clean this up.

(2) I wondered why the document said hosts MUST use port 4590.
    Certainly having a well know port is useful in many cases.  But, I
    don't see why some consortium couldn't decide they were going to use
    port 4545 or whatever.  Likewise, when setting up a callback it'd
    seem straightforward to give a port number, as well.

    I am not sure it is the biggest deal in the world, but a solution
    that leveraged late binding would strikes me as more flexible and
    hence better.

allman




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From mallman@icir.org  Thu Jan 12 09:35:00 2012
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To: Brian Trammell <trammell@tik.ee.ethz.ch>
From: Mark Allman <mallman@icir.org>
Subject: Re: [mile] tsv-dir review of draft-ietf-mile-rfc4046-bis-05 
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> > (2) I wondered why the document said hosts MUST use port 4590.
> >    Certainly having a well know port is useful in many cases.  But, I
> >    don't see why some consortium couldn't decide they were going to use
> >    port 4545 or whatever.  Likewise, when setting up a callback it'd
> >    seem straightforward to give a port number, as well.
> > 
> >    I am not sure it is the biggest deal in the world, but a solution
> >    that leveraged late binding would strikes me as more flexible and
> >    hence better.
> 
> Agreed that the restriction is too strict. However, I think it's
> simpler at this point to declare the use of ports other than the
> well-known port to be configurable out of band, as other RID system
> and consortium configuration issues are out of scope for this document
> as well.

That is what I was thinking.  I.e., I was not suggesting some way to
discover the port or whatnot.  Just not mandating that all communication
happen on the assigned port (which of course you could never do
regardless of what the RFC says).

Sounds like a good set of changes to me.

allman




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From trammell@tik.ee.ethz.ch  Thu Jan 12 08:22:31 2012
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Subject: Re: [mile] tsv-dir review of draft-ietf-mile-rfc4046-bis-05
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Hi, Mark,

Thanks for the review; I've addressed these issues in an -06 revision to =
be submitted after last call is complete. Comments thereon, inline...

On Jan 11, 2012, at 7:52 AM, Mark Allman wrote:

>=20
> 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
>=20
> This draft is basically ready for publication, but has nits that =
should
> be fixed before publication.=20
>=20
> In reading this document I flagged two issues.  Neither of them huge,
> but I'd think perhaps they should be addressed before publication.
>=20
> (1) The language is a little wonky in my opinion.  The document is
>    laying out a 'transport protocol'.  But, while this protocol does
>    transport bits of data this is absolutely not a layer 4 protocol =
and
>    so I was initially a bit confused.  This draft lays out an
>    application layer protocol (a slightly tweaked version of HTTP to =
be
>    exact).  It would seem useful to me to clean this up.

Agreed completely, and done.

> (2) I wondered why the document said hosts MUST use port 4590.
>    Certainly having a well know port is useful in many cases.  But, I
>    don't see why some consortium couldn't decide they were going to =
use
>    port 4545 or whatever.  Likewise, when setting up a callback it'd
>    seem straightforward to give a port number, as well.
>=20
>    I am not sure it is the biggest deal in the world, but a solution
>    that leveraged late binding would strikes me as more flexible and
>    hence better.

Agreed that the restriction is too strict. However, I think it's simpler =
at this point to declare the use of ports other than the well-known port =
to be configurable out of band, as other RID system and consortium =
configuration issues are out of scope for this document as well. So, the =
clarifying text would be as follows:

    Every RID system participating in a
    consortium SHOULD listen for HTTP/TLS connections on the assigned =
port.
    RID systems MAY be configurable to listen on ports other than the
    well-known port; this configuration is out of band and out of scope =
for
    this specification.

and

    When performing
    RID callback, a responding system MUST connect to the host at the
    network-layer address from which the original request was sent; =
there is
    no mechanism in RID for redirected callback. This callback SHOULD =
use TCP
    port 4590 unless configured out of band to use a different port.

Many thanks, best regards,

Brian (as 6046-bis author)=

From jsaldana@unizar.es  Tue Jan 17 10:24:30 2012
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From: jsaldana@unizar.es
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Subject: Proposal for updating TCRTP (RFC 4170))
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Hello all.

My name is Jose Saldana, from the University of Zaragoza, in Spain. I am
new in this area.

I would like to share with Transport Area a proposal, in case it
could be considered interesting. The main idea is to update TCRTP. This
is a "Best current practice" RFC =20
(http://datatracker.ietf.org/doc/rfc4170/) which was developed in =20
Audio/Video Transport working group in 2005-11. It "describes a method =20
to improve the bandwidth utilization of RTP streams over network paths =20
that carry multiple Real-time Transport Protocol (RTP) streams in
parallel between two endpoints, as in voice trunking. The method
combines standard protocols that provide compression, multiplexing, and
tunneling over a network path for the purpose of reducing the bandwidth
used when multiple RTP streams are carried over that path." (from the
introduction of RFC4170).

I have first sent this proposal to avtcore working group, but Magnus =20
Westerlund has answered this:

________

Hi,

As you are proposing to go towards a more generic scheme I don't think
this proposal really belongs in the WG, most likely not this area even.

I think this is a topic that primarily are related to Intarea and
Transport. Intarea as the owner of the tunneling work in IETF and
Transport as the actual aggregation and header compression is primarily
transport topics.

A question I would have is where this type of mechanism would make
sense. Where do you have scarce bandwidth capacity and are willing to
pay the downsides of the ingress egress complexities and the impact on
the traffic in regards to additional delay.

Cheers

Magnus
___________________

So I would like to know if the work can better fit this area.

The scheme of TCRTP is:

   RTP/UDP/IP
      |
      |
    ECRTP (compressing layer). Compresses RTP/UDP/IP headers
      |
      |
     PPPMux (multiplexing layer). Includes a number of compressed packets
into a bigger one
      |
      |
    L2TP     (tunneling layer). Permits its use end-to-end
      |
      |
      IP

However, there exist many real-time flows that do not use RTP:

- Online "First Person Shooter" (FPS) games use UDP datagrams, and they
are one of the most "real-time" applications nowadays. Some examples:
Counter Strike, Halo, Quake, etc.[1].

- Online "Massively Multiplayer Online Role Playing Games" (MMORPG):
They use TCP, but can be considered "real-time", since the actions of
the players have to be transmitted very quickly to the server. It should
be taken into account that player vs player fights are one of the
possible activities of the game. Some examples: World of Warcraft (more
than 10 million subscribers) [2].

- TCP is also getting used for media delivery. For many reasons, such as
avoiding firewalls, the standard IP/ UDP/ RTP protocol stack is
substituted in many cases by IP/ TCP/ HTTP/ FLV [3].

- Another interesting scenario is satellite communication links.
Multiplexing small packets into one packet for transmission would
improve the efficiency. Satellite links would also find it useful to
multiplex small TCP packets into one packet - this could be especially
interesting for compressing TCP ACKs.


In our recent research activities, we have adapted TCRTP scheme for its
use when the traffic is not RTP. The idea is to substitute ECRTP by
other compression scheme which can be used for UDP or TCP headers. In
2006, the IETF formed the Robust Header Compression (ROHC) working group
which created specifications for header compression over links for RTP,
UDP and TCP.  These specifications were extended to compress headers
over other networks. For most RTP flows, ROHC is more bandwidth
efficient than ECRTP, at the cost of implementation complexity. It has a
special care of context synchronization, defining some states at the
compressor and decompressor. Once the effort for designing ROHC has been
conducted, it is worth including it in the compressing and multiplexing
standard.


In a recent paper in Nov. issue of IEEE Communications Magazine, it was
shown that 30% of bandwidth saving can be achieved for 8 FPS different
games using IPv4. If IPv6 is used, savings can be up to 54% [4]. Another
recent study, which is still under review, has estimated 45% bandwidth
saving for World of Warcraft.


After talking to other researchers, including some of the authors of
RFC4170, the possible scheme of the proposal could be this one:


G.711.0 payload compression        TCP     UDP   UDP/RTP
                  |                    |      |      |
                  |                     \     |     /
                  |                       \   |   /
                  |               nothing or ROHC or ECRTP
                  |                           |
                   \                         /
                     \                     /
                 multiplexing layer (other or PPPMUX)
                            |
                            |
                tunneling layer (other or GRE or L2TP)
                            |
                            |
                           IP

It should be more "open" than TCRTP. I mean, it could be "orthogonal" to
different protocols: different header compression algorithms could be
used, depending on the application and the scenario; also different
multiplexing and tunneling protocols could be included. Finally, we have
also included the G.711.0 "branch": it could be interesting to convert
G.711 to G.711.0 (a lossless encoding of G.711) and back, using a
similar tunneling mechanism. In a similar way to TCRTP, the G.711 to
G.711.0 and vice versa conversion does not involve SIP/SDP/RTSP
signaling with the endpoints, since G.711 has a fixed RTP payload number.

Regarding the scenarios, we have considered some of them:

- A provider of a real-time service (e.g. game or voip) can have an =20
infrastructure with a
number of proxies that can group traffic from different users and send =20
it to the server.

- An Internet caf=E9, where a number (sometimes big) of players share =20
the same access link.
Bandwidth is scarce in that scenario.

- A satellite link, where the bandwidth and the pps have to be saved.


We have not written a draft. This is only the first idea, so we look
forward to hearing the opinion of the Area.


Thank you very much, and sorry for writing such a long message,

Jose Saldana

University of Zaragoza

______________

[1] Ratti, S.; Hariri, B.; Shirmohammadi, S., "A Survey of First-Person
Shooter Gaming Traffic on the Internet," Internet Computing, IEEE ,
vol.14, no.5, pp.60-69, Sept.-Oct. 2010. URL:
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5445069&isnumber=
=3D5562482
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5445069&isnumbe=
r=3D5562482>

[2] P. Svoboda, W Karner and M. Rupp, "Traffic Analysis and Modeling for
World of Warcraft," ICC '07. IEEE International Conference on
Communications, pp.1612-1617, 24-28, Jun. 2007. URL:
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D4288941&isnumber=
=3D4288671
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D4288941&isnumbe=
r=3D4288671>

[3] G. Marfia and M. Roccetti, "TCP At Last: Reconsidering TCP's Role
for Wireless Entertainment Centers at Home," IEEE Transactions on
Consumer Electronics, Vol. 56, N. 4, Nov. 2010, pp. 2233-2240. URL:
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5681095&isnumber=
=3D5681060
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5681095&isnumbe=
r=3D5681060>

[4] J. Saldana, J. Fern=E1ndez-Navajas, J. Ruiz-Mas, J. I. Aznar, E.
Viruete and  L. Casadesus, "First Person Shooters: Can a Smarter Network
Save Bandwidth without Annoying the Players?," IEEE Communications
Magazine, Consumer Communications and Networking Series, vol. 49, no.
11, pp. 190-198, Nov. 2011. URL:
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D6069728&isnumber=
=3D6069696
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D6069728&isnumbe=
r=3D6069696>




From wes@mti-systems.com  Thu Jan 19 18:38:33 2012
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On 12/23/2011 12:06 AM, Wesley Eddy wrote:
> Hello, as we plan for a potential TSVAREA meeting in Paris at
> IETF-83 (http://www.ietf.org/meeting/83/index.html), please
> let David and myself know of any topics you would like to
> discuss in a TSVAREA meeting, so that we can plan for the time
> appropriately.


Just a reminder; the meeting scheduling cut-off is approaching
(on 1/30), so please get your requests in if there are topics
of area-wide interest that you'd like to discuss in-person.


-- 
Wes Eddy
MTI Systems

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Subject: Re: Proposal for updating TCRTP (RFC 4170))
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I would agree with Magnus that the TSV Area would be a good place to =
start with a more detailed discussion of the topic, although the =
Transport Area Directors would be able to provide a more definitive =
opinion.  The one word of caution I would bring up is that the IETF =
tends to have a difficult time finding a home for architecture work that =
spans layers of the stack.  I don't know if this aspect is a requirement =
or a one potential use case scenario of your proposed work, but its =
something to keep in mind if you end up giving a presentation at the TSV =
area meeting.

-ken


On Jan 17, 2012, at 1:24 PM, jsaldana@unizar.es wrote:

> Hello all.
>=20
> My name is Jose Saldana, from the University of Zaragoza, in Spain. I =
am
> new in this area.
>=20
> I would like to share with Transport Area a proposal, in case it
> could be considered interesting. The main idea is to update TCRTP. =
This
> is a "Best current practice" RFC =
(http://datatracker.ietf.org/doc/rfc4170/) which was developed in =
Audio/Video Transport working group in 2005-11. It "describes a method =
to improve the bandwidth utilization of RTP streams over network paths =
that carry multiple Real-time Transport Protocol (RTP) streams in
> parallel between two endpoints, as in voice trunking. The method
> combines standard protocols that provide compression, multiplexing, =
and
> tunneling over a network path for the purpose of reducing the =
bandwidth
> used when multiple RTP streams are carried over that path." (from the
> introduction of RFC4170).
>=20
> I have first sent this proposal to avtcore working group, but Magnus =
Westerlund has answered this:
>=20
> ________
>=20
> Hi,
>=20
> As you are proposing to go towards a more generic scheme I don't think
> this proposal really belongs in the WG, most likely not this area =
even.
>=20
> I think this is a topic that primarily are related to Intarea and
> Transport. Intarea as the owner of the tunneling work in IETF and
> Transport as the actual aggregation and header compression is =
primarily
> transport topics.
>=20
> A question I would have is where this type of mechanism would make
> sense. Where do you have scarce bandwidth capacity and are willing to
> pay the downsides of the ingress egress complexities and the impact on
> the traffic in regards to additional delay.
>=20
> Cheers
>=20
> Magnus
> ___________________
>=20
> So I would like to know if the work can better fit this area.
>=20
> The scheme of TCRTP is:
>=20
>  RTP/UDP/IP
>     |
>     |
>   ECRTP (compressing layer). Compresses RTP/UDP/IP headers
>     |
>     |
>    PPPMux (multiplexing layer). Includes a number of compressed =
packets
> into a bigger one
>     |
>     |
>   L2TP     (tunneling layer). Permits its use end-to-end
>     |
>     |
>     IP
>=20
> However, there exist many real-time flows that do not use RTP:
>=20
> - Online "First Person Shooter" (FPS) games use UDP datagrams, and =
they
> are one of the most "real-time" applications nowadays. Some examples:
> Counter Strike, Halo, Quake, etc.[1].
>=20
> - Online "Massively Multiplayer Online Role Playing Games" (MMORPG):
> They use TCP, but can be considered "real-time", since the actions of
> the players have to be transmitted very quickly to the server. It =
should
> be taken into account that player vs player fights are one of the
> possible activities of the game. Some examples: World of Warcraft =
(more
> than 10 million subscribers) [2].
>=20
> - TCP is also getting used for media delivery. For many reasons, such =
as
> avoiding firewalls, the standard IP/ UDP/ RTP protocol stack is
> substituted in many cases by IP/ TCP/ HTTP/ FLV [3].
>=20
> - Another interesting scenario is satellite communication links.
> Multiplexing small packets into one packet for transmission would
> improve the efficiency. Satellite links would also find it useful to
> multiplex small TCP packets into one packet - this could be especially
> interesting for compressing TCP ACKs.
>=20
>=20
> In our recent research activities, we have adapted TCRTP scheme for =
its
> use when the traffic is not RTP. The idea is to substitute ECRTP by
> other compression scheme which can be used for UDP or TCP headers. In
> 2006, the IETF formed the Robust Header Compression (ROHC) working =
group
> which created specifications for header compression over links for =
RTP,
> UDP and TCP.  These specifications were extended to compress headers
> over other networks. For most RTP flows, ROHC is more bandwidth
> efficient than ECRTP, at the cost of implementation complexity. It has =
a
> special care of context synchronization, defining some states at the
> compressor and decompressor. Once the effort for designing ROHC has =
been
> conducted, it is worth including it in the compressing and =
multiplexing
> standard.
>=20
>=20
> In a recent paper in Nov. issue of IEEE Communications Magazine, it =
was
> shown that 30% of bandwidth saving can be achieved for 8 FPS different
> games using IPv4. If IPv6 is used, savings can be up to 54% [4]. =
Another
> recent study, which is still under review, has estimated 45% bandwidth
> saving for World of Warcraft.
>=20
>=20
> After talking to other researchers, including some of the authors of
> RFC4170, the possible scheme of the proposal could be this one:
>=20
>=20
> G.711.0 payload compression        TCP     UDP   UDP/RTP
>                 |                    |      |      |
>                 |                     \     |     /
>                 |                       \   |   /
>                 |               nothing or ROHC or ECRTP
>                 |                           |
>                  \                         /
>                    \                     /
>                multiplexing layer (other or PPPMUX)
>                           |
>                           |
>               tunneling layer (other or GRE or L2TP)
>                           |
>                           |
>                          IP
>=20
> It should be more "open" than TCRTP. I mean, it could be "orthogonal" =
to
> different protocols: different header compression algorithms could be
> used, depending on the application and the scenario; also different
> multiplexing and tunneling protocols could be included. Finally, we =
have
> also included the G.711.0 "branch": it could be interesting to convert
> G.711 to G.711.0 (a lossless encoding of G.711) and back, using a
> similar tunneling mechanism. In a similar way to TCRTP, the G.711 to
> G.711.0 and vice versa conversion does not involve SIP/SDP/RTSP
> signaling with the endpoints, since G.711 has a fixed RTP payload =
number.
>=20
> Regarding the scenarios, we have considered some of them:
>=20
> - A provider of a real-time service (e.g. game or voip) can have an =
infrastructure with a
> number of proxies that can group traffic from different users and send =
it to the server.
>=20
> - An Internet caf=E9, where a number (sometimes big) of players share =
the same access link.
> Bandwidth is scarce in that scenario.
>=20
> - A satellite link, where the bandwidth and the pps have to be saved.
>=20
>=20
> We have not written a draft. This is only the first idea, so we look
> forward to hearing the opinion of the Area.
>=20
>=20
> Thank you very much, and sorry for writing such a long message,
>=20
> Jose Saldana
>=20
> University of Zaragoza
>=20
> ______________
>=20
> [1] Ratti, S.; Hariri, B.; Shirmohammadi, S., "A Survey of =
First-Person
> Shooter Gaming Traffic on the Internet," Internet Computing, IEEE ,
> vol.14, no.5, pp.60-69, Sept.-Oct. 2010. URL:
> =
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5445069&isnumb=
er=3D5562482
> =
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5445069&isnum=
ber=3D5562482>
>=20
> [2] P. Svoboda, W Karner and M. Rupp, "Traffic Analysis and Modeling =
for
> World of Warcraft," ICC '07. IEEE International Conference on
> Communications, pp.1612-1617, 24-28, Jun. 2007. URL:
> =
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D4288941&isnumb=
er=3D4288671
> =
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D4288941&isnum=
ber=3D4288671>
>=20
> [3] G. Marfia and M. Roccetti, "TCP At Last: Reconsidering TCP's Role
> for Wireless Entertainment Centers at Home," IEEE Transactions on
> Consumer Electronics, Vol. 56, N. 4, Nov. 2010, pp. 2233-2240. URL:
> =
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5681095&isnumb=
er=3D5681060
> =
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D5681095&isnum=
ber=3D5681060>
>=20
> [4] J. Saldana, J. Fern=E1ndez-Navajas, J. Ruiz-Mas, J. I. Aznar, E.
> Viruete and  L. Casadesus, "First Person Shooters: Can a Smarter =
Network
> Save Bandwidth without Annoying the Players?," IEEE Communications
> Magazine, Consumer Communications and Networking Series, vol. 49, no.
> 11, pp. 190-198, Nov. 2011. URL:
> =
http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D6069728&isnumb=
er=3D6069696
> =
<http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=3D&arnumber=3D6069728&isnum=
ber=3D6069696>
>=20
>=20
>=20

