
From prasanna2602@gmail.com  Thu May  3 07:04:45 2012
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From: Prasanna Kumar <prasanna2602@gmail.com>
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Subject: [tsvwg] Query on the SCTP primary path recovery
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--e89a8fb200aa2a7ab004bf2249a3
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Hi All,

I have a query regarding the primary path recovery.

In SCTP RFC, it only talks about the setting an inactive path into active
once it receives the heartbeat ack from the peer on that path. In case if
the primary path is down for some time and it comes back later, is it
required to restore the primary path back to original one and use it for
further data transmission?

Thanks in advance.

Regards,
Prasanna :-)

--e89a8fb200aa2a7ab004bf2249a3
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Hi All,<br><br>I have a query regarding the primary path recovery.<br><br>I=
n SCTP RFC, it only talks about the setting an inactive path into active on=
ce it receives the heartbeat ack from the peer on that path. In case if the=
 primary path is down for some time and it comes back later, is it required=
 to restore the primary path back to original one and use it for further da=
ta transmission?<br>
<br>Thanks in advance.<br clear=3D"all"><br>Regards,<br>Prasanna :-)<br>

--e89a8fb200aa2a7ab004bf2249a3--

From Michael.Tuexen@lurchi.franken.de  Thu May  3 09:19:05 2012
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Subject: Re: [tsvwg] Query on the SCTP primary path recovery
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On May 3, 2012, at 4:04 PM, Prasanna Kumar wrote:

> Hi All,
>=20
> I have a query regarding the primary path recovery.
>=20
> In SCTP RFC, it only talks about the setting an inactive path into =
active once it receives the heartbeat ack from the peer on that path. In =
case if the primary path is down for some time and it comes back later, =
is it required to restore the primary path back to original one and use =
it for further data transmission?
It is an implementation decision. The SCTP implementation in FreeBSD =
does not change
the primary path once it becomes unreachable. So, if it is reachable =
again, it will get
the traffic again.

Best regards
Michael
>=20
> Thanks in advance.
>=20
> Regards,
> Prasanna :-)


From prasanna2602@gmail.com  Thu May  3 22:55:23 2012
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From: Prasanna Kumar <prasanna2602@gmail.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, tsvwg@ietf.org
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Subject: Re: [tsvwg] Query on the SCTP primary path recovery
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--f46d0447f32ce420e504bf2f90ed
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Thanks Michael,

So from your answer, it is clear that we can use the primary path for the
data transmission as earlier once it becomes active/reachable. Thanks a lot
for quick answer.

Regards,
Prasanna.

On Thu, May 3, 2012 at 9:49 PM, Michael Tuexen <
Michael.Tuexen@lurchi.franken.de> wrote:

> On May 3, 2012, at 4:04 PM, Prasanna Kumar wrote:
>
> > Hi All,
> >
> > I have a query regarding the primary path recovery.
> >
> > In SCTP RFC, it only talks about the setting an inactive path into
> active once it receives the heartbeat ack from the peer on that path. In
> case if the primary path is down for some time and it comes back later, is
> it required to restore the primary path back to original one and use it for
> further data transmission?
> It is an implementation decision. The SCTP implementation in FreeBSD does
> not change
> the primary path once it becomes unreachable. So, if it is reachable
> again, it will get
> the traffic again.
>
> Best regards
> Michael
> >
> > Thanks in advance.
> >
> > Regards,
> > Prasanna :-)
>
>

--f46d0447f32ce420e504bf2f90ed
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Thanks Michael,<br><br>So from your answer, it is clear that we can use the=
 primary path for the data transmission as earlier once it becomes active/r=
eachable. Thanks a lot for quick answer.<br><br>Regards,<br>Prasanna.<br>
<br><div class=3D"gmail_quote">On Thu, May 3, 2012 at 9:49 PM, Michael Tuex=
en <span dir=3D"ltr">&lt;<a href=3D"mailto:Michael.Tuexen@lurchi.franken.de=
" target=3D"_blank">Michael.Tuexen@lurchi.franken.de</a>&gt;</span> wrote:<=
br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left=
:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5">On May 3, 2012, at 4:04 PM, Prasann=
a Kumar wrote:<br>
<br>
&gt; Hi All,<br>
&gt;<br>
&gt; I have a query regarding the primary path recovery.<br>
&gt;<br>
&gt; In SCTP RFC, it only talks about the setting an inactive path into act=
ive once it receives the heartbeat ack from the peer on that path. In case =
if the primary path is down for some time and it comes back later, is it re=
quired to restore the primary path back to original one and use it for furt=
her data transmission?<br>

</div></div>It is an implementation decision. The SCTP implementation in Fr=
eeBSD does not change<br>
the primary path once it becomes unreachable. So, if it is reachable again,=
 it will get<br>
the traffic again.<br>
<br>
Best regards<br>
<span class=3D"HOEnZb"><font color=3D"#888888">Michael<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5">&gt;<br>
&gt; Thanks in advance.<br>
&gt;<br>
&gt; Regards,<br>
&gt; Prasanna :-)<br>
<br>
</div></div></blockquote></div><br><br clear=3D"all"><br><br>

--f46d0447f32ce420e504bf2f90ed--

From gorry@erg.abdn.ac.uk  Tue May 15 00:02:12 2012
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Subject: [tsvwg] TSVWG - current status 15th May 2012
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The list below shows the status of the working group documents as we see 
it. Please check below and comment on drafts using the list. Please do 
send any corrections/omissions to the chairs.

Best wishes,

James and Gorry
(TSVWG Chairs)

---
Recently published:

draft-ietf-tsvwg-rsvp-security-groupkeying was published as RFC 6411.
draft-ietf-tsvwg-sctpsocket was published as RFC 6458.
draft-ietf-tsvwg-sctp-strrst was published as RFC 6535.

-------------------------------------------------------------------

IDs in RFC Editor Queue:

draft-ietf-tsvwg-source-quench
(Replaced: draft-gont-tsvwg-source-quench-01.txt)
Gorry is Shepherd.

-------------------------------------------------------------------

IDs in IESG processing:

none.

-------------------------------------------------------------------

WG Drafts with Chairs:

draft-ietf-tsvwg-byte-pkt-congest
New editor added Jul 2010.
New revision 25 Oct 2010.
WG -01, June 10, 2011
IETF-81: discussed whether this should be BCP.
ADs & Chairs agree to progress as a BCP
Status changed to BCP (new-ID WG -05)
Presented at IEWTF-82 (Taipei), request for WGLC.
Gorry added as Shepherd (Jan 2012)
WGLC concluded with comments on I-D, Friday 30th March 2012.

DUE: Waiting for author corrections/comments based on WGLC.

draft-ietf-tsvwg-sctp-udp-encap
(Replaced: draft-tuexen-sctp-udp-encap)
Lars sent a note as AD agreeing to progress this work.
Work to be coordinated with DCCP work on encaps.
draft-tuexen-sctp-udp-encaps-06, January 10, 2011
Adopted as a work item 21 Sept 2010 (Gorry)
WG -02 8 Dec 2011
WGLC completed Friday 20th April 2012
DUE: Authors to update draft.


-------------------------------------------------------------------

WG Drafts:

draft-ietf-tsvwg-natsupp-tsvwg
(Replaced: draft-stewart-natsupp-tsvw)
Dependency from BEHAVE WG.
Adopted as a work item 21 Sept 2010 (Gorry).
WG -00, 29/11/2010
Uploaded as: draft-ietf-natsupp-tsvwg
WG -01, June 1, 2011
DUE: Authors to restructure draft.
DUE: Reviews need.

draft-ietf-tsvwg-intserv-multiple-tspec
WG interest in this topic recorded at IETF-78.
Charter update would be needed to progress this work.
5 Reviews needed to determine energy/technical direction.
WG -05 (presented in Beijing, IETF-79)
WG -06 (presented in Prague, IETF-80)
RSVP directorate was consulted.
2 reviews from RSVP-DIR received (Bruce Davie, ?)
2 additional reviews promised (Ken Carlberg, Francois LeF)
Chairs asked AD for a Charter update (IESG agreed)
Draft discussed at IETF-80, and request to update charter agreed
- AD advised 4 named reviewers will be required
Adopted for progression as PS, for May 2012

draft-ietf-tsvwg-rsvp-pcn
Previous version : draft-karagiannis-pcn-tsvwg-rsvp-pcn
WG interest in this topic recorded at IETF-81.
Individual -01, 2011-07-11
AD decision allowed this to be added to the milestones
The document was adopted
WG -00 8 Oct 2011
DUE: New revision need

draft-touch-tsvwg-port-use-00
(replaces draft-touch-tsvwg-port-use-00)
- Intended to be advice to protocol designers needing a port.
5 people have looked at this document, Prague IETF-80
Individual -01 July 2011
IETF-81 insufficient feedback from WG at this time.
WG needs to assess if the new draft should be a work item.
IETF-82, discussed - Chairs will ask WG to consider.
Adopted by WG
Please comment on list.

-------------------------------------------------------------------

WG action required:

draft-nishida-tsvwg-sctp-failover
Individual-00 Presented in Prague, IETF-80.
- understood not to conflict with draft-tuexen-tsvwg-sctp-multipath
DUE: WG needs to discuss the need for adoption
DUE: New revision need.

draft-tuexen-tsvwg-sctp-multipath
-00 Presented in Prague, IETF-80.
- understood not to conflict with draft-nishida-tsvwg-sctp-failover
-01, 2010-12-27
Please comment to list.

draft-tuexen-tsvwg-sctp-sack-immediately
-05, 2011-01-02
- Aug 2011, discussed on list and issues raised.
DUE: Chairs need to confirm adoption (need to do work).
DUE: New revision need (to address comments)
Please comment on list.

draft-stewart-tsvwg-sctpecn-02
Partially replaces (draft-stewart-tsvwg-sctp-nonce)
IETF-78 suggested there was interest in this topic.
Discussed at IETF-83, Paris.
DUE: ECN is within the TSVWG Charter,
will call for adoption on the list.
Please comment on list.
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The following have received recent discussion at TSVWG meetings or on
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draft-carlberg-tsvwg-ecn-reactions
Reactions to Signaling from ECN Support for RTP/RTCP
-00 presented IETF-82
WG needs to assess if the new draft should be a work item.
Please comment on list.

draft-polk-tsvwg-rsvp-app-id-vv-profiles
-01 Presented in Beijing, IETF-79.
-02, 14-Mar-2011
Presented IET-80.
WG needs to assess if the new draft should be a work item.
Seek to coordinate with music with partner ID:
draft-polk-mmusic-traffic-class-for-sdp
Please comment on list.
DUE: Chairs will liaise with MMUSIC WG (Gorry).
-------------------------------------------------------------------

draft-lefaucheur-tsvwg-rsvp-multiple-preemption

RSVP directorate to be consulted.
- Reviewers: Bruce Davie (awaiting other reviews)
Please comment to list.
WG meeting: Please ask the list if the simplest
option is enough, and use that to help decide.
A new revision will be needed.

draft-polk-tsvwg-rfc4594-update-00

Discussed at IETF-83, Paris
A Revised ID is needed.

draft-briscoe-tsvwg-ecn-encap-guidelines
-00 Presented in Prague, IETF-80.
- comments at IETF-80 (see minutes)
WG needs to assess if the new draft should be a work item.
Please comment on list.
DUE: Please check liaison status with other SDOs.

draft-saldana-tsvwg-tcmtf-00

Discussed at IETF-83, Paris
No decision was made, please comment on the list.

-------------------------------------------------------------------

Related non-WG items:

draft-ietf-behave-sctpnat-03.txt
BEHAVE WG item linked to SCTP encapsulation work.

draft-ietf-6man-udpzero
Previoiusly: draft-fairhurst-tsvwg-6man-udpzero
Draft was adopted by 6man, please discuss on the 6man list.
A WGLC of this draft is expected in 6MAN WG.
Related draft: draft-ietf-6man-udpchecksums-00
This draft was also LC'ed in TSVWG (end 3rd May 2012)

------------------------------------------------------------------

Other news:

RSVP Directorate (formed in May 2010)
Charter updated Aug 2011.

------------------------------------------------------------------



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Thread-Topic: Server is sending INIT- issue in SCTP
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=20

=20

Hi,

We are facing an issue related to SCTP. Whenever the SCTP server is
restarted, the server behaves properly for some iterations and then for

some iterations it fails to send the packet on wire. Therefore script
timeouts without getting any response.

=20

Here is the detail informations:

We are using  linux version:2.6.9-55.ELsmp

=20

Client is sending INIT to Server, association is completed & data
communication is working fine. Then after hot-swap in Server side,
Server is sending INIT to Client & association is happening as usual.
But for low load (2k LV) data communications is working fine but at high
load value (4k LV) our SCTP server is failed to process the data for
sometime & then process the next data successfully. Like this packet
loss counter is increasing which is not captured in wireshark.

=20

We have found one difference between the 2 INITs (Client sending to
Server) & (Server sending to Client)

Client INIT supports 2 parameters (ECN parameter & Forward TSN supported
parameter) but Server doesn't support.

=20

=20

Is this a kernel issue?=20

=20

=20

Thanks,

Sailabala

=20


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<p class=3DMsoNormal><span =
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color:#365F91'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'>We are facing an issue related to SCTP. Whenever the SCTP =
server
is restarted, the server behaves properly for some iterations and then =
for<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'>some iterations it fails to send the packet on wire. =
Therefore
script timeouts without getting any response.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><u><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'>Here is the detail informations</span></u><span
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";color:#244061=
'>:<o:p></o:p></span></p>

<p class=3DMsoNormal><b><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>We are using &nbsp;linux =
version:2.6.9-55.ELsmp<o:p></o:p></span></b></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'>Client is sending INIT to Server, association is =
completed &amp;
data communication is working fine. Then after hot-swap in Server side, =
Server
is sending INIT to Client &amp; association is happening as usual. But =
for low
load (2k LV) data communications is working fine but at <b>high load =
value (4k
LV)</b> our SCTP server is failed to process the data for sometime &amp; =
then
process the next data successfully. Like this packet loss counter is =
increasing
which is not captured in wireshark.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#244061'>We have found one difference between the 2 INITs (Client =
sending
to Server) &amp; (Server sending to Client)<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:8.0pt;font-family:"Verdana","sans-serif";
color:#244061'>Client INIT supports 2 parameters (<b>ECN parameter &amp;
Forward TSN supported parameter</b>) but Server doesn&#8217;t =
support.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:8.0pt;font-family:"Verdana","sans-serif";
color:#244061'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:8.0pt;font-family:"Verdana","sans-serif";
color:#244061'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:8.0pt;font-family:"Verdana","sans-serif";
color:#244061'>Is this a kernel issue? <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>Thanks,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>Sailabala<o:p></o:p></span></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

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From randall@lakerest.net  Tue May 15 03:59:45 2012
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To: Sailabala Lenka <sailabala.lenka@exfo.com>
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Subject: Re: [tsvwg] Server is sending INIT- issue in SCTP
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Sailabala:

I don't see how the client or server supporting ECN or F-TSN would make =
any
difference. Basically each side (C & S) describe in the INIT/INIT-ACK =
what
features they support. In this case one side is saying I can do
ECN and F-TSN the other side is not. If both sides do not agree then =
those
features are disabled.

Now there are always bugs. So its possible there could be an issue =
(which I kind of doubt).=20
To look for ECN you would look in the IP header in any wireshark =
trace/tcpdump. Use wireshark
to look at the packet and look at the IP header. You can check the =
Differentiated Services field. In here
you should see ECN: 0x00: Not-ECT (Not ECN-Capable Transport)

If a ECN flag is set then one of the sides is trying to do ECN when the =
other side did not
declare they understood it. Note that even if a side was doing this, it =
would not cause
packet loss *unless* a middle box was discarding packets with the ECN =
mark. Note you would
always see the packet at the sending side with a tcpdump/wireshark =
however.

For FWD-TSN the application would also have to send partial reliable =
packets and you would
see FWD-TSN's on the wire.. again the features *should* be disabled if =
both sides do not
do them, and I don't see how either could cause packet loss.=20

Note you mention Version:2.6.9-55.ELsmp for Linux, its seems strange to =
me that you don't
mention a different version on the other side. After all if one is =
declaring a different
feature it must be some other version.

This sounds more like a hardware problem with your hot-swap.

R
On May 15, 2012, at 6:06 AM, Sailabala Lenka wrote:

> =20
> =20
> Hi,
> We are facing an issue related to SCTP. Whenever the SCTP server is =
restarted, the server behaves properly for some iterations and then for
> some iterations it fails to send the packet on wire. Therefore script =
timeouts without getting any response.
> =20
> Here is the detail informations:
> We are using  linux version:2.6.9-55.ELsmp
> =20
> Client is sending INIT to Server, association is completed & data =
communication is working fine. Then after hot-swap in Server side, =
Server is sending INIT to Client & association is happening as usual. =
But for low load (2k LV) data communications is working fine but at high =
load value (4k LV) our SCTP server is failed to process the data for =
sometime & then process the next data successfully. Like this packet =
loss counter is increasing which is not captured in wireshark.
> =20
> We have found one difference between the 2 INITs (Client sending to =
Server) & (Server sending to Client)
> Client INIT supports 2 parameters (ECN parameter & Forward TSN =
supported parameter) but Server doesn=92t support.
> =20
> =20
> Is this a kernel issue?
> =20
> =20
> Thanks,
> Sailabala
> =20

-----
Randall Stewart
randall@lakerest.net





From Michael.Tuexen@lurchi.franken.de  Tue May 15 08:09:18 2012
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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To: Sailabala Lenka <sailabala.lenka@exfo.com>
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Subject: Re: [tsvwg] Server is sending INIT- issue in SCTP
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On May 15, 2012, at 12:06 PM, Sailabala Lenka wrote:

> =20
> =20
> Hi,
> We are facing an issue related to SCTP. Whenever the SCTP server is =
restarted, the server behaves properly for some iterations and then for
> some iterations it fails to send the packet on wire. Therefore script =
timeouts without getting any response.
> =20
> Here is the detail informations:
> We are using  linux version:2.6.9-55.ELsmp
> =20
> Client is sending INIT to Server, association is completed & data =
communication is working fine. Then after hot-swap in Server side, =
Server is sending INIT to Client & association is happening as usual. =
But for low load (2k LV) data communications is working fine but at high =
load value (4k LV) our SCTP server is failed to process the data for =
sometime & then process the next data successfully. Like this packet =
loss counter is increasing which is not captured in wireshark.
> =20
> We have found one difference between the 2 INITs (Client sending to =
Server) & (Server sending to Client)
> Client INIT supports 2 parameters (ECN parameter & Forward TSN =
supported parameter) but Server doesn=92t support.
> =20
> =20
> Is this a kernel issue?
Most likely it is an issue related to the LKSCTP or the kernel. Not =
sure.
I don't know why the INITs are different, but you might want to bring up =
this
issue on the Linux SCTP mailing list.

Best regards
Michael
> =20
> =20
> Thanks,
> Sailabala
> =20


From jsaldana@unizar.es  Wed May 16 00:42:14 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
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Hello all. I would like to discuss an idea that somebody suggested me in
IETF83 Paris: the current draft considers one tunnel including a number of
flows, but all these flows are of the same protocol, i.e.: one tunnel for
multiplexing a number of VoIP flows using RTP; another tunnel for a number
of UDP flows which belong to FPS games; another tunnel for a number of TCP
flows of an MMORPG game, etc.

 

But perhaps it could be possible to include in the same tunnel packets
belonging to flows of different protocols. An access provider could use the
same tunnel for services with similar delay constraints (e.g. a tunnel
including VoIP and FPS games traffic). The advantage is clear: the bigger
the number of flows, the bigger the bandwidth saving and the smaller the
added multiplexing delays.

 

However, on the other hand, maybe some problems would arise (signaling
issues, etc.).

 

So I would appreciate to receive some ideas about this:

-          Do you think it could be useful?

-          Some other scenarios in which different protocols (RTP, bare UDP,
TCP) could be tunneled together?

-          Problems that could arise?

 

Thank you,

 

Jose Saldana

 

PS: If you have not read the draft, the ideas are also included in a paper
that is going to be presented in IEEE ICC, in Ottawa, next June 10th. A
preliminary version of the paper can be found in
http://diec.unizar.es/~jsaldana/personal/widening_scope_draft.pdf

 

It has a good explanation and also includes (color) graphs of the savings
obtained for VoIP, FPS and MMORPG games

 

The current draft is in
http://datatracker.ietf.org/doc/draft-saldana-tsvwg-tcmtf/


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</o:shapelayout></xml><![endif]--></head><body lang=3DES link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US>Hello all. I would like to discuss an idea that somebody =
suggested me in IETF83 Paris: the current draft considers one tunnel =
including a number of flows, but all these flows are of the same =
protocol, i.e.: one tunnel for multiplexing a number of VoIP flows using =
RTP; another tunnel for a number of UDP flows which belong to FPS games; =
another tunnel for a number of TCP flows of an MMORPG game, =
etc.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>But perhaps it could be possible to include in the same =
tunnel packets belonging to flows of different protocols. An access =
provider could use the same tunnel for services with similar delay =
constraints (e.g. a tunnel including VoIP and FPS games traffic). The =
advantage is clear: the bigger the number of flows, the bigger the =
bandwidth saving and the smaller the added multiplexing =
delays.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>However, on the other hand, maybe some problems would arise =
(signaling issues, etc.).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>So I would appreciate to receive =
some ideas about this:<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Do you think it could =
be useful?<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Some other scenarios =
in which different protocols (RTP, bare UDP, TCP) could be tunneled =
together?<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Problems that could =
arise?<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Thank you,<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";mso-fareast-la=
nguage:ES'>Jose Saldana</span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";mso-fareast-la=
nguage:ES'><o:p></o:p></span></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US>PS: If you have not read the draft, the ideas are also =
included in a paper that is going to be presented in IEEE ICC, in =
Ottawa, next June 10th. A preliminary version of the paper can be found =
in =
http://diec.unizar.es/~jsaldana/personal/widening_scope_draft.pdf<o:p></o=
:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>It has a good explanation and also includes (color) graphs =
of the savings obtained for VoIP, FPS and MMORPG =
games<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>The current draft is in =
http://datatracker.ietf.org/doc/draft-saldana-tsvwg-tcmtf/<o:p></o:p></sp=
an></p></div></body></html>
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From Mirko.Suznjevic@fer.hr  Fri May 18 02:49:20 2012
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Hello,

=20

As I am new to this group just a short introduction, my name is Mirko =
Suznjevic, I am from University of Zagreb, Faculty of Electrical =
Engineering and Computing, and my research is mainly in the area of =
network games.

=20

Just regarding the applicability of a such solution.

As more from the perspective of a typical "gamer" or the extensive user =
of games,  I can tell that there is high concurrency of gaming and VoIP =
flows, especially in multiplayer games.

=20

While a significant part of the gaming community which uses VoIP tends =
to use external VoIP tools such as Ventrilo, Skype, TeamSpeak, and =
Mumble, more and more recently published games have VoIP support  built =
in the client application.

As an example of recent very popular games with inbuilt VoIP features  =
we can take Battlefield 3, World of Warcraft, Heroes of Newerth etc.

=20

While the gaming traffic does not end at the same server as the voice =
traffic (example from my World of Warcraft measurements is 80.239.179.39 =
for game traffic and 80.239.179.18 for VoIP traffic), it is evident that =
the majority of the network path will be the same.

=20

Also, as a real world application in which bandwidth savings are =
significant we can take big LAN parties, or big gaming events such as =
DreamHack (last event had around 20,000 players and over 1,6 million =
stream viewers, therefore organizers needed to upgrade their connection =
to 120 GB/s breaking the world record).

=20

Regards,

Mirko Suznjevic

=20

=20

From: tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] On Behalf =
Of Jose Saldana
Sent: Wednesday, May 16, 2012 9:42 AM
To: tsvwg@ietf.org
Subject: [tsvwg] A possible improvement for =
draft-saldana-tsvwg-tcmtf-02:asking for feedback

=20

Hello all. I would like to discuss an idea that somebody suggested me in =
IETF83 Paris: the current draft considers one tunnel including a number =
of flows, but all these flows are of the same protocol, i.e.: one tunnel =
for multiplexing a number of VoIP flows using RTP; another tunnel for a =
number of UDP flows which belong to FPS games; another tunnel for a =
number of TCP flows of an MMORPG game, etc.

=20

But perhaps it could be possible to include in the same tunnel packets =
belonging to flows of different protocols. An access provider could use =
the same tunnel for services with similar delay constraints (e.g. a =
tunnel including VoIP and FPS games traffic). The advantage is clear: =
the bigger the number of flows, the bigger the bandwidth saving and the =
smaller the added multiplexing delays.

=20

However, on the other hand, maybe some problems would arise (signaling =
issues, etc.).

=20

So I would appreciate to receive some ideas about this:

-        Do you think it could be useful?

-        Some other scenarios in which different protocols (RTP, bare =
UDP, TCP) could be tunneled together?

-        Problems that could arise?

=20

Thank you,

=20

Jose Saldana

=20

PS: If you have not read the draft, the ideas are also included in a =
paper that is going to be presented in IEEE ICC, in Ottawa, next June =
10th. A preliminary version of the paper can be found in =
http://diec.unizar.es/~jsaldana/personal/widening_scope_draft.pdf

=20

It has a good explanation and also includes (color) graphs of the =
savings obtained for VoIP, FPS and MMORPG games

=20

The current draft is in =
http://datatracker.ietf.org/doc/draft-saldana-tsvwg-tcmtf/


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-GB link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hello,<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>As I am new to this =
group just a short introduction, my name is Mirko Suznjevic, I am from =
University of Zagreb, Faculty of Electrical Engineering and Computing, =
and my research is mainly in the area of network =
games.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Just regarding the =
applicability of a such solution.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>As more from the =
perspective of a typical &#8220;gamer&#8221; or the extensive user of =
games, &nbsp;I can tell that there is high concurrency of gaming and =
VoIP flows, especially in multiplayer games.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>While a significant part =
of the gaming community which uses VoIP tends to use external VoIP tools =
such as Ventrilo, Skype, TeamSpeak, and Mumble, more and more recently =
published games have VoIP support &nbsp;built in the client =
application.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'>As an example of recent very popular games with =
inbuilt VoIP features &nbsp;we can take Battlefield 3, World of =
Warcraft, Heroes of Newerth etc.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>While the gaming traffic =
does not end at the same server as the voice traffic (example from my =
World of Warcraft measurements is 80.239.179.39 for game traffic and =
80.239.179.18 for VoIP traffic), it is evident that the majority of the =
network path will be the same.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Also, as a real world =
application in which bandwidth savings are significant we can take big =
LAN parties, or big gaming events such as DreamHack (last event had =
around 20,000 players and over 1,6 million stream viewers, therefore =
organizers needed to upgrade their connection to 120 GB/s breaking the =
world record).<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'>Regards,<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'color:#1F497D'>Mirko =
Suznjevic<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
tsvwg-bounces@ietf.org [mailto:tsvwg-bounces@ietf.org] <b>On Behalf Of =
</b>Jose Saldana<br><b>Sent:</b> Wednesday, May 16, 2012 9:42 =
AM<br><b>To:</b> tsvwg@ietf.org<br><b>Subject:</b> [tsvwg] A possible =
improvement for draft-saldana-tsvwg-tcmtf-02:asking for =
feedback<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US>Hello all. I would like to discuss an idea that somebody =
suggested me in IETF83 Paris: the current draft considers one tunnel =
including a number of flows, but all these flows are of the same =
protocol, i.e.: one tunnel for multiplexing a number of VoIP flows using =
RTP; another tunnel for a number of UDP flows which belong to FPS games; =
another tunnel for a number of TCP flows of an MMORPG game, =
etc.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>But perhaps it could be possible to include in the same =
tunnel packets belonging to flows of different protocols. An access =
provider could use the same tunnel for services with similar delay =
constraints (e.g. a tunnel including VoIP and FPS games traffic). The =
advantage is clear: the bigger the number of flows, the bigger the =
bandwidth saving and the smaller the added multiplexing =
delays.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>However, on the other hand, maybe some problems would arise =
(signaling issues, etc.).<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US>So I would appreciate to receive =
some ideas about this:<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Do you think it could =
be useful?<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Some other scenarios =
in which different protocols (RTP, bare UDP, TCP) could be tunneled =
together?<o:p></o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if =
!supportLists]><span lang=3DEN-US><span style=3D'mso-list:Ignore'>-<span =
style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span lang=3DEN-US>Problems that could =
arise?<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Thank you,<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Jose =
Saldana</span><span lang=3DES =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p><p class=3DMsoNormal><span =
lang=3DES><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>PS: If you have not read the draft, the ideas are also =
included in a paper that is going to be presented in IEEE ICC, in =
Ottawa, next June 10th. A preliminary version of the paper can be found =
in <a =
href=3D"http://diec.unizar.es/~jsaldana/personal/widening_scope_draft.pdf=
">http://diec.unizar.es/~jsaldana/personal/widening_scope_draft.pdf</a><o=
:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>It has a good explanation and also includes (color) graphs =
of the savings obtained for VoIP, FPS and MMORPG =
games<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>The current draft is in <a =
href=3D"http://datatracker.ietf.org/doc/draft-saldana-tsvwg-tcmtf/">http:=
//datatracker.ietf.org/doc/draft-saldana-tsvwg-tcmtf/</a><o:p></o:p></spa=
n></p></div></body></html>
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From karagian@cs.utwente.nl  Wed May 23 23:58:48 2012
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From: <karagian@cs.utwente.nl>
To: <jmpolk@cisco.com>, <tsvwg@ietf.org>
Thread-Topic: Comments on draft-polk-tsvwg-rsvp-app-id-vv-profiles-03
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Hi James, Hi all,

I think that this ID is useful and that it can become a (tsvwg) WG document=
. However, I have some comments:

Comment_1: Section 1 should enhance/enlarge the text that motivates why thi=
s document is needed!
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Comment_2: In Section 2, page 3 is mentioned:
=93No other Sub-types are defined by
   any profile within this document, but can be included by individual=20
   implementations=94
It would be better to use MAY be instead of =93can be=94, see below:
=93No other Sub-types are defined by
   any profile within this document, but MAY be included by individual=20
   implementations=94
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Comment_3: At the introductory part of Section 3 emphasize that the IANA re=
gistries associated with the application profiles are given in Section 5.2.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Comment_4: In Section 3.1 it is emphasized that the Application profiles ar=
e including the field =93VER=94, but the semantics of this field are not ex=
plained.

=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Comment_5: Section 5.2.1:
o) In  Broadcast Audio IPTV Profile, please chnage from: APP=3Dbroadcast.vi=
deo.iptv, VER=3D"
INTO:
APP=3Dbroadcast.audio.iptv, VER=3D"

o) The name of the fourth described profile should be changed from:
Broadcast Audio Live-events Profile
INTO:
Broadcast Video Live-events Profile
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Comment_6: Section 5.2.3:
o) The name of the first described profile could be changed from:
Multimedia-Conferencing Profile
INTO:
Multimedia-Conferencing Profile =96 Presentation data

o) The name of the second described profile could be changed from:
Multimedia-Conferencing Profile
INTO:
Multimedia-Conferencing Profile =96 Application Sharing

o) The name of the third described profile could be changed from:
Multimedia-Conferencing Profile
INTO:
Multimedia-Conferencing Profile =96 Whiteboarding
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Comment_7: Section 5.2.4:
o) The name of the first described profile could be changed from:
  Multimedia-Streaming Profile
INTO:
  Multimedia-Streaming Profile =96 Multiplex
o) The name of the second described profile could be changed from:
 =20
  Multimedia-Streaming Profile
INTO:
Multimedia-Streaming Profile =96 Webcast
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Comment_8: The security considerations section (Section 4) is not clear, se=
e below:
=93Given that each traffic flow is within separate reservations, and=20
   RSVP does not have the ability to police the bits within any=20
   reservation, solving for this appears to be administratively handled
   at best. This is not meant to be a 'punt', but there really is=20
   nothing this template creates that is going to make things any=20
   harder for anyone (that we know of now).=94
Please elaborate!
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Comment_9: The reference to RFC 2750 is not listed in Section 8, while t is=
 used in the text.


Best regards,
Georgios



________________________________________
Van: tsvwg-bounces@ietf.org [tsvwg-bounces@ietf.org] namens James M. Polk [=
jmpolk@cisco.com]
Verzonden: woensdag 14 maart 2012 0:42
Aan: tsvwg
Onderwerp: draft-polk-tsvwg-rsvp-app-id-vv-profiles uploaded

TSVWG

(as author)

I've uploaded the next version of

http://www.ietf.org/internet-drafts/draft-polk-tsvwg-rsvp-app-id-vv-profile=
s-03.txt

The following changes were made in this version:

- Added draft-ietf-mmusic-traffic-class-for-sdp-01.txt as a
   reference

- Changed to a new format of the profile string.

- Added many new profiles based on the new format into each parent
   category of Section 3.

- changed the GUID to refer to draft-ietf-mmusic-traffic-class-for-
   sdp-01.txt

- changed 'desktop' adjective to 'avconf' to keep in alignment with
   draft-ietf-mmusic-traffic-class-for-sdp-01.txt

- Have a complete IANA Registry proposal for each application-ID
   discussed in this draft.

- General text clean-up of the draft.

I don't believe there are any open issues with this document, and
hope to ask the WG to adopt this draft during the Paris meeting.

James=

From Karen.Nielsen@tieto.com  Thu May 24 00:24:08 2012
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From: <Karen.Nielsen@tieto.com>
To: <tsvwg@ietf.org>
Date: Thu, 24 May 2012 10:23:58 +0300
Thread-Topic: RFC4960 Section 8.1 and HB impacts on association error counter
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Hi,

The purpose of this email is to solicit views on the extent of which the fa=
te of
HBs should impact the association error counter of SCTP.

The purpose of the association error counter in RFC4960 is to monitor the a=
liveness/failure of the peer endpoint.
When speaking about aliveness is seems relevant to differentiate in between=
 the aliveness of the peer endpoint
and the "aliveness of the association" in terms of the "capability of the a=
ssociation to forward data to the peer endpoint".
Whereas the aliveness of the peer of course is required for the association=
 to be of any use,
then it is the latter that is of prime relevance for the SCTP Users and it =
is preferable if it is the latter type of "aliveness"
that governs association closures and thus the exact handling of the associ=
ation error counter.
In this context it generally seems inappropriate for HBs to contribute to t=
he association error counter.

Well - in idle situations only HBs can drive the association error counter =
and here it makes good sense that their fate
impact the association error counter (at least for the HBs that are transmi=
tted on the current primary path).

But in the situation where data is being transmitted then it is problematic=
 and even distorting to have the fate
of HBs transmitted over auxiliary paths impact the association error counte=
r. By auxiliary paths I here mean paths
different from data carrying path(s). Reasons being:

Having successful HBs on some path reset the association error counter effe=
ctively results in that data may keep
on being retransmitted in vain many more consecutive times that what the AM=
R threshold indicates, either because
the intrinsic SCTP path selection is inadequate and not moves date (re)tran=
smission to a working path or because
HBs are let through (e.g., due to their potential small packet size or due =
to HW HB ack mechanism) whereas data consistently is lost.

More importantly then if HBs that fail on auxiliary paths increment the ass=
ociation error counter, then in situations
where there are many paths and thus potentially many HBs transmitted within=
 a short period of time
(even when randomly distributed there will or can be peaks), such failed
HBs diminish the room available in the association error counter/AMR for fa=
iled data (re)transmissions.
Effectively the failure of HBs on auxiliary back-up paths renders the assoc=
iation less robust
to survive data failures on the current primary path.

RFC4960 in section 8.1 stipulates that the fate of HBs shall contribute to =
the association error counter.
(Exception being HBs transmitted for Path Verification, but that is a diffe=
rent topic):



   An endpoint shall keep a counter on the total number of consecutive

   retransmissions to its peer (this includes retransmissions to all the

   destination transport addresses of the peer if it is multi-homed),

   including unacknowledged HEARTBEAT chunks.

  ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^If the value of this

   counter exceeds the limit indicated in the protocol parameter

   'Association.Max.Retrans', the endpoint shall consider the peer

   endpoint unreachable and shall stop transmitting any more data to it

   (and thus the association enters the CLOSED state).  In addition, the

   endpoint MAY report the failure to the upper layer and optionally

   report back all outstanding user data remaining in its outbound

   queue.  The association is automatically closed when the peer

   endpoint becomes unreachable.



   The counter shall be reset each time a DATA chunk sent to that peer

   endpoint is acknowledged (by the reception of a SACK) or a HEARTBEAT

                                                             ^^^^^^^^^^

   ACK is received from the peer endpoint.

   ^^^^^^



But it is also noteworthy (or is it?) that no RFC KEYWORDS are used in the =
formulations in this respect.

As there only seems to be down-sides of having HBs contribute to the associ=
ation error counter (exception being idle situation)
I thus wonder whether the inclusion of HBs in Section 8.1. of the RFC was i=
ntended primarily to cover the idle situation and/or
if the lack of KEYWORDS somehow was motivated by similar concerns as the ab=
ove that was discussed at the time of the definition of the feature ?

Along the same lines then I am wondering if it would be considered a breach=
 of the RFC4960 if an SCTP implementation
chose to have the association error counter be driven only by the fate of d=
ata transmissions (except for the idle situation) ?

The issue becomes particularly relevant when considering the aggressive HB =
probing mechanisms
of http://tools.ietf.org/html/draft-nishida-tsvwg-sctp-failover-05,
as there here can be very many Path monitoring HBs flowing on auxiliary pat=
hs compared to data transmissions.
But the issue is of generic nature and equally relevant for a basic RFC4960=
 SCTP implementation.

All thoughts are most welcome ! :)

Thanks in advance.

BR, Karen

PS: It is relevant to notice that path failure information based on unackno=
wledged HBs are considerable less
reliable than path failure input resulting from data T3 timeouts. This as f=
ast retransmission procedures for data in many
cases may cover for single packet failures, whereas no mechanism is in plac=
e to cover for single drops of HBs.


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<o:shapelayout v:ext=3D"edit">
<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span style=3D'f=
ont-size:10.0pt;font-family:"Arial","sans-serif"'>Hi,<o:p></o:p></span></p>=
<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D=
'font-size:10.0pt;font-family:"Arial","sans-serif"'>The purpose of this ema=
il is to solicit views on the extent of which the fate of <o:p></o:p></span=
></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Aria=
l","sans-serif"'>HBs should impact the association error counter of SCTP.<o=
:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fo=
nt-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoN=
ormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The=
 purpose of the association error counter in RFC4960 is to monitor the aliv=
eness/failure of the peer endpoint.<o:p></o:p></span></p><p class=3DMsoNorm=
al><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>When s=
peaking about aliveness is seems relevant to differentiate in between the a=
liveness of the peer endpoint <o:p></o:p></span></p><p class=3DMsoNormal><s=
pan style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>and the &#8=
220;aliveness of the association&#8221; in terms of the &#8220;capability o=
f the association to forward data to the peer endpoint&#8221;. <o:p></o:p><=
/span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:=
"Arial","sans-serif"'>Whereas the aliveness of the peer of course is requir=
ed for the association to be of any use, <o:p></o:p></span></p><p class=3DM=
soNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=
then it is the latter that is of prime relevance for the SCTP Users and it =
is preferable if it is the latter type of &#8220;aliveness&#8221; <o:p></o:=
p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-fami=
ly:"Arial","sans-serif"'>that governs association closures and thus the exa=
ct handling of the association error counter.<o:p></o:p></span></p><p class=
=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-seri=
f"'>In this context it generally seems inappropriate for HBs to contribute =
to the association error counter.<o:p></o:p></span></p><p class=3DMsoNormal=
><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nb=
sp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fo=
nt-family:"Arial","sans-serif"'>Well - in idle situations only HBs can driv=
e the association error counter and here it makes good sense that their fat=
e<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt=
;font-family:"Arial","sans-serif"'>impact the association error counter (at=
 least for the HBs that are transmitted on the current primary path).<o:p><=
/o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-f=
amily:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNorma=
l><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>But in =
the situation where data is being transmitted then it is problematic and ev=
en distorting to have the fate <o:p></o:p></span></p><p class=3DMsoNormal><=
span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>of HBs tra=
nsmitted over auxiliary paths impact the association error counter. By auxi=
liary paths I here mean paths<o:p></o:p></span></p><p class=3DMsoNormal><sp=
an style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>different fr=
om data carrying path(s). Reasons being:<o:p></o:p></span></p><p class=3DMs=
oNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><=
o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10=
.0pt;font-family:"Arial","sans-serif"'>Having successful HBs on some path r=
eset the association error counter effectively results in that data may kee=
p <o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0p=
t;font-family:"Arial","sans-serif"'>on being retransmitted in vain many mor=
e consecutive times that what the AMR threshold indicates, either because<o=
:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fo=
nt-family:"Arial","sans-serif"'>the intrinsic SCTP path selection is inadeq=
uate and not moves date (re)transmission to a working path or because <o:p>=
</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-=
family:"Arial","sans-serif"'>HBs are let through (e.g., due to their potent=
ial small packet size or due to HW HB ack mechanism) whereas data consisten=
tly is lost. <o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font=
-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>More importantly then if HBs that fail on auxiliary paths incre=
ment the association error counter, then in situations <o:p></o:p></span></=
p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial",=
"sans-serif"'>where there are many paths and thus potentially many HBs tran=
smitted within a short period of time <o:p></o:p></span></p><p class=3DMsoN=
ormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>(ev=
en when randomly distributed there will or can be peaks), such failed <o:p>=
</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-=
family:"Arial","sans-serif"'>HBs diminish the room available in the associa=
tion error counter/AMR for failed data (re)transmissions.<o:p></o:p></span>=
</p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial=
","sans-serif"'>Effectively the failure of HBs on auxiliary back-up paths r=
enders the association less robust <o:p></o:p></span></p><p class=3DMsoNorm=
al><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>to sur=
vive data failures on the current primary path.<o:p></o:p></span></p><p cla=
ss=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-se=
rif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-=
size:10.0pt;font-family:"Arial","sans-serif"'>RFC4960 in section 8.1 stipul=
ates that the fate of HBs shall contribute to the association error counter=
.<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt=
;font-family:"Arial","sans-serif"'>(Exception being HBs transmitted for Pat=
h Verification, but that is a different topic):<o:p></o:p></span></p><p cla=
ss=3DMsoPlainText><span lang=3DEN-GB style=3D'font-family:"Courier New"'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-GB style=
=3D'font-family:"Courier New"'>&nbsp;&nbsp; An endpoint shall keep a counte=
r on the total number of consecutive<o:p></o:p></span></p><p class=3DMsoPla=
inText><span lang=3DEN-GB style=3D'font-family:"Courier New"'>&nbsp;&nbsp; =
retransmissions to its peer (this includes retransmissions to all the<o:p><=
/o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-GB style=3D'font-fa=
mily:"Courier New"'>&nbsp;&nbsp; destination transport addresses of the pee=
r if it is multi-homed),<o:p></o:p></span></p><p class=3DMsoPlainText><span=
 lang=3DEN-GB style=3D'font-family:"Courier New"'>&nbsp;&nbsp; including un=
acknowledged HEARTBEAT chunks.<o:p></o:p></span></p><p class=3DMsoPlainText=
><span lang=3DEN-GB style=3D'font-family:"Courier New"'>&nbsp; ^^^^^^^^^^^^=
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^If the value of this<o:p></o:p></span></p><=
p class=3DMsoPlainText><span lang=3DEN-GB style=3D'font-family:"Courier New=
"'>&nbsp;&nbsp; counter exceeds the limit indicated in the protocol paramet=
er<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-GB style=3D=
'font-family:"Courier New"'>&nbsp;&nbsp; 'Association.Max.Retrans', the end=
point shall consider the peer<o:p></o:p></span></p><p class=3DMsoPlainText>=
<span lang=3DEN-GB style=3D'font-family:"Courier New"'>&nbsp;&nbsp; endpoin=
t unreachable and shall stop transmitting any more data to it<o:p></o:p></s=
pan></p><p class=3DMsoPlainText><span lang=3DEN-GB style=3D'font-family:"Co=
urier New"'>&nbsp;&nbsp; (and thus the association enters the CLOSED state)=
.&nbsp; In addition, the<o:p></o:p></span></p><p class=3DMsoPlainText><span=
 lang=3DEN-GB style=3D'font-family:"Courier New"'>&nbsp;&nbsp; endpoint MAY=
 report the failure to the upper layer and optionally<o:p></o:p></span></p>=
<p class=3DMsoPlainText><span lang=3DEN-GB style=3D'font-family:"Courier Ne=
w"'>&nbsp;&nbsp; report back all outstanding user data remaining in its out=
bound<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-GB style=
=3D'font-family:"Courier New"'>&nbsp;&nbsp; queue.&nbsp; The association is=
 automatically closed when the peer<o:p></o:p></span></p><p class=3DMsoPlai=
nText><span lang=3DEN-GB style=3D'font-family:"Courier New"'>&nbsp;&nbsp; e=
ndpoint becomes unreachable.<o:p></o:p></span></p><p class=3DMsoPlainText><=
span lang=3DEN-GB style=3D'font-family:"Courier New"'><o:p>&nbsp;</o:p></sp=
an></p><p class=3DMsoPlainText><span lang=3DEN-GB style=3D'font-family:"Cou=
rier New"'>&nbsp;&nbsp; The counter shall be reset each time a DATA chunk s=
ent to that peer<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3D=
EN-GB style=3D'font-family:"Courier New"'>&nbsp;&nbsp; endpoint is acknowle=
dged (by the reception of a SACK) or a HEARTBEAT<o:p></o:p></span></p><p cl=
ass=3DMsoPlainText><span lang=3DEN-GB style=3D'font-family:"Courier New"'>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ^^^^^^^^^^<o:p>=
</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-GB style=3D'font-f=
amily:"Courier New"'>&nbsp;&nbsp; ACK is received from the peer endpoint.<o=
:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-GB style=3D'fon=
t-family:"Courier New"'>&nbsp;&nbsp; ^^^^^^<o:p></o:p></span></p><p class=
=3DMsoPlainText><span lang=3DEN-GB style=3D'font-family:"Courier New"'><o:p=
>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-GB style=3D=
'font-size:10.0pt;font-family:"Arial","sans-serif"'>But it is also notewort=
hy (or is it?) that no RFC KEYWORDS are used in the formulations in this re=
spect.<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:1=
0.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p clas=
s=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-ser=
if"'>As there only seems to be down-sides of having HBs contribute to the a=
ssociation error counter (exception being idle situation) <o:p></o:p></span=
></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Aria=
l","sans-serif"'>I thus wonder whether the inclusion of HBs in Section 8.1.=
 of the RFC was intended primarily to cover the idle situation and/or <o:p>=
</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-=
family:"Arial","sans-serif"'>if the lack of KEYWORDS somehow was motivated =
by similar concerns as the above that was discussed at the time of the defi=
nition of the feature ?<o:p></o:p></span></p><p class=3DMsoNormal><span sty=
le=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p><=
/span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:=
"Arial","sans-serif"'>Along the same lines then I am wondering if it would =
be considered a breach of the RFC4960 if an SCTP implementation<o:p></o:p><=
/span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:=
"Arial","sans-serif"'>chose to have the association error counter be driven=
 only by the fate of data transmissions (except for the idle situation) ?<o=
:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fo=
nt-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoN=
ormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The=
 issue becomes particularly relevant when considering the aggressive HB pro=
bing mechanisms <o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'f=
ont-size:10.0pt;font-family:"Arial","sans-serif"'>of <a href=3D"http://tool=
s.ietf.org/html/draft-nishida-tsvwg-sctp-failover-05">http://tools.ietf.org=
/html/draft-nishida-tsvwg-sctp-failover-05</a>,<o:p></o:p></span></p><p cla=
ss=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-se=
rif"'>as there here can be very many Path monitoring HBs flowing on auxilia=
ry paths compared to data transmissions.<o:p></o:p></span></p><p class=3DMs=
oNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>B=
ut the issue is of generic nature and equally relevant for a basic RFC4960 =
SCTP implementation.<o:p></o:p></span></p><p class=3DMsoNormal><span style=
=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></s=
pan></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"A=
rial","sans-serif"'>All thoughts are most welcome ! </span><span style=3D'f=
ont-size:10.0pt;font-family:Wingdings'>J</span><span style=3D'font-size:10.=
0pt;font-family:"Arial","sans-serif"'><o:p></o:p></span></p><p class=3DMsoN=
ormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:=
p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0=
pt;font-family:"Arial","sans-serif"'>Thanks in advance.<o:p></o:p></span></=
p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial",=
"sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=
=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>BR, Karen <o:p></o:p=
></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-famil=
y:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><s=
pan style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>PS: It is r=
elevant to notice that path failure information based on unacknowledged HBs=
 are considerable less<o:p></o:p></span></p><p class=3DMsoNormal><span styl=
e=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>reliable than path =
failure input resulting from data T3 timeouts. This as fast retransmission =
procedures for data in many<o:p></o:p></span></p><p class=3DMsoNormal><span=
 style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>cases may cove=
r for single packet failures, whereas no mechanism is in place to cover for=
 single drops of HBs.<o:p></o:p></span></p><p class=3DMsoNormal><span style=
=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></s=
pan></p></div></body></html>=

--_000_CF340E42AED0874C81947E18863DE77B14460F578FEXMB03eutieto_--

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Thread-Topic: How a server can be able to send INIT to client
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From: "Sailabala Lenka" <sailabala.lenka@exfo.com>
To: <tsvwg@ietf.org>
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Subject: [tsvwg] How a server can be able to send INIT to client
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Hi All,

We are having 100 IPs in client side & 1 ip in server side.

Client is sending INIT to server and like that 100 SCTP associations are
happening.

But can anybody suggest how a server can be able to send INIT to client?
And how can we implement this scenario.

(We are using 2.6.9-55.ELsmp linux version)

=20

Please suggest

=20

Thanks,

Sailabala

=20


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<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>Hi All,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>We are having 100 IPs in client side &amp; 1 ip in server =
side.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>Client is sending INIT to server and like that 100 SCTP
associations are happening.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>But can anybody suggest how a server can be able to send =
INIT to
client? And how can we implement this scenario.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>(We are using 2.6.9-55.ELsmp linux =
version)<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>&nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>Please suggest<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>&nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>Thanks,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'>Sailabala<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:9.0pt;font-family:"Verdana","sans-serif";
color:#365F91'><o:p>&nbsp;</o:p></span></p>

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</body>

</html>

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From Michael.Tuexen@lurchi.franken.de  Thu May 24 01:39:28 2012
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To: Sailabala Lenka <sailabala.lenka@exfo.com>
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Subject: Re: [tsvwg] How a server can be able to send INIT to client
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On May 24, 2012, at 9:41 AM, Sailabala Lenka wrote:

> Hi All,
> We are having 100 IPs in client side & 1 ip in server side.
> Client is sending INIT to server and like that 100 SCTP associations =
are happening.
So the client initiate an SCTP association for each of its own IP =
addresses?
> But can anybody suggest how a server can be able to send INIT to =
client? And how can we implement this scenario.
SCTP handles INIT collisions. If you are using the socket API, you can
create a 1-to-many style socket, bind it to local addresses, call listen =
and
finally send a message to the peer. This way the endpoint initiates the =
association
but will also accept incoming ones.

Best regards
Michael
> (We are using 2.6.9-55.ELsmp linux version)
> =20
> Please suggest
> =20
> Thanks,
> Sailabala
> =20


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Karen:

In writing RFC2960 and later RFC4960 one of the fears I had
was that an overwhelming number of HB's would be being sent
by many endpoints. This could in theory flood the internet
with nothing but HBs. As such it was specifically left as
no MUST/SHOULD etc in the description of the HBs. This way =
implementations
could have a lot of lee-way in figuring out how they wanted to do HBs.

So yes, I think only counting the association error counter with HB's =
when
no data is in transit is a perfectly valid method. I would however =
continue
to count the path error counter on such HB's.

Note I have often thought we need two types of HB's. One that
is validating the association and one that is validating the endpoints =
"aliveness"
as well as its individual paths. Not that I have ever thought too deeply
on how to make this work. In the ideal case we would only have HB's =
between
two hosts over each of the paths, measuring the RTT and reach-ability of
the paths for all associations between a pair of hosts. Then when =
associations
were idle, we would have some sort of heartbeat so we could detect the =
peer
endpoint going away. One could even envision the HB's carrying a "list =
of ports"
in it that told the peer what associations were still up so that the =
association
HB's from say 100 associations would not be needed=85 of course all that =
is just
random thoughts early in the morning ;-)

R


On May 24, 2012, at 3:23 AM, <Karen.Nielsen@tieto.com> =
<Karen.Nielsen@tieto.com> wrote:

> Hi,
> =20
> The purpose of this email is to solicit views on the extent of which =
the fate of
> HBs should impact the association error counter of SCTP.
> =20
> The purpose of the association error counter in RFC4960 is to monitor =
the aliveness/failure of the peer endpoint.
> When speaking about aliveness is seems relevant to differentiate in =
between the aliveness of the peer endpoint
> and the =93aliveness of the association=94 in terms of the =93capability=
 of the association to forward data to the peer endpoint=94.
> Whereas the aliveness of the peer of course is required for the =
association to be of any use,
> then it is the latter that is of prime relevance for the SCTP Users =
and it is preferable if it is the latter type of =93aliveness=94
> that governs association closures and thus the exact handling of the =
association error counter.
> In this context it generally seems inappropriate for HBs to contribute =
to the association error counter.
> =20
> Well - in idle situations only HBs can drive the association error =
counter and here it makes good sense that their fate
> impact the association error counter (at least for the HBs that are =
transmitted on the current primary path).
> =20
> But in the situation where data is being transmitted then it is =
problematic and even distorting to have the fate
> of HBs transmitted over auxiliary paths impact the association error =
counter. By auxiliary paths I here mean paths
> different from data carrying path(s). Reasons being:
> =20
> Having successful HBs on some path reset the association error counter =
effectively results in that data may keep
> on being retransmitted in vain many more consecutive times that what =
the AMR threshold indicates, either because
> the intrinsic SCTP path selection is inadequate and not moves date =
(re)transmission to a working path or because
> HBs are let through (e.g., due to their potential small packet size or =
due to HW HB ack mechanism) whereas data consistently is lost.
> =20
> More importantly then if HBs that fail on auxiliary paths increment =
the association error counter, then in situations
> where there are many paths and thus potentially many HBs transmitted =
within a short period of time
> (even when randomly distributed there will or can be peaks), such =
failed
> HBs diminish the room available in the association error counter/AMR =
for failed data (re)transmissions.
> Effectively the failure of HBs on auxiliary back-up paths renders the =
association less robust
> to survive data failures on the current primary path.
> =20
> RFC4960 in section 8.1 stipulates that the fate of HBs shall =
contribute to the association error counter.
> (Exception being HBs transmitted for Path Verification, but that is a =
different topic):
> =20
>    An endpoint shall keep a counter on the total number of consecutive
>    retransmissions to its peer (this includes retransmissions to all =
the
>    destination transport addresses of the peer if it is multi-homed),
>    including unacknowledged HEARTBEAT chunks.
>   ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^If the value of this
>    counter exceeds the limit indicated in the protocol parameter
>    'Association.Max.Retrans', the endpoint shall consider the peer
>    endpoint unreachable and shall stop transmitting any more data to =
it
>    (and thus the association enters the CLOSED state).  In addition, =
the
>    endpoint MAY report the failure to the upper layer and optionally
>    report back all outstanding user data remaining in its outbound
>    queue.  The association is automatically closed when the peer
>    endpoint becomes unreachable.
> =20
>    The counter shall be reset each time a DATA chunk sent to that peer
>    endpoint is acknowledged (by the reception of a SACK) or a =
HEARTBEAT
>                                                              =
^^^^^^^^^^
>    ACK is received from the peer endpoint.
>    ^^^^^^
> =20
> But it is also noteworthy (or is it?) that no RFC KEYWORDS are used in =
the formulations in this respect.
> =20
> As there only seems to be down-sides of having HBs contribute to the =
association error counter (exception being idle situation)
> I thus wonder whether the inclusion of HBs in Section 8.1. of the RFC =
was intended primarily to cover the idle situation and/or
> if the lack of KEYWORDS somehow was motivated by similar concerns as =
the above that was discussed at the time of the definition of the =
feature ?
> =20
> Along the same lines then I am wondering if it would be considered a =
breach of the RFC4960 if an SCTP implementation
> chose to have the association error counter be driven only by the fate =
of data transmissions (except for the idle situation) ?
> =20
> The issue becomes particularly relevant when considering the =
aggressive HB probing mechanisms
> of http://tools.ietf.org/html/draft-nishida-tsvwg-sctp-failover-05,
> as there here can be very many Path monitoring HBs flowing on =
auxiliary paths compared to data transmissions.
> But the issue is of generic nature and equally relevant for a basic =
RFC4960 SCTP implementation.
> =20
> All thoughts are most welcome ! J
> =20
> Thanks in advance.
> =20
> BR, Karen
> =20
> PS: It is relevant to notice that path failure information based on =
unacknowledged HBs are considerable less
> reliable than path failure input resulting from data T3 timeouts. This =
as fast retransmission procedures for data in many
> cases may cover for single packet failures, whereas no mechanism is in =
place to cover for single drops of HBs.
> =20

-----
Randall Stewart
randall@lakerest.net





From Karen.Nielsen@tieto.com  Thu May 24 05:33:21 2012
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From: <Karen.Nielsen@tieto.com>
To: <randall@lakerest.net>
Date: Thu, 24 May 2012 15:33:17 +0300
Thread-Topic: [tsvwg] RFC4960 Section 8.1 and HB impacts on association error counter
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Subject: Re: [tsvwg] RFC4960 Section 8.1 and HB impacts on association error counter
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Hi Randy,

Thanks a lot !=20

This was exactly what I wanted to hear !

Yes, we shall continue to include HB failures =20
in the path error counters, but I think that we will=20
from now on exclude the feedback from HBs
in the association counter, except when there is no data.

BR, Karen

-----Original Message-----
From: Randy Stewart [mailto:randall@lakerest.net]=20
Sent: 24. maj 2012 12:48
To: Nielsen Karen
Cc: tsvwg@ietf.org
Subject: Re: [tsvwg] RFC4960 Section 8.1 and HB impacts on association erro=
r counter

Karen:

In writing RFC2960 and later RFC4960 one of the fears I had
was that an overwhelming number of HB's would be being sent
by many endpoints. This could in theory flood the internet
with nothing but HBs. As such it was specifically left as
no MUST/SHOULD etc in the description of the HBs. This way implementations
could have a lot of lee-way in figuring out how they wanted to do HBs.

So yes, I think only counting the association error counter with HB's when
no data is in transit is a perfectly valid method. I would however continue
to count the path error counter on such HB's.

Note I have often thought we need two types of HB's. One that
is validating the association and one that is validating the endpoints "ali=
veness"
as well as its individual paths. Not that I have ever thought too deeply
on how to make this work. In the ideal case we would only have HB's between
two hosts over each of the paths, measuring the RTT and reach-ability of
the paths for all associations between a pair of hosts. Then when associati=
ons
were idle, we would have some sort of heartbeat so we could detect the peer
endpoint going away. One could even envision the HB's carrying a "list of p=
orts"
in it that told the peer what associations were still up so that the associ=
ation
HB's from say 100 associations would not be needed... of course all that is=
 just
random thoughts early in the morning ;-)

R


On May 24, 2012, at 3:23 AM, <Karen.Nielsen@tieto.com> <Karen.Nielsen@tieto=
.com> wrote:

> Hi,
> =20
> The purpose of this email is to solicit views on the extent of which the =
fate of
> HBs should impact the association error counter of SCTP.
> =20
> The purpose of the association error counter in RFC4960 is to monitor the=
 aliveness/failure of the peer endpoint.
> When speaking about aliveness is seems relevant to differentiate in betwe=
en the aliveness of the peer endpoint
> and the "aliveness of the association" in terms of the "capability of the=
 association to forward data to the peer endpoint".
> Whereas the aliveness of the peer of course is required for the associati=
on to be of any use,
> then it is the latter that is of prime relevance for the SCTP Users and i=
t is preferable if it is the latter type of "aliveness"
> that governs association closures and thus the exact handling of the asso=
ciation error counter.
> In this context it generally seems inappropriate for HBs to contribute to=
 the association error counter.
> =20
> Well - in idle situations only HBs can drive the association error counte=
r and here it makes good sense that their fate
> impact the association error counter (at least for the HBs that are trans=
mitted on the current primary path).
> =20
> But in the situation where data is being transmitted then it is problemat=
ic and even distorting to have the fate
> of HBs transmitted over auxiliary paths impact the association error coun=
ter. By auxiliary paths I here mean paths
> different from data carrying path(s). Reasons being:
> =20
> Having successful HBs on some path reset the association error counter ef=
fectively results in that data may keep
> on being retransmitted in vain many more consecutive times that what the =
AMR threshold indicates, either because
> the intrinsic SCTP path selection is inadequate and not moves date (re)tr=
ansmission to a working path or because
> HBs are let through (e.g., due to their potential small packet size or du=
e to HW HB ack mechanism) whereas data consistently is lost.
> =20
> More importantly then if HBs that fail on auxiliary paths increment the a=
ssociation error counter, then in situations
> where there are many paths and thus potentially many HBs transmitted with=
in a short period of time
> (even when randomly distributed there will or can be peaks), such failed
> HBs diminish the room available in the association error counter/AMR for =
failed data (re)transmissions.
> Effectively the failure of HBs on auxiliary back-up paths renders the ass=
ociation less robust
> to survive data failures on the current primary path.
> =20
> RFC4960 in section 8.1 stipulates that the fate of HBs shall contribute t=
o the association error counter.
> (Exception being HBs transmitted for Path Verification, but that is a dif=
ferent topic):
> =20
>    An endpoint shall keep a counter on the total number of consecutive
>    retransmissions to its peer (this includes retransmissions to all the
>    destination transport addresses of the peer if it is multi-homed),
>    including unacknowledged HEARTBEAT chunks.
>   ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^If the value of this
>    counter exceeds the limit indicated in the protocol parameter
>    'Association.Max.Retrans', the endpoint shall consider the peer
>    endpoint unreachable and shall stop transmitting any more data to it
>    (and thus the association enters the CLOSED state).  In addition, the
>    endpoint MAY report the failure to the upper layer and optionally
>    report back all outstanding user data remaining in its outbound
>    queue.  The association is automatically closed when the peer
>    endpoint becomes unreachable.
> =20
>    The counter shall be reset each time a DATA chunk sent to that peer
>    endpoint is acknowledged (by the reception of a SACK) or a HEARTBEAT
>                                                              ^^^^^^^^^^
>    ACK is received from the peer endpoint.
>    ^^^^^^
> =20
> But it is also noteworthy (or is it?) that no RFC KEYWORDS are used in th=
e formulations in this respect.
> =20
> As there only seems to be down-sides of having HBs contribute to the asso=
ciation error counter (exception being idle situation)
> I thus wonder whether the inclusion of HBs in Section 8.1. of the RFC was=
 intended primarily to cover the idle situation and/or
> if the lack of KEYWORDS somehow was motivated by similar concerns as the =
above that was discussed at the time of the definition of the feature ?
> =20
> Along the same lines then I am wondering if it would be considered a brea=
ch of the RFC4960 if an SCTP implementation
> chose to have the association error counter be driven only by the fate of=
 data transmissions (except for the idle situation) ?
> =20
> The issue becomes particularly relevant when considering the aggressive H=
B probing mechanisms
> of http://tools.ietf.org/html/draft-nishida-tsvwg-sctp-failover-05,
> as there here can be very many Path monitoring HBs flowing on auxiliary p=
aths compared to data transmissions.
> But the issue is of generic nature and equally relevant for a basic RFC49=
60 SCTP implementation.
> =20
> All thoughts are most welcome ! J
> =20
> Thanks in advance.
> =20
> BR, Karen
> =20
> PS: It is relevant to notice that path failure information based on unack=
nowledged HBs are considerable less
> reliable than path failure input resulting from data T3 timeouts. This as=
 fast retransmission procedures for data in many
> cases may cover for single packet failures, whereas no mechanism is in pl=
ace to cover for single drops of HBs.
> =20

-----
Randall Stewart
randall@lakerest.net





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Subject: Re: [tsvwg] A possible improvement for draft-saldana-tsvwg-tcmtf-02:	asking for feedback
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From jsaldana@unizar.es  Fri May 25 01:03:57 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: "'Diego R. Lopez'" <diego@tid.es>, "Gorry Fairhurst" <gorry@erg.abdn.ac.uk>
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Date: Fri, 25 May 2012 10:03:41 +0200
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Subject: Re: [tsvwg] A possible improvement for draft-saldana-tsvwg-tcmtf-02:	asking for feedback
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Diego,

In my opinion, there are three questions related to your concern:

(1) The size of the "context" for each flow (the information for =
re-building the headers, which is shared by the origin and destination)
(2) Sending the "context" from the origin and destination
(3) Processing capacity issues (complexity, as you say) at the origin =
(compression) and destination (decompression)

The TCMTF draft has three layers: header compression, multiplexing and =
tunneling.

Questions (1) and (2) are problems related to the header compression =
standards (IPHC, ECRTP, ROHC), since they only involve the compression =
layer of the draft. Header compression avoids the sending of many =
fields, and also uses delta compression to reduce the number of bits =
required for other fields (typically sequence number fields which =
increase by one for each packet of the flow). Thus, the size of the =
"context" for each flow is more or less the header size, i.e. some tens =
of bytes. ROHC has some other things to save, namely the synchronization =
status, etc., but it is not a high amount of data. So question (1) is =
not difficult to solve.

Regarding question (2), the context is sent from the origin to the =
destination by the packets of the flow: the first packets are sent in =
its native form and, when the context has been established, header =
compression starts. Full headers are sent periodically, and also when =
synchronization problems appear. So this question is not a problem, in =
my opinion.

Question (3) is the most difficult to solve by now:=20
- origin side: a TCMTF multiplexer has to=20
	- compress the headers of a number of flows
	- store the packets in a queue and decide which ones are multiplexed at =
the end of each period
	- add the PPPMux headers
	- build the whole tunneled bundle packet (L2TP-PPPMux-compressed =
header)
	- send the bundle to the destination

- destination side: a TCMTF demultiplexer has to "open" the bundle, =
separate the packets, and decompress and rebuild them in their native =
form

I think the most complex part may be the header =
compression-decompression. So we have to look for studies which have =
deployed tests of this. By the moment, I have read these (by the way, =
Gorry is the author of the first one, so he may have something to say):

- C. S. Tye and D. Gorry Fairhurst, =E2=80=9CA review of IP packet =
compression techniques, in PGNet, (Liverpool, UK), June 2003.

- A. Couvreur, L. M. Le-Ny, A. Minaburo, G. Rubino, B. Sericola, and L. =
Toutain, =E2=80=9CPerformance analysis of a header compression protocol: =
The ROHC unidirectional mode,=E2=80=9D Telecommunication Systems, vol. =
31, no. 6, pp. 85=E2=80=9398, 2006.

- E. Ertekin and C. Christou, =E2=80=9CInternet Protocol Header =
Compression, Robust Header Compression, and Their Applicability in the =
Global Information Grid,=E2=80=9D IEEE Commun. Mag., vol. 42, 2004, pp. =
106=E2=80=9316.

- H. P. Sze, S. C. Liew, J. Y. B. Lee, and D. C. S. Yip, =E2=80=9CA =
multiplexing scheme for H.323 voice-over-IP applications,=E2=80=9D IEEE =
J. Select. Areas Commun., vol. 20, pp. 1360=E2=80=931368, Sept. 2002.

In this last paper, the authors did implement a software prototype in =
Linux, and the processing delay was below 1ms (by the way, using =
computers of year 2002):=20
"A software prototype is built to demonstrate the feasibility of our =
scheme and its compatibility with H.323. Our software runs under the =
Linux OS. (...) We also observe that the processing times of MUX/DEMUX =
due to header manipulation and packet transmission are negligible and =
generally below 1 ms".


Finally, it must be taken into account that in these scenarios (at least =
in games), we are dealing with servers with high processing capacity, =
since they have to receive all the inputs of the players movements, =
calculate the next status of the game, and send it to the players. And =
this is done with a very high frequency.

Sorry for this long message. Best regards and thank you for your =
feedback,

Jose

-----Mensaje original-----
De: Diego R. Lopez [mailto:diego@tid.es]=20
Enviado el: jueves, 24 de mayo de 2012 17:07
Para: Jose Saldana
CC: <tsvwg@ietf.org>
Asunto: Re: [tsvwg] A possible improvement for =
draft-saldana-tsvwg-tcmtf-02: asking for feedback

Hi,

On 16 May 2012, at 09:42 , Jose Saldana wrote:

> Hello all. I would like to discuss an idea that somebody suggested me =
in IETF83 Paris: the current draft considers one tunnel including a =
number of flows, but all these flows are of the same protocol, i.e.: one =
tunnel for multiplexing a number of VoIP flows using RTP; another tunnel =
for a number of UDP flows which belong to FPS games; another tunnel for =
a number of TCP flows of an MMORPG game, etc.
>
> But perhaps it could be possible to include in the same tunnel packets =
belonging to flows of different protocols. An access provider could use =
the same tunnel for services with similar delay constraints (e.g. a =
tunnel including VoIP and FPS games traffic). The advantage is clear: =
the bigger the number of flows, the bigger the bandwidth saving and the =
smaller the added multiplexing delays.
>
> However, on the other hand, maybe some problems would arise (signaling =
issues, etc.).
>
> So I would appreciate to receive some ideas about this:
> -          Do you think it could be useful?
> -          Some other scenarios in which different protocols (RTP, =
bare UDP, TCP) could be tunneled together?
> -          Problems that could arise?

My concern is about the size of the context information you'll have to =
send to reconstruct the multiplexed flows at the other end, in addition =
with the complexity of dealing with such a richer context. Do you have =
figures about the required size?
I guess that evaluating complexity could come later...

Be goode,

--
"Esta vez no fallaremos, Doctor Infierno"

Dr Diego R. Lopez
Telefonica I+D
http://people.tid.es/diego.lopez/

e-mail: diego@tid.es
Tel:    +34 913 129 041
Mobile: +34 682 051 091
-----------------------------------------


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Cc: Gorry Fairhurst <gorry@erg.abdn.ac.uk>, "<tsvwg@ietf.org>" <tsvwg@ietf.org>
Subject: Re: [tsvwg] A possible improvement for draft-saldana-tsvwg-tcmtf-02:	asking for feedback
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From jsaldana@unizar.es  Tue May 29 01:23:36 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: "'Diego R. Lopez'" <diego@tid.es>
References: <001901cd3337$65240650$2f6c12f0$@unizar.es> <C6E05599-BDEB-403D-B29F-B4BD90585704@tid.es> <000c01cd3a4c$ea11e820$be35b860$@unizar.es> <D86DA6CE-DB28-41A5-9253-C100E61B3B13@tid.es>
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Date: Tue, 29 May 2012 10:23:26 +0200
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Subject: Re: [tsvwg] A possible improvement for draft-saldana-tsvwg-tcmtf-02:	asking for feedback
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Hi Diego,

Currently we have not done any test regarding "aggressive" TCMTF (i.e. =
multiplexing flows of different protocols). All the tests we have done =
employ the same kind of traffic for all the flows of a single tunnel:  a =
tunnel of VoIP flows, a tunnel of FPS flows, etc.

The question here is that every compressed flow needs its own context. =
So I think the first question is that the "protocol used" would have to =
be included in the context shared by MUX and DEMUX. This is not a =
problem: a small amount of bytes would suffice to store the different =
possibilities: e.g. "IPHC compressing TCP/IPv4" or "ROHC compressing =
UDP/IPv6" or "ECRTP compressing RTP/UDP/IPv4".

Some signaling should be implemented when adding a new compressed flow =
to a tunnel; and it should include, of course, the protocol to compress. =
The advantage is that the three layers of TCMTF are "orthogonal", i.e. =
the Multiplexing and the Tunneling layers do not need to know anything =
about the protocol used by each Compressed packet they are carrying.

Some scenarios where this could be interesting (more cases would be =
appreciated):

- An office which simultaneously uses remote desktop and a centralized =
VoIP system, located at a server in the central office: the same tunnel =
(remote office-central office) could be used for carrying voice packets =
and remote desktop ones.

- An ISP may use a single tunnel for aggregating flows belonging to =
different kinds of games: FPS using UDP or MMORPG using TCP

Thank you for your feedback. Best regards,

Jose
PS: Be goode

-----Mensaje original-----
De: Diego R. Lopez [mailto:diego@tid.es]=20
Enviado el: viernes, 25 de mayo de 2012 16:03
Para: Jose Saldana
CC: Gorry Fairhurst; <tsvwg@ietf.org>
Asunto: Re: [tsvwg] A possible improvement for =
draft-saldana-tsvwg-tcmtf-02: asking for feedback

Hi Jose,

On 25 May 2012, at 10:03 , Jose Saldana wrote:
> In this last paper, the authors did implement a software prototype in =
Linux, and the processing delay was below 1ms (by the way, using =
computers of year 2002):
> "A software prototype is built to demonstrate the feasibility of our =
scheme and its compatibility with H.323. Our software runs under the =
Linux OS. (...) We also observe that the processing times of MUX/DEMUX =
due to header manipulation and packet transmission are negligible and =
generally below 1 ms".
>
> Finally, it must be taken into account that in these scenarios (at =
least in games), we are dealing with servers with high processing =
capacity, since they have to receive all the inputs of the players =
movements, calculate the next status of the game, and send it to the =
players. And this is done with a very high frequency.
>
> Sorry for this long message. Best regards and thank you for your=20
> feedback,

Great reply, thanks, but I may be misunderstood. My concern was =
precisely about the
*additional* complexity that we could incur if going beyond "classic =
TCMTF"
(multiplexing flows of equal nature) into "aggressive TCMTF" (attempting =
to multiplex all flows). Do you have any evidence or figures about that?

Be goode,


--
"Esta vez no fallaremos, Doctor Infierno"

Dr Diego R. Lopez
Telefonica I+D
http://people.tid.es/diego.lopez/

e-mail: diego@tid.es
Tel:    +34 913 129 041
Mobile: +34 682 051 091
-----------------------------------------


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From jsaldana@unizar.es  Tue May 29 03:19:31 2012
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From: "Jose Saldana" <jsaldana@unizar.es>
To: <tsvwg@ietf.org>
Date: Tue, 29 May 2012 12:19:26 +0200
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Subject: [tsvwg] draft-saldana-tsvwg-tcmtf for World of Warcraft: 60% bandwidth saving
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Hi all,

 

Our research group has just got accepted a paper for a conference next July.
It presents the tests of TCMTF (the "tunnels draft") for the traffic of
World of Warcraft (an online game with 10 million players). This game uses
TCP/IP, and client-to-server headers can be compressed from 40 to 8.72 bytes
average, so this traffic can get a 60% reduction. The cause of these good
results is that the game also generates a high number of TCP ACK packets
without payload (56%). For them, the saving is very high: the compression is
applied to headers, and these packets are "only header". Two other MMORPG
games are also analyzed, and the savings are 40% and 45%. The saving is
lower for server-to-client traffic (8.5%, 22%, 20%).

 

The number of packets per second can also be reduced: if the traffic of 100
players is multiplexed,  pps can be reduced from 900 to 12.

 

We have also used a MOS estimator (based on delay and jitter) in order to
grant that subjective quality is not harmed.

 

I attach two (small) PDF graphs showing the savings.

 

Since it has not already been published, I have put the paper in my web
page, just in case somebody wants to have a look: 

 

http://diec.unizar.es/~jsaldana/personal/sergei_SPECTS_2012_in_proc.pdf

 

 

Best regards,

 

Jose Saldana


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</o:shapelayout></xml><![endif]--></head><body lang=3DES link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Hi =
all,<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US>Our research group has just got =
accepted a paper for a conference next July. It presents the tests of =
TCMTF (the &#8220;tunnels draft&#8221;) for the traffic of <i>World of =
Warcraft</i> (an online game with 10 million players). This game uses =
TCP/IP, and client-to-server headers can be compressed from 40 to 8.72 =
bytes average, so this traffic can get a 60% reduction. The cause of =
these good results is that the game also generates a high number of TCP =
ACK packets without payload (56%). For them, the saving is very high: =
the compression is applied to headers, and these packets are &#8220;only =
header&#8221;. Two other MMORPG games are also analyzed, and the savings =
are 40% and 45%. The saving is lower for server-to-client traffic (8.5%, =
22%, 20%).<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>The number of packets per second can also be reduced: if =
the traffic of 100 players is multiplexed, &nbsp;pps can be reduced from =
900 to 12.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>We have also used a MOS estimator (based on delay and =
jitter) in order to grant that subjective quality is not =
harmed.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>I attach two (small) PDF graphs showing the =
savings.<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Since it has not already been published, I have put the =
paper in my web page, just in case somebody wants to have a look: =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>http://diec.unizar.es/~jsaldana/personal/sergei_SPECTS_2012_=
in_proc.pdf<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US>Best regards,<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";mso-fareast-la=
nguage:ES'>Jose Saldana</span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";mso-fareast-la=
nguage:ES'><o:p></o:p></span></p></div></body></html>
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From wwwrun@rfc-editor.org  Tue May 29 09:40:01 2012
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Date: Tue, 29 May 2012 09:39:24 -0700 (PDT)
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Subject: [tsvwg] RFC 6633 on Deprecation of ICMP Source Quench Messages
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 6633

        Title:      Deprecation of ICMP Source Quench 
                    Messages 
        Author:     F. Gont
        Status:     Standards Track
        Stream:     IETF
        Date:       May 2012
        Mailbox:    fgont@si6networks.com
        Pages:      8
        Characters: 16367
        Updates:    RFC0792, RFC1122, RFC1812

        I-D Tag:    draft-ietf-tsvwg-source-quench-06.txt

        URL:        http://www.rfc-editor.org/rfc/rfc6633.txt

This document formally deprecates the use of ICMP Source Quench
messages by transport protocols, formally updating RFC 792, RFC 1122,
and RFC 1812.  [STANDARDS-TRACK]

This document is a product of the Transport Area Working Group Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
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The RFC Editor Team
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Date: Thu, 31 May 2012 05:29:45 -0700 (PDT)
From: Chinmaya Dwibedy <ckdwibedy@yahoo.com>
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Subject: [tsvwg] =?utf-8?q?_A_query_on_SCTP_client=E2=80=99s_=28running_un?= =?utf-8?q?der_RHEL4=29_performance_in_terms_of_aggregate_throughput_rate?=
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--1819627952-792435336-1338467385=:57530
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=0A=0A=0AHi.=0A=C2=A0=0AWe have SCTP client program (C++) running under RHE=
L4 (using kernel version: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are u=
sing one-to-one socket (also called TCP-style sockets) using connect () sys=
tem call to setup an association with a server. Then it calls sctp_sendmsg =
() library function to send a message from a socket while using the followi=
ng advanced features of SCTP i.e., SCTP_UNORDERED for un-ordered delivery o=
f the message. The sctp_sendmsg () API takes a pointer to "sockaddr" as a p=
arameter indicating the address of the destination. Since our client applic=
ation knows well that, it will be sending the multiple messages to the same=
 configured destination address, it would become more efficient if we will =
connect the socket explicitly.=C2=A0 In summary, we can say that a SCTP cli=
ent can call connect only if that process uses the SCTP socket to communica=
te with exactly one peer. I mean,calling connect () on the SCTP socket once=
 simply records the
 specified IP address and port number as being the desired communications p=
artner and send/write call.Anything written to a connected SCTP socket is a=
utomatically sent to the protocol address (e.g., IP address and port) speci=
fied by connect. =0AAfter connect() system call,=C2=A0 there will be absolu=
tely no need of again passing socket address structure containing the desti=
nation IP address and port for each message in sctp_sendmsg(). It would be =
unnecessary memory copy between user space and kernel space, which is a per=
formance hit. Rather we should pass the destination address of the message =
and length of the destination address to zero in sctp_sendmsg () API. I bel=
ieve, with this change, it will improve performance (i.e., aggregate throug=
hput rate) when using a lot (32K) associations on a single system with 8 lo=
gical CPUs (speed of the processor is 2.13GHz) and 12GB RAM. Can someone pl=
ease=C2=A0validate my understanding and correct me if I am=C2=A0wrong? Than=
ks in advance for your response.=0ARegards,=0AChinmaya
--1819627952-792435336-1338467385=:57530
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<html><body><div style=3D"color:#000; background-color:#fff; font-family:ta=
homa, new york, times, serif;font-size:12pt"><div style=3D"RIGHT: auto"><SP=
AN style=3D"RIGHT: auto"><VAR id=3Dyui-ie-cursor></VAR><BR style=3D"RIGHT: =
auto" class=3Dyui-cursor></SPAN></div>
<div><BR></div>
<DIV style=3D"FONT-FAMILY: tahoma, new york, times, serif; FONT-SIZE: 12pt"=
>
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV style=3D"RIGHT: auto" dir=3Dltr><SPAN style=3D"FONT-FAMILY: 'Comic San=
s MS'; COLOR: black; FONT-SIZE: 10pt">Hi.</SPAN></DIV>
<DIV style=3D"RIGHT: auto" dir=3Dltr><SPAN style=3D"FONT-FAMILY: 'Comic San=
s MS'; COLOR: black; FONT-SIZE: 10pt" class=3Dms__id2279></SPAN><SPAN style=
=3D"FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt"></SPAN>&nb=
sp;</DIV>
<DIV style=3D"RIGHT: auto" id=3Dyiv78442574>
<DIV style=3D"RIGHT: auto">
<DIV style=3D"BACKGROUND-COLOR: #fff; FONT-FAMILY: tahoma, new york, times,=
 serif; COLOR: #000; FONT-SIZE: 12pt">
<DIV style=3D"LINE-HEIGHT: normal; MARGIN: 0in 0in 10pt; RIGHT: auto" class=
=3Dyiv78442574MsoNormal><SPAN style=3D"FONT-FAMILY: 'Comic Sans MS'; COLOR:=
 black; FONT-SIZE: 10pt">We have SCTP client program (C++) running under RH=
EL4 (using kernel version: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are =
using one-to-one socket (also called TCP-style sockets) using connect () sy=
stem call to setup an association with a server. Then it calls sctp_sendmsg=
 () library function to send a message from a socket while using the follow=
ing advanced features of SCTP i.e., SCTP_UNORDERED for un-ordered delivery =
of the message. The sctp_sendmsg () API takes a pointer to "sockaddr" as a =
parameter indicating the address of the destination. Since our client appli=
cation knows well that, it will be sending the multiple messages to the sam=
e configured destination address, it would become more efficient if we will=
 connect the socket explicitly.<SPAN>&nbsp; </SPAN>In summary, we can say
 that a SCTP client can call connect only if that process uses the SCTP soc=
ket to communicate with exactly one peer. I mean,</SPAN><SPAN style=3D"FONT=
-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt"> c</SPAN><SPAN sty=
le=3D"FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt">alling c=
onnect () on the SCTP socket once simply records the specified IP address a=
nd port number as being the desired communications partner and send/write c=
all.</SPAN><SPAN style=3D"COLOR: black; FONT-SIZE: 12pt"><FONT face=3DCalib=
ri> </FONT></SPAN><SPAN style=3D"FONT-FAMILY: 'Comic Sans MS'; COLOR: black=
; FONT-SIZE: 10pt">Anything written to a connected SCTP socket is automatic=
ally sent to the protocol address (e.g., IP address and port) specified by =
connect. </SPAN></DIV>
<DIV style=3D"LINE-HEIGHT: normal; MARGIN: 0in 0in 10pt; BACKGROUND: white;=
 RIGHT: auto" class=3Dyiv78442574MsoNormal><SPAN style=3D"FONT-FAMILY: 'Com=
ic Sans MS'; COLOR: black; FONT-SIZE: 10pt">After connect() system call,<SP=
AN>&nbsp; </SPAN>there will be absolutely no need of again passing socket a=
ddress structure containing the destination IP address and port for each me=
ssage in sctp_sendmsg(). It would be unnecessary memory copy between user s=
pace and kernel space, which is a performance hit. Rather we should pass th=
e destination address of the message and length of the destination address =
to zero in sctp_sendmsg () API. I believe, with this change, it will improv=
e performance (i.e., aggregate throughput rate) when using a lot (32K) asso=
ciations on a single system with 8 logical CPUs (speed of the processor is =
2.13GHz) and 12GB RAM. Can someone please&nbsp;validate my understanding an=
d correct me if I am&nbsp;wrong? Thanks in advance for your
 response.</SPAN><SPAN style=3D"FONT-FAMILY: 'serif'; FONT-SIZE: 12pt"></SP=
AN></DIV>
<DIV style=3D"LINE-HEIGHT: normal; MARGIN: 0in 0in 10pt; RIGHT: auto" class=
=3Dyiv78442574MsoNormal><SPAN style=3D"FONT-FAMILY: 'Comic Sans MS'; COLOR:=
 black; FONT-SIZE: 10pt">Regards,</SPAN><SPAN style=3D"FONT-FAMILY: 'Comic =
Sans MS'; FONT-SIZE: 10pt"></SPAN></DIV>
<DIV style=3D"LINE-HEIGHT: normal; MARGIN: 0in 0in 10pt; RIGHT: auto" class=
=3Dyiv78442574MsoNormal><SPAN style=3D"FONT-FAMILY: 'Comic Sans MS'; COLOR:=
 black; FONT-SIZE: 10pt">Chinmaya<VAR id=3Dyiv78442574yui-ie-cursor></VAR><=
/SPAN></DIV></DIV></DIV></DIV><BR><BR></DIV></DIV></div></body></html>
--1819627952-792435336-1338467385=:57530--

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Date: Thu, 31 May 2012 11:35:17 -0700 (PDT)
From: Chinmaya Dwibedy <ckdwibedy@yahoo.com>
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Subject: [tsvwg] =?utf-8?q?When_SCTP_server_is_restarted=2C_DATA_chunks_st?= =?utf-8?q?art_to_get_dropped_or_lost_after_sometime_at_SCTP_client?= =?utf-8?b?4oCZcyBlbmQu?=
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Hi, =0AWe have SCTP client program (C++) running under RHEL4 (using kernel =
version: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are using one-to-one s=
tyle socket using connect () system call to setup an association with a SCT=
P server (i.e., SUT of vendor). Then it calls sctp_sendmsg () library funct=
ion to send a message from a socket while using the following advanced feat=
ures of SCTP (i.e., SCTP_UNORDERED for un-ordered delivery of the message) =
and timetolive (TTL) is set to 1000 milliseconds for a given message. We ha=
ve typical defaults values for these RTO.Initial =3D 3000, RTO.Min =3D 1000=
, RTO.Max =3D 60000 milliseconds.The PR-SCTP extension is enabled in the ke=
rnel (by default).=0AWe are using lot (4000) SCTP associations on a single =
system, all are in ESTABLISHED state and data communication goes on fine wi=
thout any drop/loss of message. But when the peer (i.e., SCTP server) gets =
restarted, =0Aa)=C2=A0=C2=A0=C2=A0 All the SCTP clients receive INIT messag=
es (with same address to the association) from SCTP server. The INIT ACK is=
 being sent with the new InitTag and Verification Tag set to the InitTag re=
ceived in INIT for each association. =0Ab)=C2=A0=C2=A0=C2=A0 The four-way h=
andshake gets completed. I believe the existing association, including its =
current state, and the corresponding TCB does not get changed. =0Ac)=C2=A0=
=C2=A0=C2=A0 Afterward, all SCTP clients use DATA chunks to exchange inform=
ation with SCTP server.=0AThe trouble is that, after sometime (30-45 minute=
s), DATA chunks start to get dropped or lost and that goes on increasing as=
 the time advances.The data was not transmitted to the peer at least once. =
Note that, the time after which this problem is being encountered is also n=
ot consistent. I mean to say, sometime we noticed after 30 minutes, 45 minu=
tes and an hour. But when we lessen the number of SCTP associations to 2000=
, such problem does not appear. =C2=A0=C2=A0=C2=A0=0A=0A=0Aa)=C2=A0=C2=A0=
=C2=A0 Is there any chance that, that data messages time-out within the RTO=
 although both the values are same? As a result, the data will be skipped a=
nd no longer transmitted.=0Ab)=C2=A0=C2=A0=C2=A0 Does it indicate that the =
message could never be transmitted to the peer (e.g., flow control prevente=
d the message from being sent before its lifetime expired), so the peer nev=
er received it? Please note that, we are using 1 Gig Intel NIC. =0AThe prel=
iminary =C2=A0analysis says that, timetolive (in sctp_sendmsg() API) can be=
 set on a message independent of Partial Reliability.The timer (i.e., sinfo=
_timetolive) only runs while the message is in the kernel send buffer and h=
as not yet been put on the wire (for the first time). As soon as the messag=
e is put on the wire the first time, the timer is dropped. It is used to in=
dicate that a time based lifetime is being applied to the data.=C2=A0 It is=
 then a number of milliseconds for which the data is attempted to be transm=
itted. If that many milliseconds elapsed and the data has not been transmit=
ted, As a result, they move the abandoned list and are never retransmitted.=
 Please feel free to correct me if I am wrong.=0AShould we try with 0 ms ti=
metolive which indicates no timeout should occur on this message and is tre=
ated as reliable? Please suggest and also let me know if you need any addit=
ional information.=0AThanking you in advance for your response. =0ARegards,=
=0AChinmaya
---725201809-900650471-1338489317=:38384
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<html><body><div style=3D"color:#000; background-color:#fff; font-family:ta=
homa, new york, times, serif;font-size:12pt"><div style=3D"MARGIN: 0in 0in =
10pt; RIGHT: auto" class=3DMsoNormal><SPAN style=3D"LINE-HEIGHT: 115%; FONT=
-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt; mso-fareast-font-f=
amily: 'Times New Roman'; mso-bidi-font-family: 'Times New Roman'">Hi, <?xm=
l:namespace prefix =3D o ns =3D "urn:schemas-microsoft-com:office:office" /=
><o:p style=3D"RIGHT: auto"></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt; m=
so-fareast-font-family: 'Times New Roman'; mso-bidi-font-family: 'Times New=
 Roman'">We have SCTP client program (C++) running under RHEL4 (using kerne=
l version: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are using one-to-one=
 style socket using connect () system call to setup an association with a S=
CTP server (i.e., SUT of vendor). Then it calls sctp_sendmsg () library fun=
ction to send a message from a socket while using the following advanced fe=
atures of SCTP (i.e., SCTP_UNORDERED for un-ordered delivery of the message=
) and timetolive </SPAN><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Com=
ic Sans MS'; FONT-SIZE: 10pt">(TTL) </SPAN><SPAN style=3D"LINE-HEIGHT: 115%=
; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt; mso-fareast-=
font-family: 'Times New Roman'; mso-bidi-font-family: 'Times New Roman'">is
 set to 1000 milliseconds </SPAN><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAM=
ILY: 'Comic Sans MS'; FONT-SIZE: 10pt">for a given message. We have typical=
 defaults values for these RTO.Initial =3D 3000, RTO.Min =3D 1000, RTO.Max =
=3D 60000 </SPAN><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans=
 MS'; COLOR: black; FONT-SIZE: 10pt; mso-fareast-font-family: 'Times New Ro=
man'; mso-bidi-font-family: 'Times New Roman'">milliseconds.</SPAN><SPAN st=
yle=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt"> T=
he PR-SCTP extension is enabled in the kernel (by default).<o:p></o:p></SPA=
N></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt; m=
so-fareast-font-family: 'Times New Roman'; mso-bidi-font-family: 'Times New=
 Roman'">We are using lot (4000) SCTP associations on a single system, all =
are in ESTABLISHED state and data communication goes on fine without any dr=
op/loss of message. But when the peer (i.e., SCTP server) gets restarted, <=
o:p></o:p></SPAN></div>
<div style=3D"TEXT-INDENT: -0.25in; MARGIN: 0in 0in 0pt 0.25in; mso-add-spa=
ce: auto; mso-list: l0 level1 lfo1" class=3DMsoListParagraphCxSpFirst><SPAN=
 style=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FO=
NT-SIZE: 10pt; mso-fareast-font-family: 'Comic Sans MS'; mso-bidi-font-fami=
ly: 'Comic Sans MS'"><SPAN style=3D"mso-list: Ignore">a)<SPAN style=3D"FONT=
: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPAN><SPAN styl=
e=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SI=
ZE: 10pt; mso-fareast-font-family: 'Times New Roman'; mso-bidi-font-family:=
 'Times New Roman'">All the SCTP clients receive INIT messages (with same a=
ddress to the association) from SCTP server. The INIT ACK is being sent wit=
h the new InitTag and Verification Tag set to the InitTag received in INIT =
for each association. <o:p></o:p></SPAN></div>
<div style=3D"TEXT-INDENT: -0.25in; MARGIN: 0in 0in 0pt 0.25in; mso-add-spa=
ce: auto; mso-list: l0 level1 lfo1" class=3DMsoListParagraphCxSpMiddle><SPA=
N style=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; F=
ONT-SIZE: 10pt; mso-fareast-font-family: 'Comic Sans MS'; mso-bidi-font-fam=
ily: 'Comic Sans MS'"><SPAN style=3D"mso-list: Ignore">b)<SPAN style=3D"FON=
T: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPAN><SPAN sty=
le=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-S=
IZE: 10pt; mso-fareast-font-family: 'Times New Roman'; mso-bidi-font-family=
: 'Times New Roman'">The four-way handshake gets completed. I believe the e=
xisting association, including its current state, and the corresponding TCB=
 does not get changed. <o:p></o:p></SPAN></div>
<div style=3D"TEXT-INDENT: -0.25in; MARGIN: 0in 0in 10pt 0.25in; mso-add-sp=
ace: auto; mso-list: l0 level1 lfo1" class=3DMsoListParagraphCxSpLast><SPAN=
 style=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FO=
NT-SIZE: 10pt; mso-fareast-font-family: 'Comic Sans MS'; mso-bidi-font-fami=
ly: 'Comic Sans MS'"><SPAN style=3D"mso-list: Ignore">c)<SPAN style=3D"FONT=
: 7pt 'Times New Roman'">&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPAN><SPAN styl=
e=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SI=
ZE: 10pt; mso-fareast-font-family: 'Times New Roman'; mso-bidi-font-family:=
 'Times New Roman'">Afterward, all SCTP clients use DATA chunks to exchange=
 information with SCTP server.<o:p></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt; m=
so-fareast-font-family: 'Times New Roman'; mso-bidi-font-family: 'Times New=
 Roman'">The trouble is that, after sometime (30-45 minutes), DATA chunks s=
tart to get dropped or lost and that goes on increasing as the time advance=
s.</SPAN><FONT face=3DCalibri> </FONT><SPAN style=3D"LINE-HEIGHT: 115%; FON=
T-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt; mso-fareast-font-=
family: 'Times New Roman'; mso-bidi-font-family: 'Times New Roman'">The dat=
a was not transmitted to the peer at least once. Note that, the time after =
which this problem is being encountered is also not consistent. I mean to s=
ay, sometime we noticed after 30 minutes, 45 minutes and an hour. But when =
we lessen the number of SCTP associations to 2000, such problem does not ap=
pear. <SPAN style=3D"mso-spacerun: yes">&nbsp;</SPAN><SPAN style=3D"mso-spa=
cerun:
 yes">&nbsp;</SPAN><SPAN style=3D"mso-spacerun: yes">&nbsp;</SPAN><BR style=
=3D"mso-special-character: line-break"><BR style=3D"mso-special-character: =
line-break"><o:p></o:p></SPAN></div>
<div style=3D"TEXT-INDENT: -0.25in; MARGIN: 0in 0in 0pt 0.5in; mso-list: l1=
 level1 lfo2" class=3DMsoListParagraphCxSpFirst><SPAN style=3D"LINE-HEIGHT:=
 115%; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt; mso-fareast-font-fami=
ly: 'Comic Sans MS'; mso-bidi-font-family: 'Comic Sans MS'"><SPAN style=3D"=
mso-list: Ignore">a)<SPAN style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp=
;&nbsp; </SPAN></SPAN></SPAN><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMILY:=
 'Comic Sans MS'; FONT-SIZE: 10pt">Is there any chance that, that data mess=
ages time-out within the RTO although both the values are same? As a result=
, the data will be skipped and no longer transmitted.<o:p></o:p></SPAN></di=
v>
<div style=3D"TEXT-INDENT: -0.25in; MARGIN: 0in 0in 10pt 0.5in; mso-list: l=
1 level1 lfo2" class=3DMsoListParagraphCxSpLast><SPAN style=3D"LINE-HEIGHT:=
 115%; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt; mso-fareast-font-fami=
ly: 'Comic Sans MS'; mso-bidi-font-family: 'Comic Sans MS'"><SPAN style=3D"=
mso-list: Ignore">b)<SPAN style=3D"FONT: 7pt 'Times New Roman'">&nbsp;&nbsp=
;&nbsp; </SPAN></SPAN></SPAN><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMILY:=
 'Comic Sans MS'; FONT-SIZE: 10pt">Does it indicate that the message could =
never be transmitted to the peer (e.g., flow control prevented the message =
from being sent before its lifetime expired), so the peer never received it=
? Please note that, we are using 1 Gig Intel NIC. <o:p></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: black; FONT-SIZE: 10pt; m=
so-fareast-font-family: 'Times New Roman'; mso-bidi-font-family: 'Times New=
 Roman'">The preliminary <SPAN style=3D"mso-spacerun: yes">&nbsp;</SPAN>ana=
lysis says that, timetolive (in sctp_sendmsg() API) can be set on a message=
 independent of Partial Reliability.</SPAN><SPAN style=3D"LINE-HEIGHT: 115%=
; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt"> The timer (i.e., sinfo_ti=
metolive) only runs while the message is in the kernel send buffer and has =
not yet been put on the wire (for the first time). As soon as the message i=
s put on the wire the first time, the timer is dropped. It is used to indic=
ate that a time based lifetime is being applied to the data.&nbsp; It is th=
en a number of milliseconds for which the data is attempted to be transmitt=
ed. If that many milliseconds elapsed and the data has not been transmitted=
,
 As a result, they move the abandoned list and are never retransmitted. Ple=
ase feel free to correct me if I am wrong.<o:p></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt">Should we try w=
ith 0 ms timetolive which indicates no timeout should occur on this message=
 and is treated as reliable? Please suggest and also let me know if you nee=
d any additional information.<o:p></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt">Thanking you in=
 advance for your response. <o:p></o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt">Regards,<o:p></=
o:p></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN style=3D"LINE-H=
EIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; FONT-SIZE: 10pt">Chinmaya<o:p></=
o:p></SPAN></div>
<div style=3D"RIGHT: auto"><VAR id=3Dyui-ie-cursor></VAR>&nbsp;</div></div>=
</body></html>
---725201809-900650471-1338489317=:38384--

From cbenson@adax.com  Thu May 31 11:43:15 2012
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Date: Thu, 31 May 2012 11:43:13 -0700 (PDT)
From: Chris Benson <cbenson@adax.com>
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To: Chinmaya Dwibedy <ckdwibedy@yahoo.com>
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Subject: Re: [tsvwg] =?utf-8?q?When_SCTP_server_is_restarted=2C_DATA_chunks_st?= =?utf-8?q?art_to_get_dropped_or_lost_after_sometime_at_SCTP_client?= =?utf-8?b?4oCZcyBlbmQu?=
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Chinmaya,

Please forgive me if this is too basic and obvious a question.

Are you sure that your "application" (SCTP-user) at the client=20
side is consuming the incoming data AT LEAST AS FAST as it arrives?

If not, then you will see something like the behaviour you=20
describe. With thanks, from Chris

On Thu, 31 May 2012, Chinmaya Dwibedy wrote:

>>  Date: Thu, 31 May 2012 11:35:17 -0700 (PDT)
>>  From: Chinmaya Dwibedy <ckdwibedy@yahoo.com>
>>  To: "tsvwg@ietf.org" <tsvwg@ietf.org>
>>  Subject: [tsvwg] When SCTP server is restarted,
>>      DATA chunks start to get dropped or lost after sometime at SCTP cli=
ent
>>       end.
>> =20
>>  Hi,=20
We have SCTP client program (C++) running under RHEL4 (using kernel version=
: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are using one-to-one style so=
cket using connect () system call to setup an association with a SCTP serve=
r (i.e., SUT of vendor). Then it calls sctp_sendmsg () library function to =
send a message from a socket while using the following advanced features of=
 SCTP (i.e., SCTP_UNORDERED for un-ordered delivery of the message) and tim=
etolive (TTL) is set to 1000 milliseconds for a given message. We have typi=
cal defaults values for these RTO.Initial =3D 3000, RTO.Min =3D 1000, RTO.M=
ax =3D 60000 milliseconds.The PR-SCTP extension is enabled in the kernel (b=
y default).
We are using lot (4000) SCTP associations on a single system, all are in ES=
TABLISHED state and data communication goes on fine without any drop/loss o=
f message. But when the peer (i.e., SCTP server) gets restarted,=20
a)=C2=A0=C2=A0=C2=A0 All the SCTP clients receive INIT messages (with same =
address to the association) from SCTP server. The INIT ACK is being sent wi=
th the new InitTag and Verification Tag set to the InitTag received in INIT=
 for each association.=20
b)=C2=A0=C2=A0=C2=A0 The four-way handshake gets completed. I believe the e=
xisting association, including its current state, and the corresponding TCB=
 does not get changed.=20
c)=C2=A0=C2=A0=C2=A0 Afterward, all SCTP clients use DATA chunks to exchang=
e information with SCTP server.
The trouble is that, after sometime (30-45 minutes), DATA chunks start to g=
et dropped or lost and that goes on increasing as the time advances.The dat=
a was not transmitted to the peer at least once. Note that, the time after =
which this problem is being encountered is also not consistent. I mean to s=
ay, sometime we noticed after 30 minutes, 45 minutes and an hour. But when =
we lessen the number of SCTP associations to 2000, such problem does not ap=
pear. =C2=A0=C2=A0=C2=A0


a)=C2=A0=C2=A0=C2=A0 Is there any chance that, that data messages time-out =
within the RTO although both the values are same? As a result, the data wil=
l be skipped and no longer transmitted.
b)=C2=A0=C2=A0=C2=A0 Does it indicate that the message could never be trans=
mitted to the peer (e.g., flow control prevented the message from being sen=
t before its lifetime expired), so the peer never received it? Please note =
that, we are using 1 Gig Intel NIC.=20
The preliminary =C2=A0analysis says that, timetolive (in sctp_sendmsg() API=
) can be set on a message independent of Partial Reliability.The timer (i.e=
=2E, sinfo_timetolive) only runs while the message is in the kernel send bu=
ffer and has not yet been put on the wire (for the first time). As soon as =
the message is put on the wire the first time, the timer is dropped. It is =
used to indicate that a time based lifetime is being applied to the data.=
=C2=A0 It is then a number of milliseconds for which the data is attempted =
to be transmitted. If that many milliseconds elapsed and the data has not b=
een transmitted, As a result, they move the abandoned list and are never re=
transmitted. Please feel free to correct me if I am wrong.
Should we try with 0 ms timetolive which indicates no timeout should occur =
on this message and is treated as reliable? Please suggest and also let me =
know if you need any additional information.
Thanking you in advance for your response.=20
Regards,
Chinmaya
--808960-2099520786-1338489793=:23407--

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Date: Thu, 31 May 2012 12:20:08 -0700 (PDT)
From: Chinmaya Dwibedy <ckdwibedy@yahoo.com>
To: Chris Benson <cbenson@adax.com>
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Subject: Re: [tsvwg] =?utf-8?q?When_SCTP_server_is_restarted=2C_DATA_chunks_st?= =?utf-8?q?art_to_get_dropped_or_lost_after_sometime_at_SCTP_client?= =?utf-8?b?4oCZcyBlbmQu?=
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---725201809-312286038-1338492008=:97916
Content-Type: text/plain; charset=utf-8
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Hi Chris,=0AThanks for your response.=0AOur application consumes the messag=
e fast enough. Note that, we are using single Intel Xeon Quad-Core L5518 pr=
ocessor, 8 logical CPUs (speed of the processor is 2.13GHz) and 12GB RAM. M=
oreover there are no other apps running on besides our application. This pr=
oblem appears after SCTP server gets restarted. There is absolutely no pack=
ets drop if we won=E2=80=99t restart the server.=0A=C2=A0=0ADo you recommen=
d us to try with 0 ms timetolive which indicates no timeout should occur on=
 this message and is treated as reliable?=0A=C2=A0=0ARegards,=0AChinmaya=0A=
=0A=0A=0A________________________________=0AFrom: Chris Benson <cbenson@ada=
x.com>=0ATo: Chinmaya Dwibedy <ckdwibedy@yahoo.com> =0ACc: "tsvwg@ietf.org"=
 <tsvwg@ietf.org> =0ASent: Friday, June 1, 2012 12:13 AM=0ASubject: Re: [ts=
vwg] When SCTP server is restarted, DATA chunks start to get dropped or los=
t after sometime at SCTP client=E2=80=99s end.=0A=0AChinmaya,=0A=0APlease f=
orgive me if this is too basic and obvious a question.=0A=0AAre you sure th=
at your "application" (SCTP-user) at the client =0Aside is consuming the in=
coming data AT LEAST AS FAST as it arrives?=0A=0AIf not, then you will see =
something like the behaviour you =0Adescribe. With thanks, from Chris=0A=0A=
On Thu, 31 May 2012, Chinmaya Dwibedy wrote:=0A=0A>>=C2=A0 Date: Thu, 31 Ma=
y 2012 11:35:17 -0700 (PDT)=0A>>=C2=A0 From: Chinmaya Dwibedy <ckdwibedy@ya=
hoo.com>=0A>>=C2=A0 To: "tsvwg@ietf.org" <tsvwg@ietf.org>=0A>>=C2=A0 Subjec=
t: [tsvwg] When SCTP server is restarted,=0A>>=C2=A0 =C2=A0 =C2=A0 DATA chu=
nks start to get dropped or lost after sometime at SCTP client=0A>>=C2=A0 =
=C2=A0 =C2=A0 end.=0A>>=C2=A0 =0A>>=C2=A0 Hi, =0AWe have SCTP client progra=
m (C++) running under RHEL4 (using kernel version: 2.6.9-55ELsmp and lksctp=
-tools-1.0.8-1). We are using one-to-one style socket using connect () syst=
em call to setup an association with a SCTP server (i.e., SUT of vendor). T=
hen it calls sctp_sendmsg () library function to send a message from a sock=
et while using the following advanced features of SCTP (i.e., SCTP_UNORDERE=
D for un-ordered delivery of the message) and timetolive (TTL) is set to 10=
00 milliseconds for a given message. We have typical defaults values for th=
ese RTO.Initial =3D 3000, RTO.Min =3D 1000, RTO.Max =3D 60000 milliseconds.=
The PR-SCTP extension is enabled in the kernel (by default).=0AWe are using=
 lot (4000) SCTP associations on a single system, all are in ESTABLISHED st=
ate and data communication goes on fine without any drop/loss of message. B=
ut when the peer (i.e., SCTP server) gets restarted, =0Aa)=C2=A0=C2=A0=C2=
=A0 All the SCTP clients receive INIT messages (with same address to the as=
sociation) from SCTP server. The INIT ACK is being sent with the new InitTa=
g and Verification Tag set to the InitTag received in INIT for each associa=
tion. =0Ab)=C2=A0=C2=A0=C2=A0 The four-way handshake gets completed. I beli=
eve the existing association, including its current state, and the correspo=
nding TCB does not get changed. =0Ac)=C2=A0=C2=A0=C2=A0 Afterward, all SCTP=
 clients use DATA chunks to exchange information with SCTP server.=0AThe tr=
ouble is that, after sometime (30-45 minutes), DATA chunks start to get dro=
pped or lost and that goes on increasing as the time advances.The data was =
not transmitted to the peer at least once. Note that, the time after which =
this problem is being encountered is also not consistent. I mean to say, so=
metime we noticed after 30 minutes, 45 minutes and an hour. But when we les=
sen the number of SCTP associations to 2000, such problem does not appear. =
=C2=A0=C2=A0=C2=A0=0A=0A=0Aa)=C2=A0=C2=A0=C2=A0 Is there any chance that, t=
hat data messages time-out within the RTO although both the values are same=
? As a result, the data will be skipped and no longer transmitted.=0Ab)=C2=
=A0=C2=A0=C2=A0 Does it indicate that the message could never be transmitte=
d to the peer (e.g., flow control prevented the message from being sent bef=
ore its lifetime expired), so the peer never received it? Please note that,=
 we are using 1 Gig Intel NIC. =0AThe preliminary =C2=A0analysis says that,=
 timetolive (in sctp_sendmsg() API) can be set on a message independent of =
Partial Reliability.The timer (i.e., sinfo_timetolive) only runs while the =
message is in the kernel send buffer and has not yet been put on the wire (=
for the first time). As soon as the message is put on the wire the first ti=
me, the timer is dropped. It is used to indicate that a time based lifetime=
 is being applied to the data.=C2=A0 It is then a number of milliseconds fo=
r which the data is attempted to be transmitted. If that many milliseconds =
elapsed and the data has not been transmitted, As a result, they move the a=
bandoned list and are never retransmitted. Please feel free to correct me i=
f I am wrong.=0AShould we try with 0 ms timetolive which indicates no timeo=
ut should occur on this message and is treated as reliable? Please suggest =
and also let me know if you need any additional information.=0AThanking you=
 in advance for your response. =0ARegards,=0AChinmaya
---725201809-312286038-1338492008=:97916
Content-Type: text/html; charset=utf-8
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ta=
homa, new york, times, serif;font-size:12pt"><SPAN style=3D"RIGHT: auto">
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN class=3Dyiv1914=
637440yui320261335400116751279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMIL=
Y: 'Comic Sans MS'; COLOR: #002060; FONT-SIZE: 10pt; mso-fareast-font-famil=
y: 'Times New Roman'">Hi Chris,<?xml:namespace prefix =3D o ns =3D "urn:sch=
emas-microsoft-com:office:office" /><o:p></o:p></SPAN></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN class=3Dyiv1914=
637440yui320261335400116751279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMIL=
Y: 'Comic Sans MS'; COLOR: #002060; FONT-SIZE: 10pt; mso-fareast-font-famil=
y: 'Times New Roman'">Thanks for your response.<o:p></o:p></SPAN></SPAN></d=
iv>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN class=3Dyiv1914=
637440yui320261335400116751279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMIL=
Y: 'Comic Sans MS'; COLOR: #002060; FONT-SIZE: 10pt; mso-fareast-font-famil=
y: 'Times New Roman'">Our application consumes the message fast enough. Not=
e that, we are using single Intel Xeon Quad-Core L5518 processor, 8 logical=
 CPUs (speed of the processor is 2.13GHz) and 12GB RAM. Moreover there are =
no other apps running on besides our application. This problem appears afte=
r SCTP server gets restarted. There is absolutely no packets drop if we won=
=E2=80=99t restart the server.<o:p></o:p></SPAN></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN class=3Dyiv1914=
637440yui320261335400116751279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMIL=
Y: 'Comic Sans MS'; COLOR: #002060; FONT-SIZE: 10pt; mso-fareast-font-famil=
y: 'Times New Roman'"><o:p>&nbsp;</o:p></SPAN></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN class=3Dyiv1914=
637440yui320261335400116751279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMIL=
Y: 'Comic Sans MS'; COLOR: #002060; FONT-SIZE: 10pt; mso-fareast-font-famil=
y: 'Times New Roman'">Do you recommend us to try with 0 ms timetolive which=
 indicates no timeout should occur on this message and is treated as reliab=
le?<o:p></o:p></SPAN></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN class=3Dyiv1914=
637440yui320261335400116751279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMIL=
Y: 'Comic Sans MS'; COLOR: #002060; FONT-SIZE: 10pt; mso-fareast-font-famil=
y: 'Times New Roman'"><o:p>&nbsp;</o:p></SPAN></SPAN></div>
<div style=3D"MARGIN: 0in 0in 10pt" class=3DMsoNormal><SPAN class=3Dyiv1914=
637440yui320261335400116751279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMIL=
Y: 'Comic Sans MS'; COLOR: #002060; FONT-SIZE: 10pt; mso-fareast-font-famil=
y: 'Times New Roman'">Regards,<o:p></o:p></SPAN></SPAN></div>
<div style=3D"RIGHT: auto"><SPAN class=3Dyiv1914637440yui320261335400116751=
279><SPAN style=3D"LINE-HEIGHT: 115%; FONT-FAMILY: 'Comic Sans MS'; COLOR: =
#002060; FONT-SIZE: 10pt; mso-fareast-font-family: 'Times New Roman'; mso-b=
idi-font-family: 'Times New Roman'; mso-bidi-theme-font: minor-bidi; mso-an=
si-language: EN-US; mso-fareast-language: EN-US; mso-bidi-language: AR-SA">=
Chinmaya<VAR id=3Dyui-ie-cursor></VAR></SPAN></SPAN><BR style=3D"RIGHT: aut=
o" class=3Dyui-cursor></SPAN></div>
<div><BR></div>
<DIV style=3D"FONT-FAMILY: tahoma, new york, times, serif; FONT-SIZE: 12pt"=
>
<DIV style=3D"FONT-FAMILY: times new roman, new york, times, serif; FONT-SI=
ZE: 12pt">
<DIV dir=3Dltr><FONT size=3D2 face=3DArial>
<DIV style=3D"BORDER-BOTTOM: #ccc 1px solid; BORDER-LEFT: #ccc 1px solid; P=
ADDING-BOTTOM: 0px; LINE-HEIGHT: 0; MARGIN: 5px 0px; PADDING-LEFT: 0px; PAD=
DING-RIGHT: 0px; HEIGHT: 0px; FONT-SIZE: 0px; BORDER-TOP: #ccc 1px solid; B=
ORDER-RIGHT: #ccc 1px solid; PADDING-TOP: 0px" class=3Dhr contentEditable=
=3Dfalse readonly=3D"true"></DIV><B><SPAN style=3D"FONT-WEIGHT: bold">From:=
</SPAN></B> Chris Benson &lt;cbenson@adax.com&gt;<BR><B><SPAN style=3D"FONT=
-WEIGHT: bold">To:</SPAN></B> Chinmaya Dwibedy &lt;ckdwibedy@yahoo.com&gt; =
<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Cc:</SPAN></B> "tsvwg@ietf.org" &l=
t;tsvwg@ietf.org&gt; <BR><B><SPAN style=3D"FONT-WEIGHT: bold">Sent:</SPAN><=
/B> Friday, June 1, 2012 12:13 AM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">S=
ubject:</SPAN></B> Re: [tsvwg] When SCTP server is restarted, DATA chunks s=
tart to get dropped or lost after sometime at SCTP client=E2=80=99s end.<BR=
></FONT></DIV><BR>Chinmaya,<BR><BR>Please forgive me if this is too basic a=
nd obvious a
 question.<BR><BR>Are you sure that your "application" (SCTP-user) at the c=
lient <BR>side is consuming the incoming data AT LEAST AS FAST as it arrive=
s?<BR><BR>If not, then you will see something like the behaviour you <BR>de=
scribe. With thanks, from Chris<BR><BR>On Thu, 31 May 2012, Chinmaya Dwibed=
y wrote:<BR><BR>&gt;&gt;&nbsp; Date: Thu, 31 May 2012 11:35:17 -0700 (PDT)<=
BR>&gt;&gt;&nbsp; From: Chinmaya Dwibedy &lt;<A href=3D"mailto:ckdwibedy@ya=
hoo.com" ymailto=3D"mailto:ckdwibedy@yahoo.com">ckdwibedy@yahoo.com</A>&gt;=
<BR>&gt;&gt;&nbsp; To: "<A href=3D"mailto:tsvwg@ietf.org" ymailto=3D"mailto=
:tsvwg@ietf.org">tsvwg@ietf.org</A>" &lt;<A href=3D"mailto:tsvwg@ietf.org" =
ymailto=3D"mailto:tsvwg@ietf.org">tsvwg@ietf.org</A>&gt;<BR>&gt;&gt;&nbsp; =
Subject: [tsvwg] When SCTP server is restarted,<BR>&gt;&gt;&nbsp; &nbsp; &n=
bsp; DATA chunks start to get dropped or lost after sometime at SCTP client=
<BR>&gt;&gt;&nbsp; &nbsp; &nbsp; end.<BR>&gt;&gt;&nbsp; <BR>&gt;&gt;&nbsp; =
Hi,
 <BR>We have SCTP client program (C++) running under RHEL4 (using kernel ve=
rsion: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are using one-to-one sty=
le socket using connect () system call to setup an association with a SCTP =
server (i.e., SUT of vendor). Then it calls sctp_sendmsg () library functio=
n to send a message from a socket while using the following advanced featur=
es of SCTP (i.e., SCTP_UNORDERED for un-ordered delivery of the message) an=
d timetolive (TTL) is set to 1000 milliseconds for a given message. We have=
 typical defaults values for these RTO.Initial =3D 3000, RTO.Min =3D 1000, =
RTO.Max =3D 60000 milliseconds.The PR-SCTP extension is enabled in the kern=
el (by default).<BR>We are using lot (4000) SCTP associations on a single s=
ystem, all are in ESTABLISHED state and data communication goes on fine wit=
hout any drop/loss of message. But when the peer (i.e., SCTP server) gets r=
estarted, <BR>a)&nbsp;&nbsp;&nbsp; All the SCTP clients receive INIT
 messages (with same address to the association) from SCTP server. The INIT=
 ACK is being sent with the new InitTag and Verification Tag set to the Ini=
tTag received in INIT for each association. <BR>b)&nbsp;&nbsp;&nbsp; The fo=
ur-way handshake gets completed. I believe the existing association, includ=
ing its current state, and the corresponding TCB does not get changed. <BR>=
c)&nbsp;&nbsp;&nbsp; Afterward, all SCTP clients use DATA chunks to exchang=
e information with SCTP server.<BR>The trouble is that, after sometime (30-=
45 minutes), DATA chunks start to get dropped or lost and that goes on incr=
easing as the time advances.The data was not transmitted to the peer at lea=
st once. Note that, the time after which this problem is being encountered =
is also not consistent. I mean to say, sometime we noticed after 30 minutes=
, 45 minutes and an hour. But when we lessen the number of SCTP association=
s to 2000, such problem does not appear.
 &nbsp;&nbsp;&nbsp;<BR><BR><BR>a)&nbsp;&nbsp;&nbsp; Is there any chance tha=
t, that data messages time-out within the RTO although both the values are =
same? As a result, the data will be skipped and no longer transmitted.<BR>b=
)&nbsp;&nbsp;&nbsp; Does it indicate that the message could never be transm=
itted to the peer (e.g., flow control prevented the message from being sent=
 before its lifetime expired), so the peer never received it? Please note t=
hat, we are using 1 Gig Intel NIC. <BR>The preliminary &nbsp;analysis says =
that, timetolive (in sctp_sendmsg() API) can be set on a message independen=
t of Partial Reliability.The timer (i.e., sinfo_timetolive) only runs while=
 the message is in the kernel send buffer and has not yet been put on the w=
ire (for the first time). As soon as the message is put on the wire the fir=
st time, the timer is dropped. It is used to indicate that a time based lif=
etime is being applied to the data.&nbsp; It is then a number of
 milliseconds for which the data is attempted to be transmitted. If that ma=
ny milliseconds elapsed and the data has not been transmitted, As a result,=
 they move the abandoned list and are never retransmitted. Please feel free=
 to correct me if I am wrong.<BR>Should we try with 0 ms timetolive which i=
ndicates no timeout should occur on this message and is treated as reliable=
? Please suggest and also let me know if you need any additional informatio=
n.<BR>Thanking you in advance for your response. <BR>Regards,<BR>Chinmaya<B=
R><BR></DIV></DIV></div></body></html>
---725201809-312286038-1338492008=:97916--

From randall@lakerest.net  Thu May 31 16:35:56 2012
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Subject: Re: [tsvwg] =?windows-1252?q?A_query_on_SCTP_client=92s_=28running_un?= =?windows-1252?q?der_RHEL4=29_performance_in_terms_of_aggregate_throughpu?= =?windows-1252?q?t_rate?=
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Chinmaya:

If you are using a connected socket aka one-2-one model, then you can =
also use
send() and that should work.

The sctp_sendmsg() is also used by the one-2-many model socket, and =
there you need a
socket address. Some implementations allow you to place a NULL in the =
socket address
of a connected socket for sctp_sendmsg(), not sure if linux does. But =
this is all
implementation specific stuff.

R
On May 31, 2012, at 8:29 AM, Chinmaya Dwibedy wrote:

>=20
>=20
> Hi.
> =20
> We have SCTP client program (C++) running under RHEL4 (using kernel =
version: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are using =
one-to-one socket (also called TCP-style sockets) using connect () =
system call to setup an association with a server. Then it calls =
sctp_sendmsg () library function to send a message from a socket while =
using the following advanced features of SCTP i.e., SCTP_UNORDERED for =
un-ordered delivery of the message. The sctp_sendmsg () API takes a =
pointer to "sockaddr" as a parameter indicating the address of the =
destination. Since our client application knows well that, it will be =
sending the multiple messages to the same configured destination =
address, it would become more efficient if we will connect the socket =
explicitly.  In summary, we can say that a SCTP client can call connect =
only if that process uses the SCTP socket to communicate with exactly =
one peer. I mean, calling connect () on the SCTP socket once simply =
records the specified IP address and port number as being the desired =
communications partner and send/write call. Anything written to a =
connected SCTP socket is automatically sent to the protocol address =
(e.g., IP address and port) specified by connect.
> After connect() system call,  there will be absolutely no need of =
again passing socket address structure containing the destination IP =
address and port for each message in sctp_sendmsg(). It would be =
unnecessary memory copy between user space and kernel space, which is a =
performance hit. Rather we should pass the destination address of the =
message and length of the destination address to zero in sctp_sendmsg () =
API. I believe, with this change, it will improve performance (i.e., =
aggregate throughput rate) when using a lot (32K) associations on a =
single system with 8 logical CPUs (speed of the processor is 2.13GHz) =
and 12GB RAM. Can someone please validate my understanding and correct =
me if I am wrong? Thanks in advance for your response.
> Regards,
> Chinmaya
>=20
>=20

-----
Randall Stewart
randall@lakerest.net





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From: Randy Stewart <randall@lakerest.net>
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Date: Thu, 31 May 2012 19:35:41 -0400
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To: Chinmaya Dwibedy <ckdwibedy@yahoo.com>
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Subject: Re: [tsvwg] =?windows-1252?q?When_SCTP_server_is_restarted=2C_DATA_ch?= =?windows-1252?q?unks_start_to_get_dropped_or_lost_after_sometime_at_SCTP?= =?windows-1252?q?_client=92s_end=2E?=
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Chinmaya:

I would use that has a starting point. Don't put a time-to-expire on it =
and see if
you no longer see this behavior. If you do, then it sounds like a bug in =
the linux kernel
implementation

R
On May 31, 2012, at 3:20 PM, Chinmaya Dwibedy wrote:

> Hi Chris,
> Thanks for your response.
> Our application consumes the message fast enough. Note that, we are =
using single Intel Xeon Quad-Core L5518 processor, 8 logical CPUs (speed =
of the processor is 2.13GHz) and 12GB RAM. Moreover there are no other =
apps running on besides our application. This problem appears after SCTP =
server gets restarted. There is absolutely no packets drop if we won=92t =
restart the server.
> =20
> Do you recommend us to try with 0 ms timetolive which indicates no =
timeout should occur on this message and is treated as reliable?
> =20
> Regards,
> Chinmaya
>=20
> From: Chris Benson <cbenson@adax.com>
> To: Chinmaya Dwibedy <ckdwibedy@yahoo.com>=20
> Cc: "tsvwg@ietf.org" <tsvwg@ietf.org>=20
> Sent: Friday, June 1, 2012 12:13 AM
> Subject: Re: [tsvwg] When SCTP server is restarted, DATA chunks start =
to get dropped or lost after sometime at SCTP client=92s end.
>=20
> Chinmaya,
>=20
> Please forgive me if this is too basic and obvious a question.
>=20
> Are you sure that your "application" (SCTP-user) at the client=20
> side is consuming the incoming data AT LEAST AS FAST as it arrives?
>=20
> If not, then you will see something like the behaviour you=20
> describe. With thanks, from Chris
>=20
> On Thu, 31 May 2012, Chinmaya Dwibedy wrote:
>=20
> >>  Date: Thu, 31 May 2012 11:35:17 -0700 (PDT)
> >>  From: Chinmaya Dwibedy <ckdwibedy@yahoo.com>
> >>  To: "tsvwg@ietf.org" <tsvwg@ietf.org>
> >>  Subject: [tsvwg] When SCTP server is restarted,
> >>      DATA chunks start to get dropped or lost after sometime at =
SCTP client
> >>      end.
> >> =20
> >>  Hi,=20
> We have SCTP client program (C++) running under RHEL4 (using kernel =
version: 2.6.9-55ELsmp and lksctp-tools-1.0.8-1). We are using =
one-to-one style socket using connect () system call to setup an =
association with a SCTP server (i.e., SUT of vendor). Then it calls =
sctp_sendmsg () library function to send a message from a socket while =
using the following advanced features of SCTP (i.e., SCTP_UNORDERED for =
un-ordered delivery of the message) and timetolive (TTL) is set to 1000 =
milliseconds for a given message. We have typical defaults values for =
these RTO.Initial =3D 3000, RTO.Min =3D 1000, RTO.Max =3D 60000 =
milliseconds.The PR-SCTP extension is enabled in the kernel (by =
default).
> We are using lot (4000) SCTP associations on a single system, all are =
in ESTABLISHED state and data communication goes on fine without any =
drop/loss of message. But when the peer (i.e., SCTP server) gets =
restarted,=20
> a)    All the SCTP clients receive INIT messages (with same address to =
the association) from SCTP server. The INIT ACK is being sent with the =
new InitTag and Verification Tag set to the InitTag received in INIT for =
each association.=20
> b)    The four-way handshake gets completed. I believe the existing =
association, including its current state, and the corresponding TCB does =
not get changed.=20
> c)    Afterward, all SCTP clients use DATA chunks to exchange =
information with SCTP server.
> The trouble is that, after sometime (30-45 minutes), DATA chunks start =
to get dropped or lost and that goes on increasing as the time =
advances.The data was not transmitted to the peer at least once. Note =
that, the time after which this problem is being encountered is also not =
consistent. I mean to say, sometime we noticed after 30 minutes, 45 =
minutes and an hour. But when we lessen the number of SCTP associations =
to 2000, such problem does not appear.   =20
>=20
>=20
> a)    Is there any chance that, that data messages time-out within the =
RTO although both the values are same? As a result, the data will be =
skipped and no longer transmitted.
> b)    Does it indicate that the message could never be transmitted to =
the peer (e.g., flow control prevented the message from being sent =
before its lifetime expired), so the peer never received it? Please note =
that, we are using 1 Gig Intel NIC.=20
> The preliminary  analysis says that, timetolive (in sctp_sendmsg() =
API) can be set on a message independent of Partial Reliability.The =
timer (i.e., sinfo_timetolive) only runs while the message is in the =
kernel send buffer and has not yet been put on the wire (for the first =
time). As soon as the message is put on the wire the first time, the =
timer is dropped. It is used to indicate that a time based lifetime is =
being applied to the data.  It is then a number of milliseconds for =
which the data is attempted to be transmitted. If that many milliseconds =
elapsed and the data has not been transmitted, As a result, they move =
the abandoned list and are never retransmitted. Please feel free to =
correct me if I am wrong.
> Should we try with 0 ms timetolive which indicates no timeout should =
occur on this message and is treated as reliable? Please suggest and =
also let me know if you need any additional information.
> Thanking you in advance for your response.=20
> Regards,
> Chinmaya
>=20

-----
Randall Stewart
randall@lakerest.net





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Subject: [tsvwg] I-D Action: draft-ietf-tsvwg-port-use-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Transport Area Working Group Working =
Group of the IETF.

	Title           : Recommendations for Transport Port Uses
	Author(s)       : Joe Touch
	Filename        : draft-ietf-tsvwg-port-use-00.txt
	Pages           : 11
	Date            : 2012-05-30

   This document provides recommendations to application and service
   designers on how to use the transport protocol port number space to
   help in its preservation. **NOTE THAT THIS CURRENT VERSION IS
   LARGELY AN OUTLINE OF ISSUES**.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-tsvwg-port-use-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-ietf-tsvwg-port-use-00.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-tsvwg-port-use/

