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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 1 Dec 2006 06:59:05 +0100
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Rohan Mahy" <rohan.mahy@gmail.com>
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Hi,=20

In my case the UA DOES realize that the TCP connection has died, and
creates a new TCP connection.

So, since the UA already knows that a new TCP connection (and therefor a
new NAT binding) has to be created the XOR-MAPPED-ADDRESS is not needed.

My issue was whether the UA uses the XMA information for something else
than detecting that the binding has changed, but for outbound it seems
like it doesn't, so there is no outbound functionality "suffering" if
the UA doesn't get the XMA.

Regards,

Christer



> A NAT won't create a new TCP binding unless it sees a SYN. =20
> If the NAT reboots, the client will detect that the=20
> connection has died, and won't be able to send any traffic=20
> (SIP or STUN) until it opens a new connection explicitly.  An=20
> XOR-MAPPED-ADDRESS attribute over TCP is completely useless.
>=20
> thanks,
> -rohan
>=20
> On 11/30/06, Christer Holmberg (JO/LMF)=20
> <christer.holmberg@ericsson.com> wrote:
> >
> > Hi,
> >
> > >I still need to test this, but I suspect that the STUN=20
> response won't=20
> > >even arrive over TCP when the NAT has rebooted and uses a=20
> different=20
> > >source port for forwarding it. The server then cannot=20
> match it with=20
> > >an existing 5-tuple socket connection and will probably send a RST
> >
> > Maybe I missunderstood, but won't the STUN request create a new=20
> > binding, in case the NAT has rebooted?
> >
> > Regards,
> >
> > Christer
> >
> >
> >
> >
> > >
> > > Regards,
> > > Jeroen
> > >
> > > Christer Holmberg (JO/LMF) wrote:
> > > > Hi,
> > > >
> > > > A question for clarification:
> > > >
> > > > The STUN response provides the UA with certain useful=20
> information=20
> > > > (XOR-ADDRESS etc), which the UA can use to see whether=20
> a NAT has=20
> > > > rebooted etc. I assume that information is not needed when
> > > using TCP?
> > > >
> > > > Regards,
> > > >
> > > > Christer
> > > >
> > > >
> > > >
> > > >
> > > >> -----Original Message-----
> > > >> From: Francois Audet [mailto:audet@nortel.com]
> > > >> Sent: 30. marraskuuta 2006 19:45
> > > >> To: sip@ietf.org
> > > >> Cc: rohan@ekabal.com
> > > >> Subject: RE: [Sip] Re: issue with oubound and the used=20
> transport=20
> > > >> protocol
> > > >>
> > > >> Didn't we agree way back NOT to use PING, CRLF, etc., and use=20
> > > >> STUN instead?
> > > >>
> > > >> Why is this debate being re-opened?
> > > >>
> > > >> Remember: I favored the non-STUN TCP approach. However, if
> > > we can't
> > > >> stick to a decision, we'll debate this forever. It already
> > > feels like
> > > >> it.
> > > >>
> > > >>> -----Original Message-----
> > > >>> From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]
> > > >>> Sent: Thursday, November 30, 2006 08:02
> > > >>> To: Christer Holmberg (JO/LMF); Rohan Mahy;=20
> > > >>> Erkki.Koivusalo@nokia.com Cc: sip@ietf.org; rohan@ekabal.com
> > > >>> Subject: RE: [Sip] Re: issue with oubound and the=20
> used transport=20
> > > >>> protocol
> > > >>>
> > > >>> Having implemented STUN over TCP/TLS I can now say that
> > > it does add
> > > >>> significant complexity while providing little benefit=20
> over CRLF
> > > >>>
> > > >>> STUN allows one to detect NAT reboot, but a CRLF will also
> > > >> trigger a
> > > >>> connection reset in such a case. STUN breaks framing in
> > > >> case the other
> > > >>> side does not support it. It uses significantly more
> > > bandwidth than
> > > >>> CRLF. It requires a lot of text to describe.
> > > >>>
> > > >>> In summary, I support CRLF for TCP, and would suggest to do
> > > >> STUN only
> > > >>> for UDP (i.e. its original purpose). STUN for TCP is=20
> a bad idea
> > > >>>
> > > >>> Regards,
> > > >>> Jeroen
> > > >>>
> > > >>> PS Edge proxy choosing the mechanism: not a good idea
> > > >> either, better
> > > >>> to simply define 1 mechanism (STUN for UDP, CRLF for TCP)
> > > >>>
> > > >>> -----Oorspronkelijk bericht -----
> > > >>> Van: "Christer Holmberg (JO/LMF)"=20
> > > >>> <christer.holmberg@ericsson.com>
> > > >>> Aan: "Rohan Mahy" <rohan.mahy@gmail.com>;
> > > Erkki.Koivusalo@nokia.com
> > > >>> CC: sip@ietf.org; rohan@ekabal.com
> > > >>> Verzonden: 30-11-06 13:56
> > > >>> Onderwerp: RE: [Sip] Re: issue with oubound and the used
> > > transport
> > > >>> protocol
> > > >>>
> > > >>>
> > > >>> Hi,
> > > >>>
> > > >>> Why don't we let the edge proxy choose which keep-alive
> > > >> mechanism to
> > > >>> use?
> > > >>>
> > > >>> The outbound spec should now say (I haven't had time to
> > > >> check, but I
> > > >>> hope me and Rohan agreed to add it :) that the edge proxy
> > > >> can insert
> > > >>> the keepalive parameter in the Path header.
> > > >>>
> > > >>> So, we define a "CRLF" value for the parameter, and the
> > > >> edge proxy can
> > > >>> then use that value for TCP flows.
> > > >>>
> > > >>> Of course, if the UA is not mandated to implement all kind of=20
> > > >>> keep-alives, it will also have to be able to indicate to the=20
> > > >>> edge proxy which keep-alive mechanisms it support.
> > > >>>
> > > >>> Regards,
> > > >>>
> > > >>> Christer
> > > >>>
> > > >>>
> > > >>>
> > > >>>
> > > >>>
> > > >>>
> > > >>>
> > > >>>
> > > >>> _______________________________________________
> > > >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > >>> This list is for NEW development of the core SIP Protocol Use=20
> > > >>> sip-implementors@cs.columbia.edu for questions on current sip=20
> > > >>> Use sipping@ietf.org for new developments on the=20
> application of=20
> > > >>> sip
> > > >>>
> > > >>
> > > >> _______________________________________________
> > > >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > >> This list is for NEW development of the core SIP Protocol Use=20
> > > >> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > >> sipping@ietf.org for new developments on the application of sip
> > > >>
> > > >
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > >
> > >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1624440923==--



From sip-bounces@ietf.org Fri Dec 01 02:51:00 2006
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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Issue: Expiration of temp-gruu
To: Paul Kyzivat <pkyzivat@cisco.com>
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Paul,

My general concern is about the unpredictability of temporary GRUUs. If the
community is happy that this is good enough for use cases they have in mind,
I can live with it. I just wanted to see whether I was alone in having these
concerns.

Concerning the specific problem you raised at the start of this thread, one
solution might be, when returning a temporary GRUU in a 2xx to a REGISTER
request, to include an integer (as a Contact parameter) that indicates the
number of temporary GRUUs assigned. So if you refresh but too late, you
would see that only one is assigned and that older ones have been lost. This
would avoid subscription to the registration event package, although the
registration event package would bring additional benefits (alerting you
immediately without waiting for the next refresh).

John 

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> Sent: 30 November 2006 19:27
> To: Elwell, John
> Cc: SIP IETF
> Subject: Re: [Sip] Issue: Expiration of temp-gruu
> 
> 
> 
> Elwell, John wrote:
> > I guess some of my concerns about temporary GRUUs are 
> starting to creep back
> > again.
> 
> Is your renewed concern related to this particular issue I posted? It 
> doesn't seem to be.
> 
> Your concern seems to be that temporary gruus are, well, *temporary*.
> 
> > It is a question of how they are used. If a UA gives a 
> temporary GRUU
> > out as a contact on a dialog and expects it to be used only 
> during the
> > lifetime of that dialog, then refreshing the target if the 
> temporary GRUU is
> > lost would get around the problem in most cases. There will 
> of course be
> > race conditions, e.g., the UA sends out a target refresh 
> but the remote UA
> > has already sent a REFER request to the old target.
> 
> Yes. In most cases this won't be an issue. And I expect that many 
> implementations will simply assume that they stay registered 
> and don't 
> need to deal with the doing a target refresh because the gruu 
> has been lost.
> 
> > What concerns me more is if there are use cases where a 
> remote UA might want
> > to use a received temporary GRUU long after the dialog has 
> terminated. If
> > the local UA keeps refreshing its subscription correctly 
> and successfully,
> > then the temporary GRUU will remain valid. But if it is lost through
> > circumstances beyond the control of the local UA, then it 
> has no means to
> > inform the remote UA that it is no longer valid and that a different
> > temporary GRUU must be used.
> 
> Yes, this is the case. Frankly I find it hard to imagine a 
> case where a 
> temp-gruu, used for anonymity, is to be used for something long after 
> the dialog has terminated. I guess there could be cases, but perhaps 
> they aren't so interesting.
> 
> Any kind of scheme that allowed temp gruus to live arbitrarily beyond 
> the life of a registration would need some other mechanism to support 
> their longevity, and also would presumably need some kind of 
> revocation 
> mechanism. It would I think need state to be kept in the network.
> 
> There may be some use for such a thing. But I think it probably calls 
> for a different mechanism. One that probably isn't suitable 
> for everyday 
> anonymous calls. So I think supporting that mechanism can be 
> viewed as 
> an independent problem.
> 
> > One example given in gruu-11 is the use of a GRUU on a web 
> page. If a
> > temporary GRUU were to be used (presumably on a temporary 
> web page), similar
> > considerations would apply.
> 
> I guess it depends on how temporary the web page is.
> 
> > Another concern is if I want to give out a different 
> temporary GRUU to
> > different remote UAs, so that I can easily turn off 
> individual temporary
> > GRUUs if I no longer wish to receive requests from the 
> parties concerned.
> > When I first register I have only one temporary GRUU, so if 
> I want to give
> > out several temporary GRUUs I would need to refresh my 
> subscription several
> > times to obtain these GRUUs.
> > 
> > I just have an uneasy feeling that the temporary GRUU 
> mechanism might be
> > somewhat orthogonal to possible use cases, although I am 
> having difficulty
> > articulating particular real life use cases.
> 
> I think we already hashed these particular issues. There are some 
> unavoidable tradeoffs here.
> 
> 	Paul
> 
> > John 
> > 
> >> -----Original Message-----
> >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] 
> >> Sent: 29 November 2006 23:21
> >> To: SIP IETF
> >> Subject: [Sip] Issue: Expiration of temp-gruu
> >>
> >> After some study, I have encountered a potential difficulty in 
> >> determining when temp-gruus expire. After discussing this 
> >> with Jonathan 
> >> we decided to bring it to the list for discussion.
> >>
> >> As now defined, when you first register you may get a 
> >> temp-gruu, which 
> >> will remain valid as long as the registration does not 
> expire or be 
> >> removed. (Removal is a special case of expiration.) Each 
> time the UA 
> >> refreshes the registration it may get a new temp-gruu which 
> >> also remains 
> >> valid for the life of the registration. The UA can accumulate 
> >> all these 
> >> temp-gruus and use them as it sees fit until they expire.
> >>
> >> The problem is: how does the UA know when the registration 
> >> has ended and 
> >> the temp-gruus associated with it rendered invalid?
> >>
> >> This seems like a simple question, and in some cases it is 
> >> simple, but 
> >> not in all cases. The following are at least some of the ways 
> >> that the 
> >> registration can be ended, causing the invalidation of temp-gruus:
> >> 1) the UA that originally registered may explicitly deregister
> >> 2) the UA the originally registered may simply permit the
> >>     registration to expire without attempting to refresh it
> >> 3) the UA may attempt to refresh the registration, but be too slow,
> >>     so that the reregistration arrives after the prior one expires
> >> 4) some other UA may deregister all the registrations, including
> >>     the one by this UA. (When the UA decides it should refresh the
> >>     registration it will actually create a new one.)
> >> 5) the registrar may remove the registration administratively
> >> 6) the registrar may crash and restart, losing all its 
> registrations
> >>
> >> In each of these cases the previously assigned temp-gruus become 
> >> invalid. If the UA continues to use them, bad things will 
> >> happen. This 
> >> is true both for calls that are active when the 
> >> deregistration occurs, 
> >> and calls that are established later. What kinds of bad 
> things happen?
> >> - requests addressed to the gruu, both in-dialog and out-of-dialog
> >>    will be undeliverable. (For instance a reINVITE or BYE in an
> >>    existing call, or an out-of-dialog INVITE/Replaces for 
> a transfer.)
> >>
> >> - request originated by the UA using the invalid gruu will fail
> >>    immediately because the gruu is invalid.
> >>
> >> - if the UA subsequently reregisters, receives an incoming 
> call, and
> >>    uses one of the old temp-gruus as its contact, then the 
> call will
> >>    be established, but subsequent in- and out-of-dialog requests to
> >>    the contact will fail.
> >>
> >> So clearly it is important for the UA to stop using 
> >> temp-gruus that have 
> >> become invalid. How can it know to do that in all of the 
> above cases?
> >>
> >> 1) is straightforward. The UA knows it is deregistering, and should
> >>     discard any cached temp-gruus. If it has a dialog with 
> one of the
> >>     temp-gruus as a contact, then it should do a target refresh and
> >>     supply some working contact. (It really should do this *before*
> >>     deregistering, and for a pub-gruu as well.)
> >>
> >> 2) the UA should take all the actions mentioned for (1) at 
> or before
> >>     the expiration time. This should be done a bit early to guard
> >>     against clock skew problems.
> >>
> >> 3) in this case, it may be impossible for the UA to know whether
> >>     the REGISTER arrived in time to be a reREGISTER or if 
> it arrived
> >>     late and was treated as a new REGISTER after the expiration of
> >>     the old one. (The response to the REGISTER includes 
> nothing that
> >>     distinguishes the two cases.) About the best that can 
> be done in
> >>     this case is to check the time at which the response to the
> >>     REGISTER is received. If it falls after the expected expiration
> >>     time for the registration, act as if the old registration had
> >>     expired.
> >>
> >> 4) This is a major problem. Of course, regardless of GRUU, the UA
> >>     won't be able to receive incoming calls until it decides it is
> >>     time to refresh the registration. In addition, until the
> >>     registration reestablished, calls originated with one of its
> >>     existing gruus will fail. Once the registration is 
> reestablished,
> >>     it still won't know its old temp-gruus are invalid. The only
> >>     remedy for this is to subscribe to the "reg" event package,
> >>     and use the resulting notifications to identify when it has
> >>     been unregistered. If it does that, it can invalidate all its
> >>     old temp-gruus, and then presumably reregister.
> >>
> >> 5) this is functionally equivalent to (4), except that perhaps
> >>     attempts to reregister may fail.
> >>
> >> 6) this is also functionally equivalent to (4).
> >>
> >> The "reg" event does provide a fairly complete solution to 
> >> this problem, 
> >> for those that are willing and able to use it. It is not 
> >> however a 100% 
> >> reliable solution. Consider case (3), where the UA has a reg event 
> >> subscription. All is well if the registration expires and a 
> >> notification 
> >> is sent indicating that, and then the reregistration occurs and a 
> >> notification is sent about that. But if they happen very 
> closely in 
> >> time, it may be that only one notification is sent, reflecting the 
> >> result of the reregistration. In that case the UA will not be 
> >> aware that 
> >> it had been unregistered for a period of time.
> >>
> >> A *complete* solution would, IMO, involve changing REGISTER 
> >> so that it 
> >> returns an indication of whether this was a new registration, or a 
> >> refresh. This could take many forms. For instance the 
> >> timestamp of the 
> >> initial registration could always be returned with each 
> >> contact, as well 
> >> as the expiration time.
> >>
> >> The big question is whether this is a big enough issue to 
> require a 
> >> solution now, as part of GRUU, or if it can be ignored as 
> >> insignificant 
> >> or postponed to future work.
> >>
> >> I'd appreciate hearing what others think about this. At the 
> >> moment I am 
> >> leaning toward doing nothing, at least for now.
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >> _______________________________________________
> >> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >> This list is for NEW development of the core SIP Protocol
> >> Use sip-implementors@cs.columbia.edu for questions on current sip
> >> Use sipping@ietf.org for new developments on the application of sip
> >>
> > 
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Subject: [Sip] Re: [Sip-implementors] Forking INVITE SIP message - regarding
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Hi Raj,=0A=0AThis scenario can happen only in case of parallel forking. I d=
on't beleive that all the 2xx responces will reach the server at the same t=
ume. If it so then i think the server will forward the 2xx response on some=
 priority basis.=0A=0A/Sree=0A=0A----- Original Message ----=0AFrom: Raj <p=
onrajit@yahoo.com>=0ATo: SIP forum <sip@vovida.org>; sip@ietf.org; sip-impl=
ementors@cs.columbia.edu=0ASent: Tuesday, 28 November, 2006 10:47:02 PM=0AS=
ubject: [Sip-implementors] Forking  INVITE SIP message - regarding=0A=0AHel=
lo,=0A   =0A  Consider the following situation.=0A   =0A              Adam =
wants to call Bob, let=A2s say that Bob is registered at three locations, t=
hat is, at home, at head office and at his branch office. Hence, when Adam =
makes a call, INVITE request is forked by the proxy and it reaches all the =
three destinations. Hence, three 200 OK SIP response will be sent back to A=
dam. =0A   =0A              Consider the situation, when Bob=A2s friends an=
swer the call at all the three locations at the same time, what will happen=
? Either three media channel will be established between them or which two =
will be discarded.=0A   =0A              Can anyone please clear this?=0A  =
 =0A  With regards=0A  Raj.=0A=0A =0A---------------------------------=0AEv=
eryone is raving about the all-new Yahoo! Mail beta.=0A____________________=
___________________________=0ASip-implementors mailing list=0ASip-implement=
ors@cs.columbia.edu=0Ahttps://lists.cs.columbia.edu/cucslists/listinfo/sip-=
implementors=0A=0A=0A=0A=0A=0A=0A=0A=09=09=0A______________________________=
____________________________=0AYahoo! India Answers: Share what you know. L=
earn something new=0Ahttp://in.answers.yahoo.com/
--0-1346572428-1164777003=:75290
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<html><head><style type=3D"text/css"><!-- DIV {margin:0px;} --></style></he=
ad><body><div style=3D"font-family:times new roman, new york, times, serif;=
font-size:12pt"><div style=3D"font-family: times new roman,new york,times,s=
erif; font-size: 12pt;">Hi Raj,<br><br>This scenario can happen only in cas=
e of parallel forking. I don't beleive that all the 2xx responces will reac=
h the server at the same tume. If it so then i think the server will forwar=
d the 2xx response on some priority basis.<br><br>/Sree<br><br><div style=
=3D"font-family: times new roman,new york,times,serif; font-size: 12pt;">--=
--- Original Message ----<br>From: Raj &lt;ponrajit@yahoo.com&gt;<br>To: SI=
P forum &lt;sip@vovida.org&gt;; sip@ietf.org; sip-implementors@cs.columbia.=
edu<br>Sent: Tuesday, 28 November, 2006 10:47:02 PM<br>Subject: [Sip-implem=
entors] Forking  INVITE SIP message - regarding<br><br><div>Hello,<br>&nbsp=
;&nbsp; <br>&nbsp;&nbsp;Consider the following situation.<br>&nbsp;&nbsp;
 <br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;Adam wants to call Bob, let=A2s say that Bob is registered at=
 three locations, that is, at home, at head office and at his branch office=
. Hence, when Adam makes a call, INVITE request is forked by the proxy and =
it reaches all the three destinations. Hence, three 200 OK SIP response wil=
l be sent back to Adam. <br>&nbsp;&nbsp; <br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Consider the situatio=
n, when Bob=A2s friends answer the call at all the three locations at the s=
ame time, what will happen? Either three media channel will be established =
between them or which two will be discarded.<br>&nbsp;&nbsp; <br>&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;C=
an anyone please clear this?<br>&nbsp;&nbsp; <br>&nbsp;&nbsp;With regards<b=
r>&nbsp;&nbsp;Raj.<br><br> <br>---------------------------------<br>Everyon=
e is raving
 about the all-new Yahoo! Mail beta.<br>___________________________________=
____________<br>Sip-implementors mailing list<br>Sip-implementors@cs.columb=
ia.edu<br><a target=3D"_blank" href=3D"https://lists.cs.columbia.edu/cucsli=
sts/listinfo/sip-implementors">https://lists.cs.columbia.edu/cucslists/list=
info/sip-implementors</a><br></div></div><br></div></div><br>=0A=09=0A=0A=
=09=0A=09=09<hr size=3D1></hr> =0AFind out what India is talking about on  =
- <a href=3D"http://us.rd.yahoo.com/mail/in/yanswers/*http://in.answers.yah=
oo.com/">Yahoo! Answers India</a> <BR> =0ASend FREE SMS to your friend's mo=
bile from Yahoo! Messenger Version 8. <a href=3D"http://us.rd.yahoo.com/mai=
l/in/messengertagline/*http://in.messenger.yahoo.com">Get it NOW</a></body>=
</html>
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1439008752==--




From sip-bounces@ietf.org Fri Dec 01 06:59:56 2006
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Subject: [Sip] Re: [Sip-implementors] Signalling and media path of SIP
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Hi Raj,
For first problem, I don't think there is any protocol to find out the 
optimal Media Path. The SDP contact address which they share in INVITE 
and 200 OK will be used for Media transmission.

For second, If you want to send BYE in the same path where your previous 
packets traversed you should have enabled Record-Route at the proxy 
servers. Else BYE will go directly to the contact address which  came in 
200 OK of INVITE(at UAC)  and ACK(at UAS).

Regards
Vinod Kagalkar
Senior Project Leader
MultiTech Software Systems
Bangalore
 

Raj wrote:

>Hello,
>   
>                Recently I read from an article that signalling path of SIP is different from the path of media. When two end points want to communicate using SIP, the signalling takes place according to the operation of SIP, but once the signalling is over and when the media channel has to be established, which protocol is used to find the optimal media path?
>   
>               Moreover, when the session has to be terminated, SIP enabled end point has to send the BYE request, in this case, whether it takes the same path of INVITE request(which initiated the session) or  a different one. I am considering the stateless proxies to be in the network.
>   
>  with regards
>  Raj.
>
> 
>---------------------------------
>Everyone is raving about the all-new Yahoo! Mail beta.
>_______________________________________________
>Sip-implementors mailing list
>Sip-implementors@cs.columbia.edu
>https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>
>
>  
>

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<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Comic Sans MS">Hi Raj,<br>
For first problem, I don't think there is any protocol to find out the
optimal Media Path. The SDP contact address which they share in INVITE
and 200 OK will be used for Media transmission.<br>
<br>
For second, If you want to send BYE in the same path where your
previous packets traversed you should have enabled Record-Route at the
proxy servers. Else BYE will go directly to the contact address which&nbsp;
came in 200 OK of INVITE(at UAC)&nbsp; and ACK(at UAS).<br>
<br>
Regards<br>
Vinod Kagalkar<br>
Senior Project Leader<br>
MultiTech Software Systems<br>
Bangalore<br>
&nbsp;<br>
</font></font><br>
Raj wrote:<br>
<blockquote cite="mid326330.41992.qm@web52913.mail.yahoo.com"
 type="cite">
  <pre wrap="">Hello,
   
                Recently I read from an article that signalling path of SIP is different from the path of media. When two end points want to communicate using SIP, the signalling takes place according to the operation of SIP, but once the signalling is over and when the media channel has to be established, which protocol is used to find the optimal media path?
   
               Moreover, when the session has to be terminated, SIP enabled end point has to send the BYE request, in this case, whether it takes the same path of INVITE request(which initiated the session) or  a different one. I am considering the stateless proxies to be in the network.
   
  with regards
  Raj.

 
---------------------------------
Everyone is raving about the all-new Yahoo! Mail beta.
_______________________________________________
Sip-implementors mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Sip-implementors@cs.columbia.edu">Sip-implementors@cs.columbia.edu</a>
<a class="moz-txt-link-freetext" href="https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors">https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors</a>


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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0220952633==--




From sip-bounces@ietf.org Fri Dec 01 07:01:37 2006
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Subject: [Sip] Re: [Sip-implementors] Signalling time - regarding
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Latency depends on a number of factors Raj,

Network speed,
Message size,
Server processing

We are running a broardband based carrier network here with an IMS application server.
A typical invite costs about 25kb and takes ~500ms to call setup.

Regards,

Warren Crossing.

Raj wrote:
> Hello to all,
>    
>                 I read an article about Skype and it was stated that it takes 2 to 3 seconds for session establishment. Can anyone tell me how long will it take for SIP for session establishment?
>    
>   with regards
>   Raj
> 
>  
> ---------------------------------
> Want to start your own business? Learn how on Yahoo! Small Business.
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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Subject: [Sip] Re: [Sip-implementors] Signalling and media path of SIP
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SDP (session description protocol) information carried in the message 
body of the SIP messages(usually INVITE and its 200 OK response) 
exchanged b/w the two endpoints, explains the terminal capabilities of 
an endpoint to its peer. This SDP also includes the IP address and UDP 
port that will be used for the media packet flow. So once the signaling 
is through, flow of media packets is end to end.

Once the dialog is established, two endpoints communicate directly 
through the URI received in the contact headers. Hence, the bye request 
is sent directly end to end bypassing any proxy.

Raj wrote:
> Hello,
>    
>                 Recently I read from an article that signalling path of SIP is different from the path of media. When two end points want to communicate using SIP, the signalling takes place according to the operation of SIP, but once the signalling is over and when the media channel has to be established, which protocol is used to find the optimal media path?
>    
>                Moreover, when the session has to be terminated, SIP enabled end point has to send the BYE request, in this case, whether it takes the same path of INVITE request(which initiated the session) or  a different one. I am considering the stateless proxies to be in the network.
>    
>   with regards
>   Raj.
>
>  
> ---------------------------------
> Everyone is raving about the all-new Yahoo! Mail beta.
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>
>   


-- 
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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Hi,

You asked me to clarify my CRLF proposal, so I try to do it here.

I have been exploring ways of rescoping drafts related to Outbound
and simplifying the mechanisms within them since the limitations
of using UDP flows became apparent in this thread. I could vote
for a number of different approaches and would be happy the WG
to agree with at least one of them. Those approaches in my=20
preference order are:

1. Outbound draft to re-scope itself to "Outbound TCP connection"
   draft without any support to UDP, STUN or multiple registrations.
   Multiple registrations and mid-dialog failover (using either TCP
   or UDP) to be specified within another draft.

   - This proposal as such did not raise any discussion=20
     on the list, probably because it would be a dramatic
     change at a quite late stage of the draft.
   - Remarkable changes needed (but I already outlined the
     "Outbound TCP connection" draft within my earlier mail :).
     http://www1.ietf.org/mail-archive/web/sip/current/msg17495.html

2. Outbound draft to replace STUN keepalive over TCP with CRLF
   keepalive over TCP. (No impact to UDP support of Outbound)

   - This option is equivalent to your formulation, which I
     can very much agree on:

> Clearly you are suggesting that we make CRLF mandatory to implement
> for TCP.  There would be no longer any need to do STUN over TCP and
> therefore no need implement code to deal with stream-based demuxing.
> The text about sigcomp over TCP and STUN over TLS over TCP could
> also go away.

   - Some WG members already expressed their support for this
     option
   - Minor changes needed to the draft

3. Outbound draft to introduce a third optional keepalive method
   for TCP: CRLF (in addition to STUN and TCP keepalives)

   - This option would be the _last resort_ to get at least some
     benefit for TCP-based UA implementation, even if it would
     not help the proxies. Proxies would still have to support
     STUN over TCP.=20
   - Minimal changes needed to the draft


> To clarify are you suggesting that an edge proxy can implement
> outbound without support for UDP flow tokens, without support for STUN
> keepalives over UDP, both, or neither?

My proposals were born "UA centric" as I am from the UA vendor camp.

Anyway if Outbound will specify both UDP and TCP as transport options,
I would believe the proxies still have to support both since the
selection of the transport is basically an UA decision (even if
assisted by DNS SRV) as it depends on if the proxy URI configured
to the UA has a transport parameter. If the proxy would not support=20
UDP & STUN, how could it tell it to the UA ? Outbound would at=20
least need a mechanism for that e.g. a specific response code,=20
option tag or something else. Or is this yet another thing where
Outbound would rely on the UA and DNS configuration to tell the
UA to avoid using UDP ?

Considering the option to drop UDP & STUN support from Outbound
draft: I would agree that combination of UDP and STUN keepalives
does not seem to provide a practical way for NAT traversal. Keeping
UDP flows alive with STUN seems not to add value in comparison
to persistent TCP flows and I could agree dropping UDP flows from
the draft. The only problem I see here that Outbound covers also
the multiregistration mechanism, which could still be useful with
UDP e.g. in cellular access context without NAT/FWs. That is why
I have been proposing re-scoping of the drafts:

- Flow keepalive mechanisms to be specified only for TCP (one draft)
- Protocol agnostic multiregistration mechanism with mid-dialog
  failover support (another draft)

Regards,

Erkki

> -----Original Message-----
> From: ext Rohan Mahy [mailto:rohan.mahy@gmail.com]
> Sent: 29 November, 2006 20:30
> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
> Cc: sip@ietf.org; rohan@ekabal.com
> Subject: Re: [Sip] Re: issue with oubound and the used transport
> protocol
>=20
>=20
> Hi Erkki,
>=20
> Clearly you are suggesting that we make CRLF mandatory to implement
> for TCP.  There would be no longer any need to do STUN over TCP and
> therefore no need implement code to deal with stream-based demuxing.
>   The text about sigcomp over TCP and STUN over TLS over TCP could
> also go away.
>=20
> To clarify are you suggesting that an edge proxy can implement
> outbound without support for UDP flow tokens, without support for STUN
> keepalives over UDP, both, or neither?
>=20
> thanks,
> -rohan
>=20
> On 11/29/06, Erkki.Koivusalo@nokia.com=20
> <Erkki.Koivusalo@nokia.com> wrote:
> >
> > Hi,
> >
> > >about STUN keepalive usage and discovery, since double=20
> CRLF is fully
> > >backwards compatible with even RFC 2543 implementations.  I am
> > >bringing this up on the list because I hope the WG can look back on
> > >this and not make the mistake of snowballing complexity to preserve
> > >UDP parity again in the future.
> >
> > Rohan is making a valid point here and I believe in Outbound there
> > is still room for change to cater for CRLF keepalives as an option
> > to STUN keepalives over TCP.
> >
> > I am calling the WG to look back on the keepalive decision made
> > earlier for TCP and proposing that Outbound should also cover
> > CRLF keepalive option.
> >
> > Reasons for this proposal:
> >
> > >The problem I see is that we have two separate communities=20
> using the
> > >outbound draft:
> > >
> > >1) folks that want to get through NATs and Firewalls in the public
> > >Internet at large - they want to use TCP and TLS with=20
> double CRLF for
> > >keepalives. One "flow" is always one connection.  The milestone for
> > >this would read "Make TCP and TLS work in the Internet".
> >
> > I agree with the intents of this community. However this is not what
> > the current Outbound provides. Outbound specifies two options for
> > keepalives for reliable connections: TCP keepalive and STUN.
> >
> > These options were agreed in IETF 65, a cite from the minutes:
> >
> > Proposal:
> >    Do STUN for TCP and UDP
> >    Do CRLF for TCP and STUN for UDP
> >    TCP KeepAlive if supported, 1 or 2 otherwise
> >
> > Resolutions:
> >    OS TCP KeepAlive should be used if the client supports it.
> >    Need something more because OS TCP KeepAlive is not supported in
> > every
> >    client OS.
> >    Consensus to use STUN for UDP
> >    Consensus to use STUN for TCP, with use of OS TCP keep=20
> alive above.
> >
> > In that meeting the basic assumption was that every Outbound
> > compliant client should implement both UDP and TCP even if that
> > is not explicitely mentioned in the draft. However latest -06
> > version now makes that debatable by stating:
> >
> >    Use of outbound with UDP flows typically prevents messages larger
> >    than 1300 bytes from being delivered using TCP as described in
> >    Section 18.1.1 RFC 3261.  Use of outbound in conjunction with UDP
> >    flows in environments where requests are expected to be=20
> larger than
> >    this size is therefore NOT RECOMMENDED.
> >
> > In practice this seems to justify implementing "TCP only" UAs
> > what comes to the Outbound support. In that case the basic=20
> assumption
> > with STUN keepalive (implement it for TCP as it is anyway needed for
> > UDP as well) would become obsolete. For implementations which can
> > not use TCP keepalive, CRLF would obviously provide a great benefit
> > without significant increase of complexity to Outbound.
> >
> >
> > P.S. Last week I made a proposal for "stripped Outbound" without
> > UDP, STUN keepalives and multiple registrations. As that proposal
> > did not receive any comments, I decided to propose this minimal
> > addition to Outbound as an alternative that might be a bit easier
> > to accept but still achieve some of the benefits.
> >
> > Regards,
> >
> > Erkki
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
> >
>=20

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From sip-bounces@ietf.org Fri Dec 01 08:37:19 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 1 Dec 2006 14:35:21 +0100
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Francois Audet" <audet@nortel.com>, "Kevin Johns" <K.Johns@CableLabs.com>,
	"Fredrik Thulin" <ft@it.su.se>
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Hi,=20

>My point was that the keep-alive didn't matter since any=20
>option will work, and you don't need one in the first place=20
>because the UDP/IPsec already has one.

Exactly.

So, do we really need to specify what other keepalive mechanisms than
STUN can be used then? Can't we simply say that one shall use STUN
unless another mechanism is provided by the transport/security layer?

>But it just occured to me... I'm thinking VPN, and you are=20
>probably thinking 3GPP. And I don't know anything about how=20
>that works. :-)

I was thinking 3GPP, yes :)

Regards,

Christer


=20
> > -----Original Message-----
> > From: Christer Holmberg (JO/LMF)
> > [mailto:christer.holmberg@ericsson.com]
> > Sent: Thursday, November 30, 2006 13:49
> > To: Audet, Francois (SC100:3055); Kevin Johns; Fredrik Thulin
> > Cc: sip@ietf.org; rohan@ekabal.com
> > Subject: RE: [Sip] Re: issue with oubound and the used transport=20
> > protocol
> >=20
> >=20
> > Hi,
> >=20
> > >I would think that UDP encapsulated IPsec would have it's own=20
> > >keep-alive.
> > >So they wouldn't need Outbound in the first place.
> >=20
> > They may want to use outbound in order to support multiple=20
> simultanous=20
> > registrations.
> >=20
> > So, if there would also be NATs, the question is what=20
> keep-alive they=20
> > should use.
> >=20
> > Regards,
> >=20
> > Christer
> >=20
> >=20
> >=20
> >=20
> > >=20
> > > But even if they didn't then the CRLF thing would still work.
> > >=20
> > > > -----Original Message-----
> > > > From: Kevin Johns [mailto:K.Johns@CableLabs.com]
> > > > Sent: Thursday, November 30, 2006 12:47
> > > > To: Fredrik Thulin; Audet, Francois (SC100:3055)
> > > > Cc: sip@ietf.org; rohan@ekabal.com
> > > > Subject: RE: [Sip] Re: issue with oubound and the used=20
> transport=20
> > > > protocol
> > > >=20
> > > > Not all architectures use TLS/TCP, some use UDP
> > encapsulated IPsec.
> > > > Would a TCP session via UDP encapsulated IPsec be impacted
> > > by this? I
> > > > do not think it would, but just wanted to double check...
> > > >=20
> > > > Kevin
> > > >=20
> > > > -----Original Message-----
> > > > From: Fredrik Thulin [mailto:ft@it.su.se]
> > > > Sent: Thursday, November 30, 2006 11:21 AM
> > > > To: Francois Audet
> > > > Cc: sip@ietf.org; rohan@ekabal.com
> > > > Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport=20
> > > > protocol
> > > >=20
> > > > Francois Audet wrote:
> > > > > Didn't we agree way back NOT to use PING, CRLF, etc., and
> > > use STUN
> > > > > instead?
> > > > >=20
> > > > > Why is this debate being re-opened?
> > > >=20
> > > > Because it seems more and more (to me at least) that
> > > Outbound with UDP
> > > > will just suck.
> > > >=20
> > > > If Outbound UA's have to get their act together and
> > > implement TLS/TCP,
> > > > then we can simplify Outbound even further by skipping the
> > > whole STUN
> > > > keepalive thing, and go with CRLF over TLS/TCP.
> > > >=20
> > > > I am very much in favor of cutting both UDP and STUN
> > > keepalives out of
> > > > Outbound. I think that would lead to faster deployment, and
> > > also make
> > > > the world a better place by eliminating some SIP over UDP.
> > > >=20
> > > > /Fredrik
> > > >=20
> > > > PS. By the way, when people say CRLF keepalives, do you
> > > mean CRLF sent
> > > > only from the UA, or CRLF from UA to edge proxy followed
> > by CRLF in
> > > > the reverse direction too?
> > > >=20
> > > >=20
> > > >=20
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > > >=20
> > > >=20
> > >=20
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >=20
> >=20
>=20

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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol -
	decide whether to send keep-alive in the first place 
Date: Fri, 1 Dec 2006 14:54:21 +0100
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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Hi,

No matter how we do the keep-alive, I think STUN is a good tool for the
UA to figure out whether it's located behind a NAT in the first place,
by checking the XMA value etc. Again, outbound can be used also in
no-NAT scenarios, so the UA can use the STUN response information to
decide whether it needs to send keep-alives in the first place.

Regards,

Christer

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From sip-bounces@ietf.org Fri Dec 01 09:23:25 2006
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From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [Sip] Issue: Expiration of temp-gruu
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For issue 3, I would say that the solution is to avoid the situation, 
e.g., by registering way in advance before the registration expires. 
Doesn't RFC 3261 recommends to re-register half time before the 
registration timer expires.

Issues 4, 5, and 6 are solved by the reg event. So, shouldn't the GRUU 
draft have a recommendation on the usage of the reg event to solve them?

/Miguel

Paul Kyzivat wrote:
> After some study, I have encountered a potential difficulty in 
> determining when temp-gruus expire. After discussing this with Jonathan 
> we decided to bring it to the list for discussion.
> 
> As now defined, when you first register you may get a temp-gruu, which 
> will remain valid as long as the registration does not expire or be 
> removed. (Removal is a special case of expiration.) Each time the UA 
> refreshes the registration it may get a new temp-gruu which also remains 
> valid for the life of the registration. The UA can accumulate all these 
> temp-gruus and use them as it sees fit until they expire.
> 
> The problem is: how does the UA know when the registration has ended and 
> the temp-gruus associated with it rendered invalid?
> 
> This seems like a simple question, and in some cases it is simple, but 
> not in all cases. The following are at least some of the ways that the 
> registration can be ended, causing the invalidation of temp-gruus:
> 1) the UA that originally registered may explicitly deregister
> 2) the UA the originally registered may simply permit the
>    registration to expire without attempting to refresh it
> 3) the UA may attempt to refresh the registration, but be too slow,
>    so that the reregistration arrives after the prior one expires
> 4) some other UA may deregister all the registrations, including
>    the one by this UA. (When the UA decides it should refresh the
>    registration it will actually create a new one.)
> 5) the registrar may remove the registration administratively
> 6) the registrar may crash and restart, losing all its registrations
> 
> In each of these cases the previously assigned temp-gruus become 
> invalid. If the UA continues to use them, bad things will happen. This 
> is true both for calls that are active when the deregistration occurs, 
> and calls that are established later. What kinds of bad things happen?
> - requests addressed to the gruu, both in-dialog and out-of-dialog
>   will be undeliverable. (For instance a reINVITE or BYE in an
>   existing call, or an out-of-dialog INVITE/Replaces for a transfer.)
> 
> - request originated by the UA using the invalid gruu will fail
>   immediately because the gruu is invalid.
> 
> - if the UA subsequently reregisters, receives an incoming call, and
>   uses one of the old temp-gruus as its contact, then the call will
>   be established, but subsequent in- and out-of-dialog requests to
>   the contact will fail.
> 
> So clearly it is important for the UA to stop using temp-gruus that have 
> become invalid. How can it know to do that in all of the above cases?
> 
> 1) is straightforward. The UA knows it is deregistering, and should
>    discard any cached temp-gruus. If it has a dialog with one of the
>    temp-gruus as a contact, then it should do a target refresh and
>    supply some working contact. (It really should do this *before*
>    deregistering, and for a pub-gruu as well.)
> 
> 2) the UA should take all the actions mentioned for (1) at or before
>    the expiration time. This should be done a bit early to guard
>    against clock skew problems.
> 
> 3) in this case, it may be impossible for the UA to know whether
>    the REGISTER arrived in time to be a reREGISTER or if it arrived
>    late and was treated as a new REGISTER after the expiration of
>    the old one. (The response to the REGISTER includes nothing that
>    distinguishes the two cases.) About the best that can be done in
>    this case is to check the time at which the response to the
>    REGISTER is received. If it falls after the expected expiration
>    time for the registration, act as if the old registration had
>    expired.
> 
> 4) This is a major problem. Of course, regardless of GRUU, the UA
>    won't be able to receive incoming calls until it decides it is
>    time to refresh the registration. In addition, until the
>    registration reestablished, calls originated with one of its
>    existing gruus will fail. Once the registration is reestablished,
>    it still won't know its old temp-gruus are invalid. The only
>    remedy for this is to subscribe to the "reg" event package,
>    and use the resulting notifications to identify when it has
>    been unregistered. If it does that, it can invalidate all its
>    old temp-gruus, and then presumably reregister.
> 
> 5) this is functionally equivalent to (4), except that perhaps
>    attempts to reregister may fail.
> 
> 6) this is also functionally equivalent to (4).
> 
> The "reg" event does provide a fairly complete solution to this problem, 
> for those that are willing and able to use it. It is not however a 100% 
> reliable solution. Consider case (3), where the UA has a reg event 
> subscription. All is well if the registration expires and a notification 
> is sent indicating that, and then the reregistration occurs and a 
> notification is sent about that. But if they happen very closely in 
> time, it may be that only one notification is sent, reflecting the 
> result of the reregistration. In that case the UA will not be aware that 
> it had been unregistered for a period of time.
> 
> A *complete* solution would, IMO, involve changing REGISTER so that it 
> returns an indication of whether this was a new registration, or a 
> refresh. This could take many forms. For instance the timestamp of the 
> initial registration could always be returned with each contact, as well 
> as the expiration time.
> 
> The big question is whether this is a big enough issue to require a 
> solution now, as part of GRUU, or if it can be ignored as insignificant 
> or postponed to future work.
> 
> I'd appreciate hearing what others think about this. At the moment I am 
> leaning toward doing nothing, at least for now.
> 
>     Thanks,
>     Paul
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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From sip-bounces@ietf.org Fri Dec 01 11:27:16 2006
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From: Paul Kyzivat <pkyzivat@cisco.com>
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To: Miguel Garcia <Miguel.An.Garcia@nokia.com>
Subject: Re: [Sip] Issue: Expiration of temp-gruu
References: <456E15CD.60805@cisco.com> <45703AA0.20007@nokia.com>
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Miguel,

So to summarize, it seems you are of the belief that the cases where 
these measures may be inadequate are sufficiently obscure to ignore, and 
that new measures to cover the obscure cases are unnecessary. Is that right?

	Paul

Miguel Garcia wrote:
> For issue 3, I would say that the solution is to avoid the situation, 
> e.g., by registering way in advance before the registration expires. 
> Doesn't RFC 3261 recommends to re-register half time before the 
> registration timer expires.
> 
> Issues 4, 5, and 6 are solved by the reg event. So, shouldn't the GRUU 
> draft have a recommendation on the usage of the reg event to solve them?
> 
> /Miguel
> 
> Paul Kyzivat wrote:
>> After some study, I have encountered a potential difficulty in 
>> determining when temp-gruus expire. After discussing this with 
>> Jonathan we decided to bring it to the list for discussion.
>>
>> As now defined, when you first register you may get a temp-gruu, which 
>> will remain valid as long as the registration does not expire or be 
>> removed. (Removal is a special case of expiration.) Each time the UA 
>> refreshes the registration it may get a new temp-gruu which also 
>> remains valid for the life of the registration. The UA can accumulate 
>> all these temp-gruus and use them as it sees fit until they expire.
>>
>> The problem is: how does the UA know when the registration has ended 
>> and the temp-gruus associated with it rendered invalid?
>>
>> This seems like a simple question, and in some cases it is simple, but 
>> not in all cases. The following are at least some of the ways that the 
>> registration can be ended, causing the invalidation of temp-gruus:
>> 1) the UA that originally registered may explicitly deregister
>> 2) the UA the originally registered may simply permit the
>>    registration to expire without attempting to refresh it
>> 3) the UA may attempt to refresh the registration, but be too slow,
>>    so that the reregistration arrives after the prior one expires
>> 4) some other UA may deregister all the registrations, including
>>    the one by this UA. (When the UA decides it should refresh the
>>    registration it will actually create a new one.)
>> 5) the registrar may remove the registration administratively
>> 6) the registrar may crash and restart, losing all its registrations
>>
>> In each of these cases the previously assigned temp-gruus become 
>> invalid. If the UA continues to use them, bad things will happen. This 
>> is true both for calls that are active when the deregistration occurs, 
>> and calls that are established later. What kinds of bad things happen?
>> - requests addressed to the gruu, both in-dialog and out-of-dialog
>>   will be undeliverable. (For instance a reINVITE or BYE in an
>>   existing call, or an out-of-dialog INVITE/Replaces for a transfer.)
>>
>> - request originated by the UA using the invalid gruu will fail
>>   immediately because the gruu is invalid.
>>
>> - if the UA subsequently reregisters, receives an incoming call, and
>>   uses one of the old temp-gruus as its contact, then the call will
>>   be established, but subsequent in- and out-of-dialog requests to
>>   the contact will fail.
>>
>> So clearly it is important for the UA to stop using temp-gruus that 
>> have become invalid. How can it know to do that in all of the above 
>> cases?
>>
>> 1) is straightforward. The UA knows it is deregistering, and should
>>    discard any cached temp-gruus. If it has a dialog with one of the
>>    temp-gruus as a contact, then it should do a target refresh and
>>    supply some working contact. (It really should do this *before*
>>    deregistering, and for a pub-gruu as well.)
>>
>> 2) the UA should take all the actions mentioned for (1) at or before
>>    the expiration time. This should be done a bit early to guard
>>    against clock skew problems.
>>
>> 3) in this case, it may be impossible for the UA to know whether
>>    the REGISTER arrived in time to be a reREGISTER or if it arrived
>>    late and was treated as a new REGISTER after the expiration of
>>    the old one. (The response to the REGISTER includes nothing that
>>    distinguishes the two cases.) About the best that can be done in
>>    this case is to check the time at which the response to the
>>    REGISTER is received. If it falls after the expected expiration
>>    time for the registration, act as if the old registration had
>>    expired.
>>
>> 4) This is a major problem. Of course, regardless of GRUU, the UA
>>    won't be able to receive incoming calls until it decides it is
>>    time to refresh the registration. In addition, until the
>>    registration reestablished, calls originated with one of its
>>    existing gruus will fail. Once the registration is reestablished,
>>    it still won't know its old temp-gruus are invalid. The only
>>    remedy for this is to subscribe to the "reg" event package,
>>    and use the resulting notifications to identify when it has
>>    been unregistered. If it does that, it can invalidate all its
>>    old temp-gruus, and then presumably reregister.
>>
>> 5) this is functionally equivalent to (4), except that perhaps
>>    attempts to reregister may fail.
>>
>> 6) this is also functionally equivalent to (4).
>>
>> The "reg" event does provide a fairly complete solution to this 
>> problem, for those that are willing and able to use it. It is not 
>> however a 100% reliable solution. Consider case (3), where the UA has 
>> a reg event subscription. All is well if the registration expires and 
>> a notification is sent indicating that, and then the reregistration 
>> occurs and a notification is sent about that. But if they happen very 
>> closely in time, it may be that only one notification is sent, 
>> reflecting the result of the reregistration. In that case the UA will 
>> not be aware that it had been unregistered for a period of time.
>>
>> A *complete* solution would, IMO, involve changing REGISTER so that it 
>> returns an indication of whether this was a new registration, or a 
>> refresh. This could take many forms. For instance the timestamp of the 
>> initial registration could always be returned with each contact, as 
>> well as the expiration time.
>>
>> The big question is whether this is a big enough issue to require a 
>> solution now, as part of GRUU, or if it can be ignored as 
>> insignificant or postponed to future work.
>>
>> I'd appreciate hearing what others think about this. At the moment I 
>> am leaning toward doing nothing, at least for now.
>>
>>     Thanks,
>>     Paul
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 1 Dec 2006 12:43:47 -0600
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From: "Francois Audet" <audet@nortel.com>
To: <Erkki.Koivusalo@nokia.com>, <rohan.mahy@gmail.com>
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I'm OK with either 1 or 2.

The difference is that with 1 we make it obvious that TCP is better.
With 2, the reader has
to figure it out by himself.=20

> -----Original Message-----
> From: Erkki.Koivusalo@nokia.com [mailto:Erkki.Koivusalo@nokia.com]=20
> Sent: Friday, December 01, 2006 04:12
> To: rohan.mahy@gmail.com
> Cc: sip@ietf.org; rohan@ekabal.com
> Subject: RE: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
>=20
> Hi,
>=20
> You asked me to clarify my CRLF proposal, so I try to do it here.
>=20
> I have been exploring ways of rescoping drafts related to=20
> Outbound and simplifying the mechanisms within them since the=20
> limitations of using UDP flows became apparent in this=20
> thread. I could vote for a number of different approaches and=20
> would be happy the WG to agree with at least one of them.=20
> Those approaches in my preference order are:
>=20
> 1. Outbound draft to re-scope itself to "Outbound TCP connection"
>    draft without any support to UDP, STUN or multiple registrations.
>    Multiple registrations and mid-dialog failover (using either TCP
>    or UDP) to be specified within another draft.
>=20
>    - This proposal as such did not raise any discussion=20
>      on the list, probably because it would be a dramatic
>      change at a quite late stage of the draft.
>    - Remarkable changes needed (but I already outlined the
>      "Outbound TCP connection" draft within my earlier mail :).
>      http://www1.ietf.org/mail-archive/web/sip/current/msg17495.html
>=20
> 2. Outbound draft to replace STUN keepalive over TCP with CRLF
>    keepalive over TCP. (No impact to UDP support of Outbound)
>=20
>    - This option is equivalent to your formulation, which I
>      can very much agree on:
>=20
> > Clearly you are suggesting that we make CRLF mandatory to implement=20
> > for TCP.  There would be no longer any need to do STUN over TCP and=20
> > therefore no need implement code to deal with stream-based demuxing.
> > The text about sigcomp over TCP and STUN over TLS over TCP=20
> could also=20
> > go away.
>=20
>    - Some WG members already expressed their support for this
>      option
>    - Minor changes needed to the draft
>=20
> 3. Outbound draft to introduce a third optional keepalive method
>    for TCP: CRLF (in addition to STUN and TCP keepalives)
>=20
>    - This option would be the _last resort_ to get at least some
>      benefit for TCP-based UA implementation, even if it would
>      not help the proxies. Proxies would still have to support
>      STUN over TCP.=20
>    - Minimal changes needed to the draft
>=20
>=20
> > To clarify are you suggesting that an edge proxy can implement=20
> > outbound without support for UDP flow tokens, without=20
> support for STUN=20
> > keepalives over UDP, both, or neither?
>=20
> My proposals were born "UA centric" as I am from the UA vendor camp.
>=20
> Anyway if Outbound will specify both UDP and TCP as transport=20
> options, I would believe the proxies still have to support=20
> both since the selection of the transport is basically an UA=20
> decision (even if assisted by DNS SRV) as it depends on if=20
> the proxy URI configured to the UA has a transport parameter.=20
> If the proxy would not support UDP & STUN, how could it tell=20
> it to the UA ? Outbound would at least need a mechanism for=20
> that e.g. a specific response code, option tag or something=20
> else. Or is this yet another thing where Outbound would rely=20
> on the UA and DNS configuration to tell the UA to avoid using UDP ?
>=20
> Considering the option to drop UDP & STUN support from Outbound
> draft: I would agree that combination of UDP and STUN=20
> keepalives does not seem to provide a practical way for NAT=20
> traversal. Keeping UDP flows alive with STUN seems not to add=20
> value in comparison to persistent TCP flows and I could agree=20
> dropping UDP flows from the draft. The only problem I see=20
> here that Outbound covers also the multiregistration=20
> mechanism, which could still be useful with UDP e.g. in=20
> cellular access context without NAT/FWs. That is why I have=20
> been proposing re-scoping of the drafts:
>=20
> - Flow keepalive mechanisms to be specified only for TCP (one draft)
> - Protocol agnostic multiregistration mechanism with mid-dialog
>   failover support (another draft)
>=20
> Regards,
>=20
> Erkki
>=20
> > -----Original Message-----
> > From: ext Rohan Mahy [mailto:rohan.mahy@gmail.com]
> > Sent: 29 November, 2006 20:30
> > To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
> > Cc: sip@ietf.org; rohan@ekabal.com
> > Subject: Re: [Sip] Re: issue with oubound and the used transport=20
> > protocol
> >=20
> >=20
> > Hi Erkki,
> >=20
> > Clearly you are suggesting that we make CRLF mandatory to implement=20
> > for TCP.  There would be no longer any need to do STUN over TCP and=20
> > therefore no need implement code to deal with stream-based demuxing.
> >   The text about sigcomp over TCP and STUN over TLS over TCP could=20
> > also go away.
> >=20
> > To clarify are you suggesting that an edge proxy can implement=20
> > outbound without support for UDP flow tokens, without=20
> support for STUN=20
> > keepalives over UDP, both, or neither?
> >=20
> > thanks,
> > -rohan
> >=20
> > On 11/29/06, Erkki.Koivusalo@nokia.com <Erkki.Koivusalo@nokia.com>=20
> > wrote:
> > >
> > > Hi,
> > >
> > > >about STUN keepalive usage and discovery, since double
> > CRLF is fully
> > > >backwards compatible with even RFC 2543 implementations.  I am=20
> > > >bringing this up on the list because I hope the WG can=20
> look back on=20
> > > >this and not make the mistake of snowballing complexity=20
> to preserve=20
> > > >UDP parity again in the future.
> > >
> > > Rohan is making a valid point here and I believe in=20
> Outbound there=20
> > > is still room for change to cater for CRLF keepalives as=20
> an option=20
> > > to STUN keepalives over TCP.
> > >
> > > I am calling the WG to look back on the keepalive decision made=20
> > > earlier for TCP and proposing that Outbound should also=20
> cover CRLF=20
> > > keepalive option.
> > >
> > > Reasons for this proposal:
> > >
> > > >The problem I see is that we have two separate communities
> > using the
> > > >outbound draft:
> > > >
> > > >1) folks that want to get through NATs and Firewalls in=20
> the public=20
> > > >Internet at large - they want to use TCP and TLS with
> > double CRLF for
> > > >keepalives. One "flow" is always one connection.  The=20
> milestone for=20
> > > >this would read "Make TCP and TLS work in the Internet".
> > >
> > > I agree with the intents of this community. However this=20
> is not what=20
> > > the current Outbound provides. Outbound specifies two options for=20
> > > keepalives for reliable connections: TCP keepalive and STUN.
> > >
> > > These options were agreed in IETF 65, a cite from the minutes:
> > >
> > > Proposal:
> > >    Do STUN for TCP and UDP
> > >    Do CRLF for TCP and STUN for UDP
> > >    TCP KeepAlive if supported, 1 or 2 otherwise
> > >
> > > Resolutions:
> > >    OS TCP KeepAlive should be used if the client supports it.
> > >    Need something more because OS TCP KeepAlive is not=20
> supported in=20
> > > every
> > >    client OS.
> > >    Consensus to use STUN for UDP
> > >    Consensus to use STUN for TCP, with use of OS TCP keep
> > alive above.
> > >
> > > In that meeting the basic assumption was that every Outbound=20
> > > compliant client should implement both UDP and TCP even=20
> if that is=20
> > > not explicitely mentioned in the draft. However latest=20
> -06 version=20
> > > now makes that debatable by stating:
> > >
> > >    Use of outbound with UDP flows typically prevents=20
> messages larger
> > >    than 1300 bytes from being delivered using TCP as described in
> > >    Section 18.1.1 RFC 3261.  Use of outbound in=20
> conjunction with UDP
> > >    flows in environments where requests are expected to be
> > larger than
> > >    this size is therefore NOT RECOMMENDED.
> > >
> > > In practice this seems to justify implementing "TCP only"=20
> UAs what=20
> > > comes to the Outbound support. In that case the basic
> > assumption
> > > with STUN keepalive (implement it for TCP as it is anyway=20
> needed for=20
> > > UDP as well) would become obsolete. For implementations which can=20
> > > not use TCP keepalive, CRLF would obviously provide a=20
> great benefit=20
> > > without significant increase of complexity to Outbound.
> > >
> > >
> > > P.S. Last week I made a proposal for "stripped Outbound" without=20
> > > UDP, STUN keepalives and multiple registrations. As that proposal=20
> > > did not receive any comments, I decided to propose this minimal=20
> > > addition to Outbound as an alternative that might be a=20
> bit easier to=20
> > > accept but still achieve some of the benefits.
> > >
> > > Regards,
> > >
> > > Erkki
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> > >
> >=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 1 Dec 2006 12:42:02 -0600
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Thread-Topic: [Sip] Re: issue with oubound and the used transport protocol
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From: "Francois Audet" <audet@nortel.com>
To: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>,
	"Kevin Johns" <K.Johns@CableLabs.com>, "Fredrik Thulin" <ft@it.su.se>
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I think we are in violent agreement.

I'm not even sure we need to say anything here. Seems to me it's a
3GPP-specific
non-issue...=20

> -----Original Message-----
> From: Christer Holmberg (JO/LMF)=20
> [mailto:christer.holmberg@ericsson.com]=20
> Sent: Friday, December 01, 2006 05:35
> To: Audet, Francois (SC100:3055); Kevin Johns; Fredrik Thulin
> Cc: sip@ietf.org; rohan@ekabal.com
> Subject: RE: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
>=20
> Hi,=20
>=20
> >My point was that the keep-alive didn't matter since any option will=20
> >work, and you don't need one in the first place because the=20
> UDP/IPsec=20
> >already has one.
>=20
> Exactly.
>=20
> So, do we really need to specify what other keepalive=20
> mechanisms than STUN can be used then? Can't we simply say=20
> that one shall use STUN unless another mechanism is provided=20
> by the transport/security layer?
>=20
> >But it just occured to me... I'm thinking VPN, and you are probably=20
> >thinking 3GPP. And I don't know anything about how that works. :-)
>=20
> I was thinking 3GPP, yes :)
>=20
> Regards,
>=20
> Christer
>=20
>=20
> =20
> > > -----Original Message-----
> > > From: Christer Holmberg (JO/LMF)
> > > [mailto:christer.holmberg@ericsson.com]
> > > Sent: Thursday, November 30, 2006 13:49
> > > To: Audet, Francois (SC100:3055); Kevin Johns; Fredrik Thulin
> > > Cc: sip@ietf.org; rohan@ekabal.com
> > > Subject: RE: [Sip] Re: issue with oubound and the used transport=20
> > > protocol
> > >=20
> > >=20
> > > Hi,
> > >=20
> > > >I would think that UDP encapsulated IPsec would have it's own=20
> > > >keep-alive.
> > > >So they wouldn't need Outbound in the first place.
> > >=20
> > > They may want to use outbound in order to support multiple
> > simultanous
> > > registrations.
> > >=20
> > > So, if there would also be NATs, the question is what
> > keep-alive they
> > > should use.
> > >=20
> > > Regards,
> > >=20
> > > Christer
> > >=20
> > >=20
> > >=20
> > >=20
> > > >=20
> > > > But even if they didn't then the CRLF thing would still work.
> > > >=20
> > > > > -----Original Message-----
> > > > > From: Kevin Johns [mailto:K.Johns@CableLabs.com]
> > > > > Sent: Thursday, November 30, 2006 12:47
> > > > > To: Fredrik Thulin; Audet, Francois (SC100:3055)
> > > > > Cc: sip@ietf.org; rohan@ekabal.com
> > > > > Subject: RE: [Sip] Re: issue with oubound and the used
> > transport
> > > > > protocol
> > > > >=20
> > > > > Not all architectures use TLS/TCP, some use UDP
> > > encapsulated IPsec.
> > > > > Would a TCP session via UDP encapsulated IPsec be impacted
> > > > by this? I
> > > > > do not think it would, but just wanted to double check...
> > > > >=20
> > > > > Kevin
> > > > >=20
> > > > > -----Original Message-----
> > > > > From: Fredrik Thulin [mailto:ft@it.su.se]
> > > > > Sent: Thursday, November 30, 2006 11:21 AM
> > > > > To: Francois Audet
> > > > > Cc: sip@ietf.org; rohan@ekabal.com
> > > > > Subject: Re: [Sip] Re: issue with oubound and the used
> > transport
> > > > > protocol
> > > > >=20
> > > > > Francois Audet wrote:
> > > > > > Didn't we agree way back NOT to use PING, CRLF, etc., and
> > > > use STUN
> > > > > > instead?
> > > > > >=20
> > > > > > Why is this debate being re-opened?
> > > > >=20
> > > > > Because it seems more and more (to me at least) that
> > > > Outbound with UDP
> > > > > will just suck.
> > > > >=20
> > > > > If Outbound UA's have to get their act together and
> > > > implement TLS/TCP,
> > > > > then we can simplify Outbound even further by skipping the
> > > > whole STUN
> > > > > keepalive thing, and go with CRLF over TLS/TCP.
> > > > >=20
> > > > > I am very much in favor of cutting both UDP and STUN
> > > > keepalives out of
> > > > > Outbound. I think that would lead to faster deployment, and
> > > > also make
> > > > > the world a better place by eliminating some SIP over UDP.
> > > > >=20
> > > > > /Fredrik
> > > > >=20
> > > > > PS. By the way, when people say CRLF keepalives, do you
> > > > mean CRLF sent
> > > > > only from the UA, or CRLF from UA to edge proxy followed
> > > by CRLF in
> > > > > the reverse direction too?
> > > > >=20
> > > > >=20
> > > > >=20
> > > > > _______________________________________________
> > > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > > sip-implementors@cs.columbia.edu for questions on
> > current sip Use
> > > > > sipping@ietf.org for new developments on the=20
> application of sip
> > > > >=20
> > > > >=20
> > > >=20
> > > > _______________________________________________
> > > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > > This list is for NEW development of the core SIP Protocol Use=20
> > > > sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> > > > sipping@ietf.org for new developments on the application of sip
> > > >=20
> > >=20
> >=20
>=20

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The main question is really to decide on one of two mechanisms:

1. use reg-event if you are worried about use cases where this race 
condition can occur (4-6 in Paul's original email)

2. add a parameter to Contact in the register response. Specifically, 
I'd recommend a Contact header field parameter which indicates the time 
at which the first contact with a given instance ID was registered to 
that AOR. This value gets linked with each temp gruu, and when the value 
changes, all of those temp-gruu get invalidated.

Solution (2) is much simpler, but not needed if you think you'll always 
have reg-event anyway. I don't think there is widespread support for 
reg-event on endpoints yet. It is required by IMS.

I am somewhat on the fence on this one. I do worry that reg-event is 
genreally a very heavy solution, and may not see much usage. Option 2 is 
really trivially done, either in gruu or elsewhere. The main issue is 
that it is another late change.

-JOnathan R.



Paul Kyzivat wrote:

> Miguel,
> 
> So to summarize, it seems you are of the belief that the cases where 
> these measures may be inadequate are sufficiently obscure to ignore, and 
> that new measures to cover the obscure cases are unnecessary. Is that 
> right?
> 
>     Paul
> 
> Miguel Garcia wrote:
> 
>> For issue 3, I would say that the solution is to avoid the situation, 
>> e.g., by registering way in advance before the registration expires. 
>> Doesn't RFC 3261 recommends to re-register half time before the 
>> registration timer expires.
>>
>> Issues 4, 5, and 6 are solved by the reg event. So, shouldn't the GRUU 
>> draft have a recommendation on the usage of the reg event to solve them?
>>
>> /Miguel
>>
>> Paul Kyzivat wrote:
>>
>>> After some study, I have encountered a potential difficulty in 
>>> determining when temp-gruus expire. After discussing this with 
>>> Jonathan we decided to bring it to the list for discussion.
>>>
>>> As now defined, when you first register you may get a temp-gruu, 
>>> which will remain valid as long as the registration does not expire 
>>> or be removed. (Removal is a special case of expiration.) Each time 
>>> the UA refreshes the registration it may get a new temp-gruu which 
>>> also remains valid for the life of the registration. The UA can 
>>> accumulate all these temp-gruus and use them as it sees fit until 
>>> they expire.
>>>
>>> The problem is: how does the UA know when the registration has ended 
>>> and the temp-gruus associated with it rendered invalid?
>>>
>>> This seems like a simple question, and in some cases it is simple, 
>>> but not in all cases. The following are at least some of the ways 
>>> that the registration can be ended, causing the invalidation of 
>>> temp-gruus:
>>> 1) the UA that originally registered may explicitly deregister
>>> 2) the UA the originally registered may simply permit the
>>>    registration to expire without attempting to refresh it
>>> 3) the UA may attempt to refresh the registration, but be too slow,
>>>    so that the reregistration arrives after the prior one expires
>>> 4) some other UA may deregister all the registrations, including
>>>    the one by this UA. (When the UA decides it should refresh the
>>>    registration it will actually create a new one.)
>>> 5) the registrar may remove the registration administratively
>>> 6) the registrar may crash and restart, losing all its registrations
>>>
>>> In each of these cases the previously assigned temp-gruus become 
>>> invalid. If the UA continues to use them, bad things will happen. 
>>> This is true both for calls that are active when the deregistration 
>>> occurs, and calls that are established later. What kinds of bad 
>>> things happen?
>>> - requests addressed to the gruu, both in-dialog and out-of-dialog
>>>   will be undeliverable. (For instance a reINVITE or BYE in an
>>>   existing call, or an out-of-dialog INVITE/Replaces for a transfer.)
>>>
>>> - request originated by the UA using the invalid gruu will fail
>>>   immediately because the gruu is invalid.
>>>
>>> - if the UA subsequently reregisters, receives an incoming call, and
>>>   uses one of the old temp-gruus as its contact, then the call will
>>>   be established, but subsequent in- and out-of-dialog requests to
>>>   the contact will fail.
>>>
>>> So clearly it is important for the UA to stop using temp-gruus that 
>>> have become invalid. How can it know to do that in all of the above 
>>> cases?
>>>
>>> 1) is straightforward. The UA knows it is deregistering, and should
>>>    discard any cached temp-gruus. If it has a dialog with one of the
>>>    temp-gruus as a contact, then it should do a target refresh and
>>>    supply some working contact. (It really should do this *before*
>>>    deregistering, and for a pub-gruu as well.)
>>>
>>> 2) the UA should take all the actions mentioned for (1) at or before
>>>    the expiration time. This should be done a bit early to guard
>>>    against clock skew problems.
>>>
>>> 3) in this case, it may be impossible for the UA to know whether
>>>    the REGISTER arrived in time to be a reREGISTER or if it arrived
>>>    late and was treated as a new REGISTER after the expiration of
>>>    the old one. (The response to the REGISTER includes nothing that
>>>    distinguishes the two cases.) About the best that can be done in
>>>    this case is to check the time at which the response to the
>>>    REGISTER is received. If it falls after the expected expiration
>>>    time for the registration, act as if the old registration had
>>>    expired.
>>>
>>> 4) This is a major problem. Of course, regardless of GRUU, the UA
>>>    won't be able to receive incoming calls until it decides it is
>>>    time to refresh the registration. In addition, until the
>>>    registration reestablished, calls originated with one of its
>>>    existing gruus will fail. Once the registration is reestablished,
>>>    it still won't know its old temp-gruus are invalid. The only
>>>    remedy for this is to subscribe to the "reg" event package,
>>>    and use the resulting notifications to identify when it has
>>>    been unregistered. If it does that, it can invalidate all its
>>>    old temp-gruus, and then presumably reregister.
>>>
>>> 5) this is functionally equivalent to (4), except that perhaps
>>>    attempts to reregister may fail.
>>>
>>> 6) this is also functionally equivalent to (4).
>>>
>>> The "reg" event does provide a fairly complete solution to this 
>>> problem, for those that are willing and able to use it. It is not 
>>> however a 100% reliable solution. Consider case (3), where the UA has 
>>> a reg event subscription. All is well if the registration expires and 
>>> a notification is sent indicating that, and then the reregistration 
>>> occurs and a notification is sent about that. But if they happen very 
>>> closely in time, it may be that only one notification is sent, 
>>> reflecting the result of the reregistration. In that case the UA will 
>>> not be aware that it had been unregistered for a period of time.
>>>
>>> A *complete* solution would, IMO, involve changing REGISTER so that 
>>> it returns an indication of whether this was a new registration, or a 
>>> refresh. This could take many forms. For instance the timestamp of 
>>> the initial registration could always be returned with each contact, 
>>> as well as the expiration time.
>>>
>>> The big question is whether this is a big enough issue to require a 
>>> solution now, as part of GRUU, or if it can be ignored as 
>>> insignificant or postponed to future work.
>>>
>>> I'd appreciate hearing what others think about this. At the moment I 
>>> am leaning toward doing nothing, at least for now.
>>>
>>>     Thanks,
>>>     Paul
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com

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From sip-bounces@ietf.org Fri Dec 01 19:34:55 2006
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From: "Rohan Mahy" <rohan.mahy@gmail.com>
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Subject: [Sip] Re: CRLF draft?
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Hi Attila,

This was described in an earlier version of the outbound draft.  The
text from the last version it appeared is below:

thanks,
-rohan

<section title="Double CRLF Processing">
<t>If the SIP server is acting as the TCP client and initiated the TCP
connection (meaning that this host did the active open), then the SIP
server MUST NOT perform any of the processing in this section. The
following only applies when the SIP server is acting as the TCP server
(meaning that this host did the passive open).</t>

<t>When the server receives a CRLF before the start line of a message
on a flow, it MUST send some data back on that same flow within 3
seconds. If no message is actively being sent, it SHOULD send back a
CRLF after waiting at least 1 second. The reason for waiting at least
1 second is that if the other end has an incorrect implementation and
incorrectly echoes the CRLF, this will stop the flow from going into a
live-lock state.</t>
</section>

On 11/30/06, Attila Sipos <Attila.Sipos@vegastream.com> wrote:
>
> Every now and then I "hear" people mention CRLF.
> Is there a draft for this?
> Or a draft which mentions this?
> I assume it's something where, instead of a full SIP
> message, just a CRLF is sent to keep a NAT binding open.
> Is it only meant for UDP or just for TCP or both?
> Regards,
> Attila
>
>

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Subject: RE: [Sip] Re: SIPS and the PSTN
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From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Samir Srivastava" <samirsr@nortel.com>,
	"Francois Audet" <audet@nortel.com>,
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We don't vote, we operate by rough consensus.

We have rough consensus as a result of IETF#66 and the subsequent
approval of the proceedings to proceed with an informative document.

That means that to overturn that, you need rough consensus in entirely
the opposite direction. Given that there is at least a body of people
interested in completing this document, you seem to have a significant
hurdle to overcome.

Regards

Keith

> -----Original Message-----
> From: Samir Srivastava [mailto:samirsr@nortel.com]=20
> Sent: 29 November 2006 12:13
> To: Drage, Keith (Keith); Francois Audet; Jonathan Rosenberg
> Cc: SIP WG; Dean Willis
> Subject: RE: [Sip] Re: SIPS and the PSTN
>=20
> I would like to have the chair's permission for the voting on=20
> the need of this document on the list. Who would like to see=20
> this document to progress ?
>=20
> Thx
> Samir=20
>=20
> >-----Original Message-----
> >From: Drage, Keith (Keith) [mailto:drage@lucent.com]
> >Sent: Wednesday, November 29, 2006 3:45 AM
> >To: Srivastava, Samir (SC100:8826); Audet, Francois (SC100:3055);=20
> >Jonathan Rosenberg
> >Cc: SIP WG; Dean Willis
> >Subject: RE: [Sip] Re: SIPS and the PSTN
> >
> >(As WG chair)
> >
> >If you wish the IETF rules allow you to do this.
> >
> >To make very clear the position:
> >
> >If you wish to achieve what you want, which is deprecation=20
> of various=20
> >requirements in RFC 3261, then you have already been asked=20
> to provide=20
> >the appropriate discussion material to this internet-draft.=20
> Part of the=20
> >output of the internet-draft is to make recommendations on=20
> improvement,=20
> >which can include deprecation. But we want to see the WG making an=20
> >informed decision, based on a clear understanding of what is=20
> the impact=20
> >on existing implementations, why the existing mechanism=20
> causes damage=20
> >in the community, and so on.
> >
> >The scope of the internet-draft is made very clear in the slides
> >(http://www3.ietf.org/proceedings/06nov/slides/sip-2.pdf) -=20
> see slide=20
> >2.
> >
> >Regards
> >
> >Keith
> >
> >> -----Original Message-----
> >> From: Samir Srivastava [mailto:samirsr@nortel.com]
> >> Sent: 28 November 2006 23:45
> >> To: Francois Audet; Jonathan Rosenberg
> >> Cc: SIP WG; Drage, Keith (Keith); Dean Willis
> >> Subject: RE: [Sip] Re: SIPS and the PSTN
> >>=20
> >>=20
> >> IMHO, after SIPS we are committing one more mistake. After knowing=20
> >> SIPS is completely broken, we are attempting to justify=20
> the need on=20
> >> the charter. I am failing to see the value of any these documents.
> >> Rather go to IESG directly and get this milestone removed from the=20
> >> charter. All this is merely wastage of cycles. BTW, I=20
> don't want to=20
> >> open the complete ranting mode now.
> >>=20
> >> Thx
> >> Samir
> >>=20
> >> > -----Original Message-----
> >> > From: Audet, Francois (SC100:3055)
> >> > Sent: Tuesday, November 28, 2006 3:23 PM
> >> > To: Jonathan Rosenberg
> >> > Cc: SIP WG; Drage, Keith (Keith); Dean Willis
> >> > Subject: RE: [Sip] Re: SIPS and the PSTN
> >> >=20
> >> > Yeah, except that I've been instructed by the Chairs that this
> >> document
> >> > is 'informational' only.
> >> >=20
> >> > I assume that it means that I can't make normative
> >> statements of any
> >> > kind. Is this true?
> >> >=20
> >>=20
> >>=20
> >
>=20

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From sip-bounces@ietf.org Mon Dec 04 01:32:09 2006
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Subject: [Sip] sip, doMessage
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hello,

im a newbie. im trying out some stuff with xlite. can anyone tell me
how to implement the doMessage method for instant messaging?

an example code would be very helpful. Im trying to implement it in
the findme example that comes with the weblogic sip server.

thank you in advance.

best regards,
Cal

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From sip-bounces@ietf.org Mon Dec 04 04:59:26 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
	-decide whether to send keep-alive in the first place 
Date: Mon, 4 Dec 2006 11:57:51 +0200
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Hi,

>No matter how we do the keep-alive, I think STUN is a good tool for the
>UA to figure out whether it's located behind a NAT in the first place

Please remember that UA could figure that out also with rport.
It should work both for UDP and TCP as well.

>by checking the XMA value etc. Again, outbound can be used also in
>no-NAT scenarios, so the UA can use the STUN response information to
>decide whether it needs to send keep-alives in the first place.

Generically STUN does not help for that, since keepalives are also
needed due to firewalls and STUN can not help the UA to detect the
presence of firewall.

In my understanding muxing STUN with SIP seemed as a good idea for
UDP flows. However if the value of UDP flows is rather questionable,
I would claim the same applies also to STUN keepalives, too.

Regards,

Erkki

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Subject: Re: [Sip] Issue: Expiration of temp-gruu
From: Scott Lawrence <slawrence@pingtel.com>
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On Wed, 2006-11-29 at 18:20 -0500, Paul Kyzivat wrote:

[assorted ambiguous cases that make it unclear whether or not a
temp-gruu is still valid]

> The "reg" event does provide a fairly complete solution to this problem, 
> for those that are willing and able to use it. It is not however a 100% 
> reliable solution. 

A UA that wants to determine whether or not a temp-gruu it has for
itself is still valid can do that test any time: send an OPTIONS request
to the domain of the gruu addressed to that temp-gruu.  If you get the
request back, it's still good.  No change is needed to gruu, and no
support for the reg event package is needed by either the UA or the
registrar.

-- 


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From sip-bounces@ietf.org Mon Dec 04 09:05:41 2006
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] strawman text for FASTEST proposal in outbound
Date: Mon, 4 Dec 2006 15:03:01 +0100
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The word "prevents" sounds like it is a feature and a good thing. It's 
more like a side effect. I propose changing that word to "prohibits".

Also, I want to second Markus on his comment. If outbound is supposed 
to work with presence and IM or any event package for that matter, then 
recommending not to use outbound is not a solution.

Hisham

On Nov 21, 2006, at 8:59 PM, Rohan Mahy wrote:

> Hi,
>
> To make the discussion about the outbound transport issues more
> concrete, here is some strawman text for the "FASTEST" direction.
>
> thanks,
> -rohan
>
> PROPOSED TEXT:
> Use of outbound with UDP flows prevents messages larger than 1300
> bytes from being delivered using TCP as described in Section 18.1.1
> RFC 3261.  Use of outbound in conjunction with UDP flows in
> environments where requests are expected to be larger than this size
> is therefore NOT RECOMMENDED.
>
> _______________________________________________
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> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Mon, 4 Dec 2006 15:31:52 +0100
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On Nov 20, 2006, at 2:23 PM, Aki Niemi wrote:

> ma, 2006-11-20 kello 10:02 +0100, ext Christer Holmberg (JO/LMF)
> kirjoitti:
>>>
>>> Please enumerate the cases.
>>>
>>> Unless there are concrete cases where TCP simply does not
>>> work, I would strongly recommend we drop UDP from outbound
>>> altogether. Remember, UDP already kind of works in presence
>>> of NAT; this is why we have for instance rport.
>>
>> Outbound is not an alternative to rport. Outbound provides you with
>> keep-alives, and multiple flows (in case one fails), which rport
>> doesn't. rport simply tells you where to send some stuff.
>
> I never said it was a replacement. However, the world is full of
> deployments that use UDP and work through NAT just fine (kind of); some
> use rport, some use something else. Clearly, these folks aren't holding
> out their deployments waiting for outbound to finish.

No, but we do so by breaking SIP rules. Something we want to move away 
from.

Hisham

>
>


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Subject: Re: [Sip] I-D on handling large UDP responses
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Hmm - clearly whatever Scott and I had agreed might work is probably  
irrelevant at this point since it clearly was not something we all  
agreed to or can even remember. It was just an hallway conversation  
between Scott and I. BTW, my vague recollection of it was I thought  
we had agreed that saying the response body would not be included  
unless it was less the total message was less than 1500 octets and  
provide advice on how to trim it down to be that size.

On Nov 20, 2006, at 9:05 AM, Scott Lawrence wrote:

> On Mon, 2006-11-13 at 11:20 +0100, Drage, Keith (Keith) wrote:
>> Question repeat, as it does not seem to have been answered.
>
>>> From: Drage, Keith (Keith) [mailto:drage@lucent.com]
>>> Sent: 04 October 2006 19:46
>>> To: Cullen Jennings; Dean Willis
>>> Cc: IETF SIP List; Vijay K. Gurbani
>>> Subject: RE: [Sip] I-D on handling large UDP responses
>>>
>>> Can you tell us what you agreed on. The proceedings seem to
>>> have missed it, and I don't remember it appearing on the list.
>
> I am not at all sure that we agreed on anything with respect to this.
>
> -- 
> Scott Lawrence  tel:+1-781-938-5306;ext=162 or  
> sip:slawrence@pingtel.com
>   sipXpbx project coordinator - SIPfoundry    http:// 
> www.sipfoundry.org/sipX
>   Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/

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I just checked the latest SIPit report, and it seems that only 5 
implementations supported reg event, as opposed to 29 that supported 
refer, 23 message-summary, or 14 presence.

Well, this probably indicates that reg-event is not widely deployed to 
justify a solution based on it. Therefore, I don't mind Jonathan's 
solution #2

/Miguel

Jonathan Rosenberg wrote:
> The main question is really to decide on one of two mechanisms:
> 
> 1. use reg-event if you are worried about use cases where this race 
> condition can occur (4-6 in Paul's original email)
> 
> 2. add a parameter to Contact in the register response. Specifically, 
> I'd recommend a Contact header field parameter which indicates the time 
> at which the first contact with a given instance ID was registered to 
> that AOR. This value gets linked with each temp gruu, and when the value 
> changes, all of those temp-gruu get invalidated.
> 
> Solution (2) is much simpler, but not needed if you think you'll always 
> have reg-event anyway. I don't think there is widespread support for 
> reg-event on endpoints yet. It is required by IMS.
> 
> I am somewhat on the fence on this one. I do worry that reg-event is 
> genreally a very heavy solution, and may not see much usage. Option 2 is 
> really trivially done, either in gruu or elsewhere. The main issue is 
> that it is another late change.
> 
> -JOnathan R.
> 
> 
> 
> Paul Kyzivat wrote:
> 
>> Miguel,
>>
>> So to summarize, it seems you are of the belief that the cases where 
>> these measures may be inadequate are sufficiently obscure to ignore, 
>> and that new measures to cover the obscure cases are unnecessary. Is 
>> that right?
>>
>>     Paul
>>
>> Miguel Garcia wrote:
>>
>>> For issue 3, I would say that the solution is to avoid the situation, 
>>> e.g., by registering way in advance before the registration expires. 
>>> Doesn't RFC 3261 recommends to re-register half time before the 
>>> registration timer expires.
>>>
>>> Issues 4, 5, and 6 are solved by the reg event. So, shouldn't the 
>>> GRUU draft have a recommendation on the usage of the reg event to 
>>> solve them?
>>>
>>> /Miguel
>>>
>>> Paul Kyzivat wrote:
>>>
>>>> After some study, I have encountered a potential difficulty in 
>>>> determining when temp-gruus expire. After discussing this with 
>>>> Jonathan we decided to bring it to the list for discussion.
>>>>
>>>> As now defined, when you first register you may get a temp-gruu, 
>>>> which will remain valid as long as the registration does not expire 
>>>> or be removed. (Removal is a special case of expiration.) Each time 
>>>> the UA refreshes the registration it may get a new temp-gruu which 
>>>> also remains valid for the life of the registration. The UA can 
>>>> accumulate all these temp-gruus and use them as it sees fit until 
>>>> they expire.
>>>>
>>>> The problem is: how does the UA know when the registration has ended 
>>>> and the temp-gruus associated with it rendered invalid?
>>>>
>>>> This seems like a simple question, and in some cases it is simple, 
>>>> but not in all cases. The following are at least some of the ways 
>>>> that the registration can be ended, causing the invalidation of 
>>>> temp-gruus:
>>>> 1) the UA that originally registered may explicitly deregister
>>>> 2) the UA the originally registered may simply permit the
>>>>    registration to expire without attempting to refresh it
>>>> 3) the UA may attempt to refresh the registration, but be too slow,
>>>>    so that the reregistration arrives after the prior one expires
>>>> 4) some other UA may deregister all the registrations, including
>>>>    the one by this UA. (When the UA decides it should refresh the
>>>>    registration it will actually create a new one.)
>>>> 5) the registrar may remove the registration administratively
>>>> 6) the registrar may crash and restart, losing all its registrations
>>>>
>>>> In each of these cases the previously assigned temp-gruus become 
>>>> invalid. If the UA continues to use them, bad things will happen. 
>>>> This is true both for calls that are active when the deregistration 
>>>> occurs, and calls that are established later. What kinds of bad 
>>>> things happen?
>>>> - requests addressed to the gruu, both in-dialog and out-of-dialog
>>>>   will be undeliverable. (For instance a reINVITE or BYE in an
>>>>   existing call, or an out-of-dialog INVITE/Replaces for a transfer.)
>>>>
>>>> - request originated by the UA using the invalid gruu will fail
>>>>   immediately because the gruu is invalid.
>>>>
>>>> - if the UA subsequently reregisters, receives an incoming call, and
>>>>   uses one of the old temp-gruus as its contact, then the call will
>>>>   be established, but subsequent in- and out-of-dialog requests to
>>>>   the contact will fail.
>>>>
>>>> So clearly it is important for the UA to stop using temp-gruus that 
>>>> have become invalid. How can it know to do that in all of the above 
>>>> cases?
>>>>
>>>> 1) is straightforward. The UA knows it is deregistering, and should
>>>>    discard any cached temp-gruus. If it has a dialog with one of the
>>>>    temp-gruus as a contact, then it should do a target refresh and
>>>>    supply some working contact. (It really should do this *before*
>>>>    deregistering, and for a pub-gruu as well.)
>>>>
>>>> 2) the UA should take all the actions mentioned for (1) at or before
>>>>    the expiration time. This should be done a bit early to guard
>>>>    against clock skew problems.
>>>>
>>>> 3) in this case, it may be impossible for the UA to know whether
>>>>    the REGISTER arrived in time to be a reREGISTER or if it arrived
>>>>    late and was treated as a new REGISTER after the expiration of
>>>>    the old one. (The response to the REGISTER includes nothing that
>>>>    distinguishes the two cases.) About the best that can be done in
>>>>    this case is to check the time at which the response to the
>>>>    REGISTER is received. If it falls after the expected expiration
>>>>    time for the registration, act as if the old registration had
>>>>    expired.
>>>>
>>>> 4) This is a major problem. Of course, regardless of GRUU, the UA
>>>>    won't be able to receive incoming calls until it decides it is
>>>>    time to refresh the registration. In addition, until the
>>>>    registration reestablished, calls originated with one of its
>>>>    existing gruus will fail. Once the registration is reestablished,
>>>>    it still won't know its old temp-gruus are invalid. The only
>>>>    remedy for this is to subscribe to the "reg" event package,
>>>>    and use the resulting notifications to identify when it has
>>>>    been unregistered. If it does that, it can invalidate all its
>>>>    old temp-gruus, and then presumably reregister.
>>>>
>>>> 5) this is functionally equivalent to (4), except that perhaps
>>>>    attempts to reregister may fail.
>>>>
>>>> 6) this is also functionally equivalent to (4).
>>>>
>>>> The "reg" event does provide a fairly complete solution to this 
>>>> problem, for those that are willing and able to use it. It is not 
>>>> however a 100% reliable solution. Consider case (3), where the UA 
>>>> has a reg event subscription. All is well if the registration 
>>>> expires and a notification is sent indicating that, and then the 
>>>> reregistration occurs and a notification is sent about that. But if 
>>>> they happen very closely in time, it may be that only one 
>>>> notification is sent, reflecting the result of the reregistration. 
>>>> In that case the UA will not be aware that it had been unregistered 
>>>> for a period of time.
>>>>
>>>> A *complete* solution would, IMO, involve changing REGISTER so that 
>>>> it returns an indication of whether this was a new registration, or 
>>>> a refresh. This could take many forms. For instance the timestamp of 
>>>> the initial registration could always be returned with each contact, 
>>>> as well as the expiration time.
>>>>
>>>> The big question is whether this is a big enough issue to require a 
>>>> solution now, as part of GRUU, or if it can be ignored as 
>>>> insignificant or postponed to future work.
>>>>
>>>> I'd appreciate hearing what others think about this. At the moment I 
>>>> am leaning toward doing nothing, at least for now.
>>>>
>>>>     Thanks,
>>>>     Paul
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>
>>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Mon, 4 Dec 2006 15:57:44 +0100
To: "Jeroen van Bemmel" <jbemmel@zonnet.nl>
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I like this proposal, although I would not use MUST. Either of these is 
optional. Both is recommended.

Hisham

On Nov 17, 2006, at 6:20 PM, Jeroen van Bemmel wrote:

> The outbound draft needs to say: "if a UA establishes a flow using an 
> unreliable transport, it MUST also establish one over a reliable 
> transport" (to support switch over). Moreover, I would suggest to add 
> "it is RECOMMENDED that the UA establishes a flow using a reliable 
> transport"
>
> Regards,
> Jeroen
>
>


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Subject: Re: [Sip] I-D on handling large UDP responses
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On Mon, 2006-12-04 at 06:39 -0800, Cullen Jennings wrote:
> Hmm - clearly whatever Scott and I had agreed might work is probably  
> irrelevant at this point since it clearly was not something we all  
> agreed to or can even remember. It was just an hallway conversation  
> between Scott and I. BTW, my vague recollection of it was I thought  
> we had agreed that saying the response body would not be included  
> unless it was less the total message was less than 1500 octets and  
> provide advice on how to trim it down to be that size.

That matches my recollection regarding the handling of hop limit
responses (and any other similar diagnostic responses), but I didn't
think that was the current question.  

I was referring to a general rule for what to do with _any_ large
response on a path that includes UDP.  My recollection is that there are
essentially three points of view on this:

      * Do Nothing - let UDP either work or not (there seemed to be
        important disagreement on how often this is a problem), and if
        it doesn't work, don't do anything different to make it easier
        to detect why not.

      * Diagnostic Provisional Response - add a small 1xx response or
        responses (as described in
        draft-gurbani-sip-large-udp-response-00) to aid in diagnosis of
        failures.  This ran into objections from some people with
        respect to non-INVITE transactions, and from others who seemed
        to object on broader grounds.

      * Always Return Error - instead of returning whatever final
        response you were going to return (whether positive or
        negative), return an error indicating that the response would be
        too large.

Alas, I think we've only gotten as far as the 'Rough' in 'Rough
Consensus' on this.

Obviously, I prefer some form of Diagnostic Provisional response.  I do
believe this is a significant problem.  I prefer the diagnositc approach
because it allows things to work if the large UDP response actually gets
through, but gives you some hint as to why not when it doesn't.   I
think that the relative cost of the extra 1xx is acceptable when
compared to the cost of the retransmissions when things are broken.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/


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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
	-decide whether to send keep-alive in the first place 
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>No matter how we do the keep-alive, I think STUN is a good 
>tool for the UA to figure out whether it's located behind a 
>NAT in the first place, by checking the XMA value etc. Again, 
>outbound can be used also in no-NAT scenarios, so the UA can 
>use the STUN response information to decide whether it needs 
>to send keep-alives in the first place.

I disagree; that will generally not work.

Many networks deploy stateful firewalls that do connection tracking. They
will not show up as NAT, but will require keepalives. I've seen these
especially in mobile networks, but I would imagine many enterprises deploy
similar stuff.

My experience has been that if you want robust connectivity for your SIP
app, use TCP (you get 30x more performance from middleboxes) and *always*
use keepalives, with their frequency tuned towards the lowest common
keepalive interval found in the field.

Clearly, YMMV.

Cheers,
Aki


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Date: Mon, 04 Dec 2006 16:49:18 +0200
From: Miguel Garcia <Miguel.An.Garcia@nokia.com>
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To: Paul Kyzivat <pkyzivat@cisco.com>
Subject: Re: [Sip] Issue: Expiration of temp-gruu
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Correct. I think so.

/Miguel

Paul Kyzivat wrote:
> Miguel,
> 
> So to summarize, it seems you are of the belief that the cases where 
> these measures may be inadequate are sufficiently obscure to ignore, and 
> that new measures to cover the obscure cases are unnecessary. Is that 
> right?
> 
>     Paul
> 
> Miguel Garcia wrote:
>> For issue 3, I would say that the solution is to avoid the situation, 
>> e.g., by registering way in advance before the registration expires. 
>> Doesn't RFC 3261 recommends to re-register half time before the 
>> registration timer expires.
>>
>> Issues 4, 5, and 6 are solved by the reg event. So, shouldn't the GRUU 
>> draft have a recommendation on the usage of the reg event to solve them?
>>
>> /Miguel
>>
>> Paul Kyzivat wrote:
>>> After some study, I have encountered a potential difficulty in 
>>> determining when temp-gruus expire. After discussing this with 
>>> Jonathan we decided to bring it to the list for discussion.
>>>
>>> As now defined, when you first register you may get a temp-gruu, 
>>> which will remain valid as long as the registration does not expire 
>>> or be removed. (Removal is a special case of expiration.) Each time 
>>> the UA refreshes the registration it may get a new temp-gruu which 
>>> also remains valid for the life of the registration. The UA can 
>>> accumulate all these temp-gruus and use them as it sees fit until 
>>> they expire.
>>>
>>> The problem is: how does the UA know when the registration has ended 
>>> and the temp-gruus associated with it rendered invalid?
>>>
>>> This seems like a simple question, and in some cases it is simple, 
>>> but not in all cases. The following are at least some of the ways 
>>> that the registration can be ended, causing the invalidation of 
>>> temp-gruus:
>>> 1) the UA that originally registered may explicitly deregister
>>> 2) the UA the originally registered may simply permit the
>>>    registration to expire without attempting to refresh it
>>> 3) the UA may attempt to refresh the registration, but be too slow,
>>>    so that the reregistration arrives after the prior one expires
>>> 4) some other UA may deregister all the registrations, including
>>>    the one by this UA. (When the UA decides it should refresh the
>>>    registration it will actually create a new one.)
>>> 5) the registrar may remove the registration administratively
>>> 6) the registrar may crash and restart, losing all its registrations
>>>
>>> In each of these cases the previously assigned temp-gruus become 
>>> invalid. If the UA continues to use them, bad things will happen. 
>>> This is true both for calls that are active when the deregistration 
>>> occurs, and calls that are established later. What kinds of bad 
>>> things happen?
>>> - requests addressed to the gruu, both in-dialog and out-of-dialog
>>>   will be undeliverable. (For instance a reINVITE or BYE in an
>>>   existing call, or an out-of-dialog INVITE/Replaces for a transfer.)
>>>
>>> - request originated by the UA using the invalid gruu will fail
>>>   immediately because the gruu is invalid.
>>>
>>> - if the UA subsequently reregisters, receives an incoming call, and
>>>   uses one of the old temp-gruus as its contact, then the call will
>>>   be established, but subsequent in- and out-of-dialog requests to
>>>   the contact will fail.
>>>
>>> So clearly it is important for the UA to stop using temp-gruus that 
>>> have become invalid. How can it know to do that in all of the above 
>>> cases?
>>>
>>> 1) is straightforward. The UA knows it is deregistering, and should
>>>    discard any cached temp-gruus. If it has a dialog with one of the
>>>    temp-gruus as a contact, then it should do a target refresh and
>>>    supply some working contact. (It really should do this *before*
>>>    deregistering, and for a pub-gruu as well.)
>>>
>>> 2) the UA should take all the actions mentioned for (1) at or before
>>>    the expiration time. This should be done a bit early to guard
>>>    against clock skew problems.
>>>
>>> 3) in this case, it may be impossible for the UA to know whether
>>>    the REGISTER arrived in time to be a reREGISTER or if it arrived
>>>    late and was treated as a new REGISTER after the expiration of
>>>    the old one. (The response to the REGISTER includes nothing that
>>>    distinguishes the two cases.) About the best that can be done in
>>>    this case is to check the time at which the response to the
>>>    REGISTER is received. If it falls after the expected expiration
>>>    time for the registration, act as if the old registration had
>>>    expired.
>>>
>>> 4) This is a major problem. Of course, regardless of GRUU, the UA
>>>    won't be able to receive incoming calls until it decides it is
>>>    time to refresh the registration. In addition, until the
>>>    registration reestablished, calls originated with one of its
>>>    existing gruus will fail. Once the registration is reestablished,
>>>    it still won't know its old temp-gruus are invalid. The only
>>>    remedy for this is to subscribe to the "reg" event package,
>>>    and use the resulting notifications to identify when it has
>>>    been unregistered. If it does that, it can invalidate all its
>>>    old temp-gruus, and then presumably reregister.
>>>
>>> 5) this is functionally equivalent to (4), except that perhaps
>>>    attempts to reregister may fail.
>>>
>>> 6) this is also functionally equivalent to (4).
>>>
>>> The "reg" event does provide a fairly complete solution to this 
>>> problem, for those that are willing and able to use it. It is not 
>>> however a 100% reliable solution. Consider case (3), where the UA has 
>>> a reg event subscription. All is well if the registration expires and 
>>> a notification is sent indicating that, and then the reregistration 
>>> occurs and a notification is sent about that. But if they happen very 
>>> closely in time, it may be that only one notification is sent, 
>>> reflecting the result of the reregistration. In that case the UA will 
>>> not be aware that it had been unregistered for a period of time.
>>>
>>> A *complete* solution would, IMO, involve changing REGISTER so that 
>>> it returns an indication of whether this was a new registration, or a 
>>> refresh. This could take many forms. For instance the timestamp of 
>>> the initial registration could always be returned with each contact, 
>>> as well as the expiration time.
>>>
>>> The big question is whether this is a big enough issue to require a 
>>> solution now, as part of GRUU, or if it can be ignored as 
>>> insignificant or postponed to future work.
>>>
>>> I'd appreciate hearing what others think about this. At the moment I 
>>> am leaning toward doing nothing, at least for now.
>>>
>>>     Thanks,
>>>     Paul
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>>

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

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From: "Aki Niemi" <aki.niemi@nokia.com>
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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+1 for #1 

>-----Original Message-----
>From: ext sip-bounces@ietf.org [mailto:sip-bounces@ietf.org] 
>Sent: 01 December, 2006 14:12
>To: rohan.mahy@gmail.com
>Cc: sip@ietf.org; rohan@ekabal.com
>Subject: RE: [Sip] Re: issue with oubound and the used 
>transport protocol
>
>
>Hi,
>
>You asked me to clarify my CRLF proposal, so I try to do it here.
>
>I have been exploring ways of rescoping drafts related to 
>Outbound and simplifying the mechanisms within them since the 
>limitations of using UDP flows became apparent in this thread. 
>I could vote for a number of different approaches and would be 
>happy the WG to agree with at least one of them. Those 
>approaches in my preference order are:
>
>1. Outbound draft to re-scope itself to "Outbound TCP connection"
>   draft without any support to UDP, STUN or multiple registrations.
>   Multiple registrations and mid-dialog failover (using either TCP
>   or UDP) to be specified within another draft.
>
>   - This proposal as such did not raise any discussion 
>     on the list, probably because it would be a dramatic
>     change at a quite late stage of the draft.
>   - Remarkable changes needed (but I already outlined the
>     "Outbound TCP connection" draft within my earlier mail :).
>     http://www1.ietf.org/mail-archive/web/sip/current/msg17495.html
>
>2. Outbound draft to replace STUN keepalive over TCP with CRLF
>   keepalive over TCP. (No impact to UDP support of Outbound)
>
>   - This option is equivalent to your formulation, which I
>     can very much agree on:
>
>> Clearly you are suggesting that we make CRLF mandatory to implement 
>> for TCP.  There would be no longer any need to do STUN over TCP and 
>> therefore no need implement code to deal with stream-based demuxing.
>> The text about sigcomp over TCP and STUN over TLS over TCP 
>could also 
>> go away.
>
>   - Some WG members already expressed their support for this
>     option
>   - Minor changes needed to the draft
>
>3. Outbound draft to introduce a third optional keepalive method
>   for TCP: CRLF (in addition to STUN and TCP keepalives)
>
>   - This option would be the _last resort_ to get at least some
>     benefit for TCP-based UA implementation, even if it would
>     not help the proxies. Proxies would still have to support
>     STUN over TCP. 
>   - Minimal changes needed to the draft
>
>
>> To clarify are you suggesting that an edge proxy can implement 
>> outbound without support for UDP flow tokens, without 
>support for STUN 
>> keepalives over UDP, both, or neither?
>
>My proposals were born "UA centric" as I am from the UA vendor camp.
>
>Anyway if Outbound will specify both UDP and TCP as transport 
>options, I would believe the proxies still have to support 
>both since the selection of the transport is basically an UA 
>decision (even if assisted by DNS SRV) as it depends on if the 
>proxy URI configured to the UA has a transport parameter. If 
>the proxy would not support UDP & STUN, how could it tell it 
>to the UA ? Outbound would at least need a mechanism for that 
>e.g. a specific response code, option tag or something else. 
>Or is this yet another thing where Outbound would rely on the 
>UA and DNS configuration to tell the UA to avoid using UDP ?
>
>Considering the option to drop UDP & STUN support from Outbound
>draft: I would agree that combination of UDP and STUN 
>keepalives does not seem to provide a practical way for NAT 
>traversal. Keeping UDP flows alive with STUN seems not to add 
>value in comparison to persistent TCP flows and I could agree 
>dropping UDP flows from the draft. The only problem I see here 
>that Outbound covers also the multiregistration mechanism, 
>which could still be useful with UDP e.g. in cellular access 
>context without NAT/FWs. That is why I have been proposing 
>re-scoping of the drafts:
>
>- Flow keepalive mechanisms to be specified only for TCP (one draft)
>- Protocol agnostic multiregistration mechanism with mid-dialog
>  failover support (another draft)
>
>Regards,
>
>Erkki
>
>> -----Original Message-----
>> From: ext Rohan Mahy [mailto:rohan.mahy@gmail.com]
>> Sent: 29 November, 2006 20:30
>> To: Koivusalo Erkki (Nokia-TP-MSW/Helsinki)
>> Cc: sip@ietf.org; rohan@ekabal.com
>> Subject: Re: [Sip] Re: issue with oubound and the used transport 
>> protocol
>> 
>> 
>> Hi Erkki,
>> 
>> Clearly you are suggesting that we make CRLF mandatory to implement 
>> for TCP.  There would be no longer any need to do STUN over TCP and 
>> therefore no need implement code to deal with stream-based demuxing.
>>   The text about sigcomp over TCP and STUN over TLS over TCP could 
>> also go away.
>> 
>> To clarify are you suggesting that an edge proxy can implement 
>> outbound without support for UDP flow tokens, without 
>support for STUN 
>> keepalives over UDP, both, or neither?
>> 
>> thanks,
>> -rohan
>> 
>> On 11/29/06, Erkki.Koivusalo@nokia.com <Erkki.Koivusalo@nokia.com> 
>> wrote:
>> >
>> > Hi,
>> >
>> > >about STUN keepalive usage and discovery, since double
>> CRLF is fully
>> > >backwards compatible with even RFC 2543 implementations.  I am 
>> > >bringing this up on the list because I hope the WG can 
>look back on 
>> > >this and not make the mistake of snowballing complexity 
>to preserve 
>> > >UDP parity again in the future.
>> >
>> > Rohan is making a valid point here and I believe in Outbound there 
>> > is still room for change to cater for CRLF keepalives as an option 
>> > to STUN keepalives over TCP.
>> >
>> > I am calling the WG to look back on the keepalive decision made 
>> > earlier for TCP and proposing that Outbound should also cover CRLF 
>> > keepalive option.
>> >
>> > Reasons for this proposal:
>> >
>> > >The problem I see is that we have two separate communities
>> using the
>> > >outbound draft:
>> > >
>> > >1) folks that want to get through NATs and Firewalls in 
>the public 
>> > >Internet at large - they want to use TCP and TLS with
>> double CRLF for
>> > >keepalives. One "flow" is always one connection.  The 
>milestone for 
>> > >this would read "Make TCP and TLS work in the Internet".
>> >
>> > I agree with the intents of this community. However this 
>is not what 
>> > the current Outbound provides. Outbound specifies two options for 
>> > keepalives for reliable connections: TCP keepalive and STUN.
>> >
>> > These options were agreed in IETF 65, a cite from the minutes:
>> >
>> > Proposal:
>> >    Do STUN for TCP and UDP
>> >    Do CRLF for TCP and STUN for UDP
>> >    TCP KeepAlive if supported, 1 or 2 otherwise
>> >
>> > Resolutions:
>> >    OS TCP KeepAlive should be used if the client supports it.
>> >    Need something more because OS TCP KeepAlive is not 
>supported in 
>> > every
>> >    client OS.
>> >    Consensus to use STUN for UDP
>> >    Consensus to use STUN for TCP, with use of OS TCP keep
>> alive above.
>> >
>> > In that meeting the basic assumption was that every Outbound 
>> > compliant client should implement both UDP and TCP even if that is 
>> > not explicitely mentioned in the draft. However latest -06 version 
>> > now makes that debatable by stating:
>> >
>> >    Use of outbound with UDP flows typically prevents 
>messages larger
>> >    than 1300 bytes from being delivered using TCP as described in
>> >    Section 18.1.1 RFC 3261.  Use of outbound in 
>conjunction with UDP
>> >    flows in environments where requests are expected to be
>> larger than
>> >    this size is therefore NOT RECOMMENDED.
>> >
>> > In practice this seems to justify implementing "TCP only" UAs what 
>> > comes to the Outbound support. In that case the basic
>> assumption
>> > with STUN keepalive (implement it for TCP as it is anyway 
>needed for 
>> > UDP as well) would become obsolete. For implementations which can 
>> > not use TCP keepalive, CRLF would obviously provide a 
>great benefit 
>> > without significant increase of complexity to Outbound.
>> >
>> >
>> > P.S. Last week I made a proposal for "stripped Outbound" without 
>> > UDP, STUN keepalives and multiple registrations. As that proposal 
>> > did not receive any comments, I decided to propose this minimal 
>> > addition to Outbound as an alternative that might be a bit 
>easier to 
>> > accept but still achieve some of the benefits.
>> >
>> > Regards,
>> >
>> > Erkki
>> >
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol Use 
>> > sip-implementors@cs.columbia.edu for questions on current sip Use 
>> > sipping@ietf.org for new developments on the application of sip
>> >
>> 
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use 
>sip-implementors@cs.columbia.edu for questions on current sip 
>Use sipping@ietf.org for new developments on the application of sip
>


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Subject: RE: [Sip] Re: issue with oubound and the used transport
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Hi Aki,
=20
You are correct regarding firewalls. I was only thinking about NATs.
=20
Regards,
=20
Christer
=20
=20

________________________________

From: Aki Niemi [mailto:aki.niemi@nokia.com]
Sent: Mon 04/12/2006 11:26
To: Christer Holmberg (JO/LMF); 'Francois Audet'; 'Kevin Johns'; =
'Fredrik Thulin'
Cc: sip@ietf.org; rohan@ekabal.com
Subject: RE: [Sip] Re: issue with oubound and the used transport =
protocol-decide whether to send keep-alive in the first place=20



>No matter how we do the keep-alive, I think STUN is a good
>tool for the UA to figure out whether it's located behind a
>NAT in the first place, by checking the XMA value etc. Again,
>outbound can be used also in no-NAT scenarios, so the UA can
>use the STUN response information to decide whether it needs
>to send keep-alives in the first place.

I disagree; that will generally not work.

Many networks deploy stateful firewalls that do connection tracking. =
They
will not show up as NAT, but will require keepalives. I've seen these
especially in mobile networks, but I would imagine many enterprises =
deploy
similar stuff.

My experience has been that if you want robust connectivity for your SIP
app, use TCP (you get 30x more performance from middleboxes) and =
*always*
use keepalives, with their frequency tuned towards the lowest common
keepalive interval found in the field.

Clearly, YMMV.

Cheers,
Aki


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Use sip-implementors@cs.columbia.edu for questions on current sip
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Scott,

Its certainly true that a gruu can always be tested to see if it works. 
That would however raise the question of when one ought to do the test. 
In normal cases you will know the status of your gruu by regular 
methods, so you would have no reason to test it. So you would have to 
test whenever you thought failure might have negative consequences. Some 
of the times when failure might have bad consequences are: when 
responding to an incoming INVITE, and any time during a dialog. I guess 
you could send an OPTIONS and await the response before responding to an 
INVITE, and it might even be masked by alerting, but it seems kind of 
annoying to have to do that. But no matter how often you do it during a 
dialog you still might miss the moment your gruu becomes invalid and the 
other end of the dialog sends you a message. (Extremely unlikely, but 
possible.)

I think the other existing solutions get "pretty close", and OPTIONS 
doesn't significantly improve on that.

	Paul

Scott Lawrence wrote:
> On Wed, 2006-11-29 at 18:20 -0500, Paul Kyzivat wrote:
> 
> [assorted ambiguous cases that make it unclear whether or not a
> temp-gruu is still valid]
> 
>> The "reg" event does provide a fairly complete solution to this problem, 
>> for those that are willing and able to use it. It is not however a 100% 
>> reliable solution. 
> 
> A UA that wants to determine whether or not a temp-gruu it has for
> itself is still valid can do that test any time: send an OPTIONS request
> to the domain of the gruu addressed to that temp-gruu.  If you get the
> request back, it's still good.  No change is needed to gruu, and no
> support for the reg event package is needed by either the UA or the
> registrar.
> 

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This is a multi-part message in MIME format.

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Hi all,

The SIPS Guidelines Working Group draft has just been issued.

All the changes agreed to in the mailing list and in San Diego, as well
as  have been=20
incorporated.=20

The process as outlined in San Diego has been followed:

- The document is now an Informational document that clarifies RFC 3261
(as opposed
  to modify it).

- There is a new section 3.1 that attempts to capture the consensus of
the group
  that SIPS is valid within SIP. What happens outside of SIP (e.g.,
  PSTN) is out-of-scope. People can review that section and comment on
it. That was
  the agreement in response to Dean's Point #4.

- There is a new section 3.2 clarifying Best-effort use of TLS (i.e.,
don't use SIPS
  for best-effort just use SIP transported over TLS). That was Dean's
Point #3.

- Section 4.1's title has been changed to make it obvious that SIPS is
hop-by-hop
  and not end-to-end.

- Appendix A has been added to discuss Further Steps in Specification.
This section
  consist of brief discussion of areas that we believe may eventually
require
  further specifications. It is expected that this section will require
beefing-up.
  Please comment on it. It currently list 2 areas:

  - Deprecation of Last Hop Exception, reflecting the agreement from the
San Diego
    meeting. That was Dean's point #2.

  - True end-to-end security.

- Appendix B "To be done": Please have a look at this section. =20


> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20
> Sent: Monday, December 04, 2006 12:50
> To: i-d-announce@ietf.org
> Subject: I-D ACTION:draft-ietf-sip-sips-00.txt
>=20
> A New Internet-Draft is available from the on-line=20
> Internet-Drafts directories.
>=20
>=20
> 	Title		: Guidelines for the use of the SIPS=20
> URI Scheme in the Session Initiation Protocol (SIP)
> 	Author(s)	: F. Audet
> 	Filename	: draft-ietf-sip-sips-00.txt
> 	Pages		: 39
> 	Date		: 2006-12-4
> =09
>    This document providing clarifications and guidelines=20
> concerning the
>    use of SIPS URI scheme in the Session Initiation Protocol=20
> (SIP).  It
>    does not make normative changes to SIP.  This document=20
> also provides
>    discussion of possible future steps in specification, and the pros
>    and cons of making such changes.
>=20
>=20
>=20
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-sip-sips-00.txt
>=20
> To remove yourself from the I-D Announcement list, send a message to=20
> i-d-announce-request@ietf.org with the word unsubscribe in=20
> the body of=20
> the message.=20
> You can also visit=20
> https://www1.ietf.org/mailman/listinfo/I-D-announce=20
> to change your subscription settings.
>=20
> Internet-Drafts are also available by anonymous FTP. Login with the=20
> username "anonymous" and a password of your e-mail address. After=20
> logging in, type "cd internet-drafts" and then=20
> "get draft-ietf-sip-sips-00.txt".
>=20
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html=20
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>=20
> Internet-Drafts can also be obtained by e-mail.
>=20
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-sip-sips-00.txt".
> =09
> NOTE:	The mail server at ietf.org can return the document in
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>=20
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>=20

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_______________________________________________
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Would a third party observing multiple uses of temp-gruu be able to  
correlate them from this time-stamp and therefore make a reasonable  
deduction that all the temp-gruus were bound to the same Contact and  
therefore used by the same user?

Would we care?

--
Dean


On Dec 4, 2006, at 8:54 AM, Miguel Garcia wrote:

> I just checked the latest SIPit report, and it seems that only 5  
> implementations supported reg event, as opposed to 29 that  
> supported refer, 23 message-summary, or 14 presence.
>
> Well, this probably indicates that reg-event is not widely deployed  
> to justify a solution based on it. Therefore, I don't mind  
> Jonathan's solution #2
>
> /Miguel
>
> Jonathan Rosenberg wrote:
>> The main question is really to decide on one of two mechanisms:
>> 1. use reg-event if you are worried about use cases where this  
>> race condition can occur (4-6 in Paul's original email)
>> 2. add a parameter to Contact in the register response.  
>> Specifically, I'd recommend a Contact header field parameter which  
>> indicates the time at which the first contact with a given  
>> instance ID was registered to that AOR. This value gets linked  
>> with each temp gruu, and when the value changes, all of those temp- 
>> gruu get invalidated.
>> Solution (2) is much simpler, but not needed if you think you'll  
>> always have reg-event anyway. I don't think there is widespread  
>> support for reg-event on endpoints yet. It is required by IMS.
>> I am somewhat on the fence on this one. I do worry that reg-event  
>> is genreally a very heavy solution, and may not see much usage.  
>> Option 2 is really trivially done, either in gruu or elsewhere.  
>> The main issue is that it is another late change.
>> -JOnathan R.
>> Paul Kyzivat wrote:
>>> Miguel,
>>>
>>> So to summarize, it seems you are of the belief that the cases  
>>> where these measures may be inadequate are sufficiently obscure  
>>> to ignore, and that new measures to cover the obscure cases are  
>>> unnecessary. Is that right?
>>>
>>>     Paul
>>>
>>> Miguel Garcia wrote:
>>>
>>>> For issue 3, I would say that the solution is to avoid the  
>>>> situation, e.g., by registering way in advance before the  
>>>> registration expires. Doesn't RFC 3261 recommends to re-register  
>>>> half time before the registration timer expires.
>>>>
>>>> Issues 4, 5, and 6 are solved by the reg event. So, shouldn't  
>>>> the GRUU draft have a recommendation on the usage of the reg  
>>>> event to solve them?
>>>>
>>>> /Miguel
>>>>
>>>> Paul Kyzivat wrote:
>>>>
>>>>> After some study, I have encountered a potential difficulty in  
>>>>> determining when temp-gruus expire. After discussing this with  
>>>>> Jonathan we decided to bring it to the list for discussion.
>>>>>
>>>>> As now defined, when you first register you may get a temp- 
>>>>> gruu, which will remain valid as long as the registration does  
>>>>> not expire or be removed. (Removal is a special case of  
>>>>> expiration.) Each time the UA refreshes the registration it may  
>>>>> get a new temp-gruu which also remains valid for the life of  
>>>>> the registration. The UA can accumulate all these temp-gruus  
>>>>> and use them as it sees fit until they expire.
>>>>>
>>>>> The problem is: how does the UA know when the registration has  
>>>>> ended and the temp-gruus associated with it rendered invalid?
>>>>>
>>>>> This seems like a simple question, and in some cases it is  
>>>>> simple, but not in all cases. The following are at least some  
>>>>> of the ways that the registration can be ended, causing the  
>>>>> invalidation of temp-gruus:
>>>>> 1) the UA that originally registered may explicitly deregister
>>>>> 2) the UA the originally registered may simply permit the
>>>>>    registration to expire without attempting to refresh it
>>>>> 3) the UA may attempt to refresh the registration, but be too  
>>>>> slow,
>>>>>    so that the reregistration arrives after the prior one expires
>>>>> 4) some other UA may deregister all the registrations, including
>>>>>    the one by this UA. (When the UA decides it should refresh the
>>>>>    registration it will actually create a new one.)
>>>>> 5) the registrar may remove the registration administratively
>>>>> 6) the registrar may crash and restart, losing all its  
>>>>> registrations
>>>>>
>>>>> In each of these cases the previously assigned temp-gruus  
>>>>> become invalid. If the UA continues to use them, bad things  
>>>>> will happen. This is true both for calls that are active when  
>>>>> the deregistration occurs, and calls that are established  
>>>>> later. What kinds of bad things happen?
>>>>> - requests addressed to the gruu, both in-dialog and out-of-dialog
>>>>>   will be undeliverable. (For instance a reINVITE or BYE in an
>>>>>   existing call, or an out-of-dialog INVITE/Replaces for a  
>>>>> transfer.)
>>>>>
>>>>> - request originated by the UA using the invalid gruu will fail
>>>>>   immediately because the gruu is invalid.
>>>>>
>>>>> - if the UA subsequently reregisters, receives an incoming  
>>>>> call, and
>>>>>   uses one of the old temp-gruus as its contact, then the call  
>>>>> will
>>>>>   be established, but subsequent in- and out-of-dialog requests to
>>>>>   the contact will fail.
>>>>>
>>>>> So clearly it is important for the UA to stop using temp-gruus  
>>>>> that have become invalid. How can it know to do that in all of  
>>>>> the above cases?
>>>>>
>>>>> 1) is straightforward. The UA knows it is deregistering, and  
>>>>> should
>>>>>    discard any cached temp-gruus. If it has a dialog with one  
>>>>> of the
>>>>>    temp-gruus as a contact, then it should do a target refresh and
>>>>>    supply some working contact. (It really should do this *before*
>>>>>    deregistering, and for a pub-gruu as well.)
>>>>>
>>>>> 2) the UA should take all the actions mentioned for (1) at or  
>>>>> before
>>>>>    the expiration time. This should be done a bit early to guard
>>>>>    against clock skew problems.
>>>>>
>>>>> 3) in this case, it may be impossible for the UA to know whether
>>>>>    the REGISTER arrived in time to be a reREGISTER or if it  
>>>>> arrived
>>>>>    late and was treated as a new REGISTER after the expiration of
>>>>>    the old one. (The response to the REGISTER includes nothing  
>>>>> that
>>>>>    distinguishes the two cases.) About the best that can be  
>>>>> done in
>>>>>    this case is to check the time at which the response to the
>>>>>    REGISTER is received. If it falls after the expected expiration
>>>>>    time for the registration, act as if the old registration had
>>>>>    expired.
>>>>>
>>>>> 4) This is a major problem. Of course, regardless of GRUU, the UA
>>>>>    won't be able to receive incoming calls until it decides it is
>>>>>    time to refresh the registration. In addition, until the
>>>>>    registration reestablished, calls originated with one of its
>>>>>    existing gruus will fail. Once the registration is  
>>>>> reestablished,
>>>>>    it still won't know its old temp-gruus are invalid. The only
>>>>>    remedy for this is to subscribe to the "reg" event package,
>>>>>    and use the resulting notifications to identify when it has
>>>>>    been unregistered. If it does that, it can invalidate all its
>>>>>    old temp-gruus, and then presumably reregister.
>>>>>
>>>>> 5) this is functionally equivalent to (4), except that perhaps
>>>>>    attempts to reregister may fail.
>>>>>
>>>>> 6) this is also functionally equivalent to (4).
>>>>>
>>>>> The "reg" event does provide a fairly complete solution to this  
>>>>> problem, for those that are willing and able to use it. It is  
>>>>> not however a 100% reliable solution. Consider case (3), where  
>>>>> the UA has a reg event subscription. All is well if the  
>>>>> registration expires and a notification is sent indicating  
>>>>> that, and then the reregistration occurs and a notification is  
>>>>> sent about that. But if they happen very closely in time, it  
>>>>> may be that only one notification is sent, reflecting the  
>>>>> result of the reregistration. In that case the UA will not be  
>>>>> aware that it had been unregistered for a period of time.
>>>>>
>>>>> A *complete* solution would, IMO, involve changing REGISTER so  
>>>>> that it returns an indication of whether this was a new  
>>>>> registration, or a refresh. This could take many forms. For  
>>>>> instance the timestamp of the initial registration could always  
>>>>> be returned with each contact, as well as the expiration time.
>>>>>
>>>>> The big question is whether this is a big enough issue to  
>>>>> require a solution now, as part of GRUU, or if it can be  
>>>>> ignored as insignificant or postponed to future work.
>>>>>
>>>>> I'd appreciate hearing what others think about this. At the  
>>>>> moment I am leaning toward doing nothing, at least for now.
>>>>>
>>>>>     Thanks,
>>>>>     Paul
>>>>>
>>>>> _______________________________________________
>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sipping@ietf.org for new developments on the application of  
>>>>> sip
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip
>>>
>
> -- 
> Miguel A. Garcia           tel:+358-50-4804586
> sip:miguel.garcia@neonsite.net
> Nokia Research Center      Helsinki, Finland
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


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From sip-bounces@ietf.org Tue Dec 05 04:06:28 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 11:04:05 +0200
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From: <Markus.Isomaki@nokia.com>
To: <hisham.khartabil@telio.no>, <jbemmel@zonnet.nl>
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Hi,

I don't see the point in this proposal. If the UA can establish a
flow/connection for a reliable transport (TCP), why would it need the
unreliable transport (UDP) flow in the first place? All traffic could be
sent over TCP anyway.=20

Markus =20

>-----Original Message-----
>From: ext Hisham Khartabil [mailto:hisham.khartabil@telio.no]=20
>Sent: 04 December, 2006 16:58
>To: Jeroen van Bemmel
>Cc: fluffy@cisco.com; Rohan Mahy; sip@ietf.org; Leis,Peter
>Subject: Re: [Sip] Re: issue with oubound and the used=20
>transport protocol
>
>I like this proposal, although I would not use MUST. Either of=20
>these is optional. Both is recommended.
>
>Hisham
>
>On Nov 17, 2006, at 6:20 PM, Jeroen van Bemmel wrote:
>
>> The outbound draft needs to say: "if a UA establishes a flow=20
>using an=20
>> unreliable transport, it MUST also establish one over a reliable=20
>> transport" (to support switch over). Moreover, I would=20
>suggest to add=20
>> "it is RECOMMENDED that the UA establishes a flow using a reliable=20
>> transport"
>>
>> Regards,
>> Jeroen
>>
>>
>
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use=20
>sip-implementors@cs.columbia.edu for questions on current sip=20
>Use sipping@ietf.org for new developments on the application of sip
>

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From sip-bounces@ietf.org Tue Dec 05 04:08:11 2006
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 10:07:32 +0100
To: <Markus.Isomaki@nokia.com>
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Because some service providers don't want to use TCP.

Hisham

On Dec 5, 2006, at 10:04 AM, <Markus.Isomaki@nokia.com> wrote:

> Hi,
>
> I don't see the point in this proposal. If the UA can establish a
> flow/connection for a reliable transport (TCP), why would it need the
> unreliable transport (UDP) flow in the first place? All traffic could 
> be
> sent over TCP anyway.
>
> Markus
>
>> -----Original Message-----
>> From: ext Hisham Khartabil [mailto:hisham.khartabil@telio.no]
>> Sent: 04 December, 2006 16:58
>> To: Jeroen van Bemmel
>> Cc: fluffy@cisco.com; Rohan Mahy; sip@ietf.org; Leis,Peter
>> Subject: Re: [Sip] Re: issue with oubound and the used
>> transport protocol
>>
>> I like this proposal, although I would not use MUST. Either of
>> these is optional. Both is recommended.
>>
>> Hisham
>>
>> On Nov 17, 2006, at 6:20 PM, Jeroen van Bemmel wrote:
>>
>>> The outbound draft needs to say: "if a UA establishes a flow
>> using an
>>> unreliable transport, it MUST also establish one over a reliable
>>> transport" (to support switch over). Moreover, I would
>> suggest to add
>>> "it is RECOMMENDED that the UA establishes a flow using a reliable
>>> transport"
>>>
>>> Regards,
>>> Jeroen
>>>
>>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
>


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From sip-bounces@ietf.org Tue Dec 05 04:14:38 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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Hi,

The proposal says that the UA MUST open a reliable transport connection
too. So, the service provider infra would need to support TCP
connections from all UAs using outbound. So, if the service provider or
the proxy vendor would be concerned about the number of TCP connections,
this would not help. But are you saying that the issue would not the
number of connections but the traffic over those connections? Can you
elaborate this?

Markus
=20

>-----Original Message-----
>From: ext Hisham Khartabil [mailto:hisham.khartabil@telio.no]=20
>Sent: 05 December, 2006 11:08
>To: Isomaki Markus (Nokia-SIR/Espoo)
>Cc: peter.leis@siemens.com; fluffy@cisco.com;=20
>rohan@ekabal.com; sip@ietf.org; jbemmel@zonnet.nl
>Subject: Re: [Sip] Re: issue with oubound and the used=20
>transport protocol
>
>Because some service providers don't want to use TCP.
>
>Hisham
>
>On Dec 5, 2006, at 10:04 AM, <Markus.Isomaki@nokia.com> wrote:
>
>> Hi,
>>
>> I don't see the point in this proposal. If the UA can establish a=20
>> flow/connection for a reliable transport (TCP), why would it=20
>need the=20
>> unreliable transport (UDP) flow in the first place? All=20
>traffic could=20
>> be sent over TCP anyway.
>>
>> Markus
>>
>>> -----Original Message-----
>>> From: ext Hisham Khartabil [mailto:hisham.khartabil@telio.no]
>>> Sent: 04 December, 2006 16:58
>>> To: Jeroen van Bemmel
>>> Cc: fluffy@cisco.com; Rohan Mahy; sip@ietf.org; Leis,Peter
>>> Subject: Re: [Sip] Re: issue with oubound and the used transport=20
>>> protocol
>>>
>>> I like this proposal, although I would not use MUST. Either=20
>of these=20
>>> is optional. Both is recommended.
>>>
>>> Hisham
>>>
>>> On Nov 17, 2006, at 6:20 PM, Jeroen van Bemmel wrote:
>>>
>>>> The outbound draft needs to say: "if a UA establishes a flow
>>> using an
>>>> unreliable transport, it MUST also establish one over a reliable=20
>>>> transport" (to support switch over). Moreover, I would
>>> suggest to add
>>>> "it is RECOMMENDED that the UA establishes a flow using a reliable=20
>>>> transport"
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>>
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol Use=20
>>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>>> sipping@ietf.org for new developments on the application of sip
>>>
>>
>
>

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Hisham Khartabil wrote:
> Because some service providers don't want to use TCP.

Spending IETF working group time and efforts on making standards that 
try to cope with people implementing only the bits and pieces they seem 
fit from earlier standards is both irresponsible and stupid.

I know you know that TCP+TLS is mandated for a proxy by RFC3261.

/Fredrik

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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 10:23:56 +0100
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On Dec 5, 2006, at 10:14 AM, <Markus.Isomaki@nokia.com> wrote:

> Hi,
>
> The proposal says that the UA MUST open a reliable transport connection
> too.

My proposal didn't say MUST.

Hisham

> So, the service provider infra would need to support TCP
> connections from all UAs using outbound. So, if the service provider or
> the proxy vendor would be concerned about the number of TCP 
> connections,
> this would not help. But are you saying that the issue would not the
> number of connections but the traffic over those connections? Can you
> elaborate this?
>
> Markus
>
>
>> -----Original Message-----
>> From: ext Hisham Khartabil [mailto:hisham.khartabil@telio.no]
>> Sent: 05 December, 2006 11:08
>> To: Isomaki Markus (Nokia-SIR/Espoo)
>> Cc: peter.leis@siemens.com; fluffy@cisco.com;
>> rohan@ekabal.com; sip@ietf.org; jbemmel@zonnet.nl
>> Subject: Re: [Sip] Re: issue with oubound and the used
>> transport protocol
>>
>> Because some service providers don't want to use TCP.
>>
>> Hisham
>>
>> On Dec 5, 2006, at 10:04 AM, <Markus.Isomaki@nokia.com> wrote:
>>
>>> Hi,
>>>
>>> I don't see the point in this proposal. If the UA can establish a
>>> flow/connection for a reliable transport (TCP), why would it
>> need the
>>> unreliable transport (UDP) flow in the first place? All
>> traffic could
>>> be sent over TCP anyway.
>>>
>>> Markus
>>>
>>>> -----Original Message-----
>>>> From: ext Hisham Khartabil [mailto:hisham.khartabil@telio.no]
>>>> Sent: 04 December, 2006 16:58
>>>> To: Jeroen van Bemmel
>>>> Cc: fluffy@cisco.com; Rohan Mahy; sip@ietf.org; Leis,Peter
>>>> Subject: Re: [Sip] Re: issue with oubound and the used transport
>>>> protocol
>>>>
>>>> I like this proposal, although I would not use MUST. Either
>> of these
>>>> is optional. Both is recommended.
>>>>
>>>> Hisham
>>>>
>>>> On Nov 17, 2006, at 6:20 PM, Jeroen van Bemmel wrote:
>>>>
>>>>> The outbound draft needs to say: "if a UA establishes a flow
>>>> using an
>>>>> unreliable transport, it MUST also establish one over a reliable
>>>>> transport" (to support switch over). Moreover, I would
>>>> suggest to add
>>>>> "it is RECOMMENDED that the UA establishes a flow using a reliable
>>>>> transport"
>>>>>
>>>>> Regards,
>>>>> Jeroen
>>>>>
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol Use
>>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>> sipping@ietf.org for new developments on the application of sip
>>>>
>>>
>>
>>
>


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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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On Dec 5, 2006, at 10:21 AM, Fredrik Thulin wrote:

> Hisham Khartabil wrote:
>> Because some service providers don't want to use TCP.
>
> Spending IETF working group time and efforts on making standards that 
> try to cope with people implementing only the bits and pieces they 
> seem fit from earlier standards is both irresponsible and stupid.

Spoken like a true academic. This has nothing to do with 
implementation, but has a lot to do with deployment. You can implement 
all the MUSTs you want, but you cannot mandate its use, TCP and TCP/TLS 
is one great example of that.

If the IETF does not want to solve real life deployment problems but 
instead wants to solve academic ones, then service providers will start 
deploying proprietary solutions to fit their immediate needs and 
vendors will implement products that no service provider cares for.

Thanks,
Hisham

>
> I know you know that TCP+TLS is mandated for a proxy by RFC3261.
>
> /Fredrik
>


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From sip-bounces@ietf.org Tue Dec 05 06:48:15 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
To: "Hisham Khartabil" <hisham.khartabil@telio.no>,
	"Fredrik Thulin" <ft@it.su.se>
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While I would not necessarily have phrased it as Fredrik did, he is
right.

If you don't want to reflect the previous decisions of the SIP WG that
already exist in published deliverables, then you should be seeking to
change those published deliverables.

For security, the way to do that is by discussing and including
appropriate text in draft-ietf-sip-sips-00. I don't believe I have seen
much Hisham sourced input.

If we don't do this we end up down a compatibility discussions rathole
from which the group will never emerge.

Regards

Keith

> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no]=20
> Sent: 05 December 2006 10:09
> To: Fredrik Thulin
> Cc: fluffy@cisco.com; rohan@ekabal.com; sip@ietf.org;=20
> peter.leis@siemens.com; Markus.Isomaki@nokia.com
> Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
>=20
> On Dec 5, 2006, at 10:21 AM, Fredrik Thulin wrote:
>=20
> > Hisham Khartabil wrote:
> >> Because some service providers don't want to use TCP.
> >
> > Spending IETF working group time and efforts on making=20
> standards that=20
> > try to cope with people implementing only the bits and pieces they=20
> > seem fit from earlier standards is both irresponsible and stupid.
>=20
> Spoken like a true academic. This has nothing to do with=20
> implementation, but has a lot to do with deployment. You can=20
> implement all the MUSTs you want, but you cannot mandate its=20
> use, TCP and TCP/TLS is one great example of that.
>=20
> If the IETF does not want to solve real life deployment=20
> problems but instead wants to solve academic ones, then=20
> service providers will start deploying proprietary solutions=20
> to fit their immediate needs and vendors will implement=20
> products that no service provider cares for.
>=20
> Thanks,
> Hisham
>=20
> >
> > I know you know that TCP+TLS is mandated for a proxy by RFC3261.
> >
> > /Fredrik
> >
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Hisham Khartabil wrote:
> 
> On Dec 5, 2006, at 10:21 AM, Fredrik Thulin wrote:
> 
>> Hisham Khartabil wrote:
>>> Because some service providers don't want to use TCP.
>>
>> Spending IETF working group time and efforts on making standards that 
>> try to cope with people implementing only the bits and pieces they 
>> seem fit from earlier standards is both irresponsible and stupid.
> 
> Spoken like a true academic. This has nothing to do with implementation, 
> but has a lot to do with deployment. You can implement all the MUSTs you 
> want, but you cannot mandate its use, TCP and TCP/TLS is one great 
> example of that.
> 
> If the IETF does not want to solve real life deployment problems but 
> instead wants to solve academic ones, then service providers will start 
> deploying proprietary solutions to fit their immediate needs and vendors 
> will implement products that no service provider cares for.

After sending I realized that my words were a bit harsh, for which I 
apologize. I guess you mistake the university domain name of mine with 
academic reasoning, but the truth is that I am _not_ interested in SIP 
in an academic way. I'm actually trying to deploy SIP on Stockholm 
university.

Even though I have implemented lots of things (including draft-outbound 
03) my primary concern is deployment, I can assure you. If there is 
anything I need for deployment it is simplicity. Outbound with TCP/TLS 
is a piece of cake - Outbound with UDP clearly is not. It is not TCP/TLS 
that is adding complexity, but UDP. I mentioned in another e-mail that 
my code to deframe SIP messages received over streams were less than 130 
lines of code. My STUN demuxing-responding code for UDP is about the 
double of that.

I'm assuming that you are a provider of SIP service to end users who 
might often be behind NAT. Can you please explain how you see it as the 
most deployable solution to base it all on UDP instead of offering 
TCP/TLS in your service and even advocating the use of UACs that support 
TCP/TLS?

/Fredrik


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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 13:14:36 +0100
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Fredrik Thulin" <ft@it.su.se>,
	"Hisham Khartabil" <hisham.khartabil@telio.no>
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Hi,

I still fail to see all the "complexity" that UDP adds.

Regards,

Christer=20

> -----Original Message-----
> From: Fredrik Thulin [mailto:ft@it.su.se]=20
> Sent: 5. joulukuuta 2006 14:08
> To: Hisham Khartabil
> Cc: fluffy@cisco.com; rohan@ekabal.com; sip@ietf.org;=20
> peter.leis@siemens.com; Markus.Isomaki@nokia.com
> Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
> Hisham Khartabil wrote:
> >=20
> > On Dec 5, 2006, at 10:21 AM, Fredrik Thulin wrote:
> >=20
> >> Hisham Khartabil wrote:
> >>> Because some service providers don't want to use TCP.
> >>
> >> Spending IETF working group time and efforts on making=20
> standards that=20
> >> try to cope with people implementing only the bits and pieces they=20
> >> seem fit from earlier standards is both irresponsible and stupid.
> >=20
> > Spoken like a true academic. This has nothing to do with=20
> > implementation, but has a lot to do with deployment. You=20
> can implement=20
> > all the MUSTs you want, but you cannot mandate its use, TCP and=20
> > TCP/TLS is one great example of that.
> >=20
> > If the IETF does not want to solve real life deployment=20
> problems but=20
> > instead wants to solve academic ones, then service providers will=20
> > start deploying proprietary solutions to fit their=20
> immediate needs and=20
> > vendors will implement products that no service provider cares for.
>=20
> After sending I realized that my words were a bit harsh, for=20
> which I apologize. I guess you mistake the university domain=20
> name of mine with academic reasoning, but the truth is that I=20
> am _not_ interested in SIP in an academic way. I'm actually=20
> trying to deploy SIP on Stockholm university.
>=20
> Even though I have implemented lots of things (including=20
> draft-outbound
> 03) my primary concern is deployment, I can assure you. If=20
> there is anything I need for deployment it is simplicity.=20
> Outbound with TCP/TLS is a piece of cake - Outbound with UDP=20
> clearly is not. It is not TCP/TLS that is adding complexity,=20
> but UDP. I mentioned in another e-mail that my code to=20
> deframe SIP messages received over streams were less than 130=20
> lines of code. My STUN demuxing-responding code for UDP is=20
> about the double of that.
>=20
> I'm assuming that you are a provider of SIP service to end=20
> users who might often be behind NAT. Can you please explain=20
> how you see it as the most deployable solution to base it all=20
> on UDP instead of offering TCP/TLS in your service and even=20
> advocating the use of UACs that support TCP/TLS?
>=20
> /Fredrik
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Christer Holmberg (JO/LMF) wrote:
> Hi,
> 
> I still fail to see all the "complexity" that UDP adds.

 From the top of my head,

   1) requiring STUN keepalives and de-muxing

   2) much more keepalives (every 29s instead of 120s)

   3) having to do resends yourself (not specific to Outbound)

   4) being insufficient in itself (having limitation of request size),
      therefor not adding any value when you _still_ have to do TCP/TLS
      as well as UDP

/Fredrik


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Dean Willis wrote:
> 
> Would a third party observing multiple uses of temp-gruu be able to 
> correlate them from this time-stamp and therefore make a reasonable 
> deduction that all the temp-gruus were bound to the same Contact and 
> therefore used by the same user?

The timestamp is only in the response to REGISTER. It isn't embedded in 
the gruu itself and so isn't visible to those receiving the gruu as a 
remote target or whatever.

(There is likely a timestamp *in* the gruu as well (along with the AOR 
and instance id), used to determine its validity. But when using that 
technique to construct temp gruus it will be necessary to encrypt all 
that info so that only the registrar/proxy can extract it.

	Paul

> Would we care?
> 
> -- 
> Dean
> 
> 
> On Dec 4, 2006, at 8:54 AM, Miguel Garcia wrote:
> 
>> I just checked the latest SIPit report, and it seems that only 5 
>> implementations supported reg event, as opposed to 29 that supported 
>> refer, 23 message-summary, or 14 presence.
>>
>> Well, this probably indicates that reg-event is not widely deployed to 
>> justify a solution based on it. Therefore, I don't mind Jonathan's 
>> solution #2
>>
>> /Miguel
>>
>> Jonathan Rosenberg wrote:
>>> The main question is really to decide on one of two mechanisms:
>>> 1. use reg-event if you are worried about use cases where this race 
>>> condition can occur (4-6 in Paul's original email)
>>> 2. add a parameter to Contact in the register response. Specifically, 
>>> I'd recommend a Contact header field parameter which indicates the 
>>> time at which the first contact with a given instance ID was 
>>> registered to that AOR. This value gets linked with each temp gruu, 
>>> and when the value changes, all of those temp-gruu get invalidated.
>>> Solution (2) is much simpler, but not needed if you think you'll 
>>> always have reg-event anyway. I don't think there is widespread 
>>> support for reg-event on endpoints yet. It is required by IMS.
>>> I am somewhat on the fence on this one. I do worry that reg-event is 
>>> genreally a very heavy solution, and may not see much usage. Option 2 
>>> is really trivially done, either in gruu or elsewhere. The main issue 
>>> is that it is another late change.
>>> -JOnathan R.
>>> Paul Kyzivat wrote:
>>>> Miguel,
>>>>
>>>> So to summarize, it seems you are of the belief that the cases where 
>>>> these measures may be inadequate are sufficiently obscure to ignore, 
>>>> and that new measures to cover the obscure cases are unnecessary. Is 
>>>> that right?
>>>>
>>>>     Paul
>>>>
>>>> Miguel Garcia wrote:
>>>>
>>>>> For issue 3, I would say that the solution is to avoid the 
>>>>> situation, e.g., by registering way in advance before the 
>>>>> registration expires. Doesn't RFC 3261 recommends to re-register 
>>>>> half time before the registration timer expires.
>>>>>
>>>>> Issues 4, 5, and 6 are solved by the reg event. So, shouldn't the 
>>>>> GRUU draft have a recommendation on the usage of the reg event to 
>>>>> solve them?
>>>>>
>>>>> /Miguel
>>>>>
>>>>> Paul Kyzivat wrote:
>>>>>
>>>>>> After some study, I have encountered a potential difficulty in 
>>>>>> determining when temp-gruus expire. After discussing this with 
>>>>>> Jonathan we decided to bring it to the list for discussion.
>>>>>>
>>>>>> As now defined, when you first register you may get a temp-gruu, 
>>>>>> which will remain valid as long as the registration does not 
>>>>>> expire or be removed. (Removal is a special case of expiration.) 
>>>>>> Each time the UA refreshes the registration it may get a new 
>>>>>> temp-gruu which also remains valid for the life of the 
>>>>>> registration. The UA can accumulate all these temp-gruus and use 
>>>>>> them as it sees fit until they expire.
>>>>>>
>>>>>> The problem is: how does the UA know when the registration has 
>>>>>> ended and the temp-gruus associated with it rendered invalid?
>>>>>>
>>>>>> This seems like a simple question, and in some cases it is simple, 
>>>>>> but not in all cases. The following are at least some of the ways 
>>>>>> that the registration can be ended, causing the invalidation of 
>>>>>> temp-gruus:
>>>>>> 1) the UA that originally registered may explicitly deregister
>>>>>> 2) the UA the originally registered may simply permit the
>>>>>>    registration to expire without attempting to refresh it
>>>>>> 3) the UA may attempt to refresh the registration, but be too slow,
>>>>>>    so that the reregistration arrives after the prior one expires
>>>>>> 4) some other UA may deregister all the registrations, including
>>>>>>    the one by this UA. (When the UA decides it should refresh the
>>>>>>    registration it will actually create a new one.)
>>>>>> 5) the registrar may remove the registration administratively
>>>>>> 6) the registrar may crash and restart, losing all its registrations
>>>>>>
>>>>>> In each of these cases the previously assigned temp-gruus become 
>>>>>> invalid. If the UA continues to use them, bad things will happen. 
>>>>>> This is true both for calls that are active when the 
>>>>>> deregistration occurs, and calls that are established later. What 
>>>>>> kinds of bad things happen?
>>>>>> - requests addressed to the gruu, both in-dialog and out-of-dialog
>>>>>>   will be undeliverable. (For instance a reINVITE or BYE in an
>>>>>>   existing call, or an out-of-dialog INVITE/Replaces for a transfer.)
>>>>>>
>>>>>> - request originated by the UA using the invalid gruu will fail
>>>>>>   immediately because the gruu is invalid.
>>>>>>
>>>>>> - if the UA subsequently reregisters, receives an incoming call, and
>>>>>>   uses one of the old temp-gruus as its contact, then the call will
>>>>>>   be established, but subsequent in- and out-of-dialog requests to
>>>>>>   the contact will fail.
>>>>>>
>>>>>> So clearly it is important for the UA to stop using temp-gruus 
>>>>>> that have become invalid. How can it know to do that in all of the 
>>>>>> above cases?
>>>>>>
>>>>>> 1) is straightforward. The UA knows it is deregistering, and should
>>>>>>    discard any cached temp-gruus. If it has a dialog with one of the
>>>>>>    temp-gruus as a contact, then it should do a target refresh and
>>>>>>    supply some working contact. (It really should do this *before*
>>>>>>    deregistering, and for a pub-gruu as well.)
>>>>>>
>>>>>> 2) the UA should take all the actions mentioned for (1) at or before
>>>>>>    the expiration time. This should be done a bit early to guard
>>>>>>    against clock skew problems.
>>>>>>
>>>>>> 3) in this case, it may be impossible for the UA to know whether
>>>>>>    the REGISTER arrived in time to be a reREGISTER or if it arrived
>>>>>>    late and was treated as a new REGISTER after the expiration of
>>>>>>    the old one. (The response to the REGISTER includes nothing that
>>>>>>    distinguishes the two cases.) About the best that can be done in
>>>>>>    this case is to check the time at which the response to the
>>>>>>    REGISTER is received. If it falls after the expected expiration
>>>>>>    time for the registration, act as if the old registration had
>>>>>>    expired.
>>>>>>
>>>>>> 4) This is a major problem. Of course, regardless of GRUU, the UA
>>>>>>    won't be able to receive incoming calls until it decides it is
>>>>>>    time to refresh the registration. In addition, until the
>>>>>>    registration reestablished, calls originated with one of its
>>>>>>    existing gruus will fail. Once the registration is reestablished,
>>>>>>    it still won't know its old temp-gruus are invalid. The only
>>>>>>    remedy for this is to subscribe to the "reg" event package,
>>>>>>    and use the resulting notifications to identify when it has
>>>>>>    been unregistered. If it does that, it can invalidate all its
>>>>>>    old temp-gruus, and then presumably reregister.
>>>>>>
>>>>>> 5) this is functionally equivalent to (4), except that perhaps
>>>>>>    attempts to reregister may fail.
>>>>>>
>>>>>> 6) this is also functionally equivalent to (4).
>>>>>>
>>>>>> The "reg" event does provide a fairly complete solution to this 
>>>>>> problem, for those that are willing and able to use it. It is not 
>>>>>> however a 100% reliable solution. Consider case (3), where the UA 
>>>>>> has a reg event subscription. All is well if the registration 
>>>>>> expires and a notification is sent indicating that, and then the 
>>>>>> reregistration occurs and a notification is sent about that. But 
>>>>>> if they happen very closely in time, it may be that only one 
>>>>>> notification is sent, reflecting the result of the reregistration. 
>>>>>> In that case the UA will not be aware that it had been 
>>>>>> unregistered for a period of time.
>>>>>>
>>>>>> A *complete* solution would, IMO, involve changing REGISTER so 
>>>>>> that it returns an indication of whether this was a new 
>>>>>> registration, or a refresh. This could take many forms. For 
>>>>>> instance the timestamp of the initial registration could always be 
>>>>>> returned with each contact, as well as the expiration time.
>>>>>>
>>>>>> The big question is whether this is a big enough issue to require 
>>>>>> a solution now, as part of GRUU, or if it can be ignored as 
>>>>>> insignificant or postponed to future work.
>>>>>>
>>>>>> I'd appreciate hearing what others think about this. At the moment 
>>>>>> I am leaning toward doing nothing, at least for now.
>>>>>>
>>>>>>     Thanks,
>>>>>>     Paul
>>>>>>
>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol
>>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>>>>
>>
>> --Miguel A. Garcia           tel:+358-50-4804586
>> sip:miguel.garcia@neonsite.net
>> Nokia Research Center      Helsinki, Finland
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>
> 

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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 16:33:23 +0200
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>If the IETF does not want to solve real life deployment=20
>problems but instead wants to solve academic ones, then=20
>service providers will start deploying proprietary solutions=20
>to fit their immediate needs and vendors will implement=20
>products that no service provider cares for.

Can you pinpont exactly the real life deployment problems you mention
with using TCP? Does it have something to do with server-side
scalability?

I personally believe going TCP-only would actually help deployment
rather than hinder it, but I'd be really interested to be convinced
otherwise.

Cheers,
Aki

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   From: Hisham Khartabil <hisham.khartabil@telio.no>

   Because some service providers don't want to use TCP.

Why?  I think adding a little more detail to this statement would
help.  Stated this plainly, my mind appends "... because they are
petulant".

But if service providers are avoiding TCP because there is some
genuine burden associated with using TCP, that must be expressible as
a requirement, one which may well have been overlooked.  So expressing
the requirement in as much detail as possible will help us progress.

Not knowing much about the subject, my guess would be that TCP has
overheads, in that one must keep significant state about each
connection and perhaps send keepalive packets periodically.  But
outbound seems to incorporate those already; I don't see how TCP could
be more burdensome than UDP-with-outbound.

Dale

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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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On Dec 5, 2006, at 9:28 AM, Dale.Worley@comcast.net wrote:

>    From: Hisham Khartabil <hisham.khartabil@telio.no>
>
>    Because some service providers don't want to use TCP.
>
> Why?  I think adding a little more detail to this statement would
> help.  Stated this plainly, my mind appends "... because they are
> petulant".
>
> But if service providers are avoiding TCP because there is some
> genuine burden associated with using TCP, that must be expressible as
> a requirement, one which may well have been overlooked.  So expressing
> the requirement in as much detail as possible will help us progress.
>
> Not knowing much about the subject, my guess would be that TCP has
> overheads, in that one must keep significant state about each
> connection and perhaps send keepalive packets periodically.  But
> outbound seems to incorporate those already; I don't see how TCP could
> be more burdensome than UDP-with-outbound.
>

The simplest answer, which we've discussed here previously, is that a  
naive implementation of a proxy using TCP is limited by the number of  
sockets provided by the OS, which is usually something less than the  
number of file handles and tends to be less than 64k. This caps the  
"fan out" of a proxy" at a number smaller than some operators would  
like. Of course, it's possible to write a more sophisticated proxy  
using select/poll that doesn't have this limitation, but only on OS  
that support it.

There's another answer occasionally used that is far less  
supportable: many people in the mobile community have historically  
believed that TCP is less suitable than UDP for mobile networks,  
because the reliability and timing properties of those networks are  
different. These networks also don't use NAT or "outbound" per se,  
and therefore generally do not do keepalives at the IP level. Given  
this, UDP implementations are more air-efficient than naive TCP  
implementations because of TCP keepalive and retransmission. Of  
course, one can make a slightly smarter TCP implementation that does  
very little TCP keepalive, so once again we've learned how to mostly  
overcome this objection.

One other thing that UDP makes easier is fault tolerance across a  
proxy farm with replicated state. Of course, we've also learned how  
to do this with TCP (and even with TLS) at reasonable levels of  
efficiency.

In other words, to the best of my knowledge, the only good reason to  
keep using UDP is because you have it deployed already and don't want  
to change, because change is expensive and disruptive. Of course, if  
one is going to phase in "outbound", one is going to have to make  
similar changes.

We should keep this in mind -- Change isn't mandatory. Neither is  
survival. But if you're unwilling to do the first, you're unlikely to  
achieve the second.

--
Dean

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From sip-bounces@ietf.org Tue Dec 05 11:32:48 2006
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Subject: Re: [Sip] Issue: Expiration of temp-gruu
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On Mon, 2006-12-04 at 17:15 -0500, Paul Kyzivat wrote:
> Scott,
> 
> Its certainly true that a gruu can always be tested to see if it works. 
> That would however raise the question of when one ought to do the test. 
> In normal cases you will know the status of your gruu by regular 
> methods, so you would have no reason to test it. So you would have to 
> test whenever you thought failure might have negative consequences. Some 
> of the times when failure might have bad consequences are: when 
> responding to an incoming INVITE, and any time during a dialog. I guess 
> you could send an OPTIONS and await the response before responding to an 
> INVITE, and it might even be masked by alerting, but it seems kind of 
> annoying to have to do that. But no matter how often you do it during a 
> dialog you still might miss the moment your gruu becomes invalid and the 
> other end of the dialog sends you a message. (Extremely unlikely, but 
> possible.)
> 
> I think the other existing solutions get "pretty close", and OPTIONS 
> doesn't significantly improve on that.

I don't like requiring the registrar to return a timestamp because it's
yet another piece of state that it has to keep for every registration.

I don't think it actually helps all that much - going back to your
original problem statements:

> 3) the UA may attempt to refresh the registration, but be too slow,
>     so that the reregistration arrives after the prior one expires
> 4) some other UA may deregister all the registrations, including
>     the one by this UA. (When the UA decides it should refresh the
>     registration it will actually create a new one.)
> 5) the registrar may remove the registration administratively
> 6) the registrar may crash and restart, losing all its registrations

Case 3 can be avoided by just not waiting too long (overlap by a few
minutes instead of a few seconds - if that's still problematic then you
probably have other problems that are worse anyway).

Cases 4,5, and 6 are essentially all the same from the point of view of
the UA; your registration state at the server is lost because of some
event you are not aware of.  This is true of any registration, and is
not specific to temp-gruu or public gruu; it can happen with a Contact
value you provide without gruu at all.  A gruu is slightly worse in that
loosing the state can affect in-dialog requests for established dialogs,
but I think that's just an unavoidable chance that you take when you
choose to use a gruu of either kind.  Put another way, if you use a
gruu, you accept the fact that you're relying on an additional service
for in-dialog connectivity - it's either worth the benefits or it isn't.




-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/


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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 08:30:12 -0800
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Keith,

I don't believe Hisham was objecting here to MUST implement.  I think 
Hisham is objecting to adding a MUST use statement to outbound.

thanks,
-rohan


On Dec 5, 2006, at 3:45, Drage, Keith ((Keith)) wrote:
> While I would not necessarily have phrased it as Fredrik did, he is
> right.
>
> If you don't want to reflect the previous decisions of the SIP WG that
> already exist in published deliverables, then you should be seeking to
> change those published deliverables.
>
> For security, the way to do that is by discussing and including
> appropriate text in draft-ietf-sip-sips-00. I don't believe I have seen
> much Hisham sourced input.
>
> If we don't do this we end up down a compatibility discussions rathole
> from which the group will never emerge.
>
> Regards
>
> Keith
>
>> -----Original Message-----
>> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no]
>> Sent: 05 December 2006 10:09
>> To: Fredrik Thulin
>> Cc: fluffy@cisco.com; rohan@ekabal.com; sip@ietf.org;
>> peter.leis@siemens.com; Markus.Isomaki@nokia.com
>> Subject: Re: [Sip] Re: issue with oubound and the used
>> transport protocol
>>
>>
>> On Dec 5, 2006, at 10:21 AM, Fredrik Thulin wrote:
>>
>>> Hisham Khartabil wrote:
>>>> Because some service providers don't want to use TCP.
>>>
>>> Spending IETF working group time and efforts on making
>> standards that
>>> try to cope with people implementing only the bits and pieces they
>>> seem fit from earlier standards is both irresponsible and stupid.
>>
>> Spoken like a true academic. This has nothing to do with
>> implementation, but has a lot to do with deployment. You can
>> implement all the MUSTs you want, but you cannot mandate its
>> use, TCP and TCP/TLS is one great example of that.
>>
>> If the IETF does not want to solve real life deployment
>> problems but instead wants to solve academic ones, then
>> service providers will start deploying proprietary solutions
>> to fit their immediate needs and vendors will implement
>> products that no service provider cares for.
>>
>> Thanks,
>> Hisham
>>
>>>
>>> I know you know that TCP+TLS is mandated for a proxy by RFC3261.
>>>
>>> /Fredrik
>>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>


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Subject: Re: [Sip] Issue: Expiration of temp-gruu
From: Scott Lawrence <slawrence@pingtel.com>
To: Paul Kyzivat <pkyzivat@cisco.com>
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On Mon, 2006-12-04 at 17:15 -0500, Paul Kyzivat wrote:
> Scott,
> 
> Its certainly true that a gruu can always be tested to see if it works. 
> That would however raise the question of when one ought to do the test. 
> In normal cases you will know the status of your gruu by regular 
> methods, so you would have no reason to test it. So you would have to 
> test whenever you thought failure might have negative consequences. Some 
> of the times when failure might have bad consequences are: when 
> responding to an incoming INVITE, and any time during a dialog. I guess 
> you could send an OPTIONS and await the response before responding to an 
> INVITE, and it might even be masked by alerting, but it seems kind of 
> annoying to have to do that. But no matter how often you do it during a 
> dialog you still might miss the moment your gruu becomes invalid and the 
> other end of the dialog sends you a message. (Extremely unlikely, but 
> possible.)
> 
> I think the other existing solutions get "pretty close", and OPTIONS 
> doesn't significantly improve on that.

I don't like requiring the registrar to return a timestamp because it's
yet another piece of state that it has to keep for every registration.

I don't think it actually helps all that much - going back to your
original problem statements:

> 3) the UA may attempt to refresh the registration, but be too slow,
>     so that the reregistration arrives after the prior one expires
> 4) some other UA may deregister all the registrations, including
>     the one by this UA. (When the UA decides it should refresh the
>     registration it will actually create a new one.)
> 5) the registrar may remove the registration administratively
> 6) the registrar may crash and restart, losing all its registrations

Case 3 can be avoided by just not waiting too long (overlap by a few
minutes instead of a few seconds - if that's still problematic then you
probably have other problems that are worse anyway).

Cases 4,5, and 6 are essentially all the same from the point of view of
the UA; your registration state at the server is lost because of some
event you are not aware of.  This is true of any registration, and is
not specific to temp-gruu or public gruu; it can happen with a Contact
value you provide without gruu at all.  A gruu is slightly worse in that
loosing the state can affect in-dialog requests for established dialogs,
but I think that's just an unavoidable chance that you take when you
choose to use a gruu of either kind.  Put another way, if you use a
gruu, you accept the fact that you're relying on an additional service
for in-dialog connectivity - it's either worth the benefits or it isn't.




-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/


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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 08:30:12 -0800
To: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
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Keith,

I don't believe Hisham was objecting here to MUST implement.  I think 
Hisham is objecting to adding a MUST use statement to outbound.

thanks,
-rohan


On Dec 5, 2006, at 3:45, Drage, Keith ((Keith)) wrote:
> While I would not necessarily have phrased it as Fredrik did, he is
> right.
>
> If you don't want to reflect the previous decisions of the SIP WG that
> already exist in published deliverables, then you should be seeking to
> change those published deliverables.
>
> For security, the way to do that is by discussing and including
> appropriate text in draft-ietf-sip-sips-00. I don't believe I have seen
> much Hisham sourced input.
>
> If we don't do this we end up down a compatibility discussions rathole
> from which the group will never emerge.
>
> Regards
>
> Keith
>
>> -----Original Message-----
>> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no]
>> Sent: 05 December 2006 10:09
>> To: Fredrik Thulin
>> Cc: fluffy@cisco.com; rohan@ekabal.com; sip@ietf.org;
>> peter.leis@siemens.com; Markus.Isomaki@nokia.com
>> Subject: Re: [Sip] Re: issue with oubound and the used
>> transport protocol
>>
>>
>> On Dec 5, 2006, at 10:21 AM, Fredrik Thulin wrote:
>>
>>> Hisham Khartabil wrote:
>>>> Because some service providers don't want to use TCP.
>>>
>>> Spending IETF working group time and efforts on making
>> standards that
>>> try to cope with people implementing only the bits and pieces they
>>> seem fit from earlier standards is both irresponsible and stupid.
>>
>> Spoken like a true academic. This has nothing to do with
>> implementation, but has a lot to do with deployment. You can
>> implement all the MUSTs you want, but you cannot mandate its
>> use, TCP and TCP/TLS is one great example of that.
>>
>> If the IETF does not want to solve real life deployment
>> problems but instead wants to solve academic ones, then
>> service providers will start deploying proprietary solutions
>> to fit their immediate needs and vendors will implement
>> products that no service provider cares for.
>>
>> Thanks,
>> Hisham
>>
>>>
>>> I know you know that TCP+TLS is mandated for a proxy by RFC3261.
>>>
>>> /Fredrik
>>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>


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Thread-Topic: watcher XML: why package parameter in watcher-list
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Hello,

A NOTIFY request containing watcher XML contains also an Event header
that indicates the event package this NOTIFY relates to. However,
RFC3858 instructs us to add a mandatory "package" parameter to each
watcher list of the watcher XML.=20

Why should there be a "package" parameter in every watcher-list? Why is
the event package of the Event header in the NOTIFY request
insufficient?=20

Thank you,

Tamar Barzuza.


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<p class=3DMsoNormal dir=3DLTR =
style=3D'text-align:left;direction:ltr;unicode-bidi:
embed'><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Hello,</span></font></p>

<p class=3DMsoNormal dir=3DLTR =
style=3D'text-align:left;direction:ltr;unicode-bidi:
embed'><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>A
NOTIFY request containing watcher XML contains also an Event header that
indicates the event package this NOTIFY relates to. However, RFC3858 =
instructs
us to add a mandatory &quot;package&quot; parameter to each watcher list =
of the
watcher XML. </span></font></p>

<p class=3DMsoNormal dir=3DLTR =
style=3D'text-align:left;direction:ltr;unicode-bidi:
embed'><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Why
should there be a &quot;package&quot; parameter in every watcher-list? =
Why is
the event package of the Event header in the NOTIFY request =
insufficient? </span></font></p>

<p class=3DMsoNormal dir=3DLTR =
style=3D'text-align:left;direction:ltr;unicode-bidi:
embed'><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Thank
you,</span></font></p>

<p class=3DMsoNormal dir=3DLTR =
style=3D'text-align:left;direction:ltr;unicode-bidi:
embed'><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial'>Tamar
Barzuza.</span></font></p>

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_______________________________________________
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--===============1234864021==--




From sip-bounces@ietf.org Tue Dec 05 14:25:38 2006
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On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:

>
>> If the IETF does not want to solve real life deployment
>> problems but instead wants to solve academic ones, then
>> service providers will start deploying proprietary solutions
>> to fit their immediate needs and vendors will implement
>> products that no service provider cares for.
>
> Can you pinpont exactly the real life deployment problems you mention
> with using TCP? Does it have something to do with server-side
> scalability?
>
> I personally believe going TCP-only would actually help deployment
> rather than hinder it, but I'd be really interested to be convinced
> otherwise.

I could ask you the converse. I'd really be interested to be convinced 
how TCP deployment improves scalability.

Hisham

>
> Cheers,
> Aki
>


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   From: Dean Willis <dean.willis@softarmor.com>

   In other words, to the best of my knowledge, the only good reason to  
   keep using UDP is because you have it deployed already and don't want  
   to change, because change is expensive and disruptive.

Which is actually a legitimate requirement, if the amount of
money/trouble is economically significant in an industry.

   Of course, if one is going to phase in "outbound", one is going to
   have to make similar changes.

It seems to me that that is the bottom line -- installing
UDP-with-outbound-keepalives is going to be nearly as much work as
installing TCP, so better to use TCP, which has better long-term
prospects.

I would, however, like to hear an analysis from someone who is
forseeing trouble from TCP.

Dale

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From sip-bounces@ietf.org Tue Dec 05 14:35:23 2006
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 20:34:31 +0100
To: Dale.Worley@comcast.net
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Dale,

Besides scalability issues, there are no technical issues. The issues 
are with having to throw all the infrastructure we currently have out 
of the window and start all over again. Having 110k+ customers live 
does not make is an easy and justifiable task.

Why do we prefer to run away from the problem and ignore it pretending 
it does not "really" exist instead of trying to fix it?

Hisham

On Dec 5, 2006, at 4:28 PM, Dale.Worley@comcast.net wrote:

>    From: Hisham Khartabil <hisham.khartabil@telio.no>
>
>    Because some service providers don't want to use TCP.
>
> Why?  I think adding a little more detail to this statement would
> help.  Stated this plainly, my mind appends "... because they are
> petulant".
>
> But if service providers are avoiding TCP because there is some
> genuine burden associated with using TCP, that must be expressible as
> a requirement, one which may well have been overlooked.  So expressing
> the requirement in as much detail as possible will help us progress.
>
> Not knowing much about the subject, my guess would be that TCP has
> overheads, in that one must keep significant state about each
> connection and perhaps send keepalive packets periodically.  But
> outbound seems to incorporate those already; I don't see how TCP could
> be more burdensome than UDP-with-outbound.
>
> Dale
>
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From sip-bounces@ietf.org Tue Dec 05 15:26:35 2006
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Hisham Khartabil wrote:
> 
> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
...
>> I personally believe going TCP-only would actually help deployment
>> rather than hinder it, but I'd be really interested to be convinced
>> otherwise.
> 
> I could ask you the converse. I'd really be interested to be convinced 
> how TCP deployment improves scalability.

To name two things where TCP would provide better scalability than UDP 
for a SIP service provider :

   1) Say that your services maximum capacity for non-INVITEs is 1000
      requests per second. If you get 2000 in a single second, you won't
      only have to deal with all those 2000 requests, but also with quite
      a few retransmissions sent after 500ms. With TCP, the wall isn't as
      dense.

   2) UDP sockets have a shared receive buffer, most commonly around 64k
      I believe (even though the maximum seems to be 256k on FreeBSD at
      least). With SIP messages of perhaps 700 bytes on average, you
      have to be able to move messages out of that buffer pretty fast to
      keep it from filling up. A full buffer means dropped packets. I've
      also heard that on for example Linux, every packet actually
      occupies an even 2k (or was it 4k) in the buffer, so 64k buffer is
      actually at most 32 packets at a time.

And then there are of course other aspects than scalability where UDP is 
an inferior solution, most notably security related things like for 
example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).

/Fredrik


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Scott,

To paraphrase, it seems that you think we have enough mechanism now, and 
don't need any more to deal with the issues raised in this thread. Is 
that right?

	Paul

Scott Lawrence wrote:
> On Mon, 2006-12-04 at 17:15 -0500, Paul Kyzivat wrote:
>> Scott,
>>
>> Its certainly true that a gruu can always be tested to see if it works. 
>> That would however raise the question of when one ought to do the test. 
>> In normal cases you will know the status of your gruu by regular 
>> methods, so you would have no reason to test it. So you would have to 
>> test whenever you thought failure might have negative consequences. Some 
>> of the times when failure might have bad consequences are: when 
>> responding to an incoming INVITE, and any time during a dialog. I guess 
>> you could send an OPTIONS and await the response before responding to an 
>> INVITE, and it might even be masked by alerting, but it seems kind of 
>> annoying to have to do that. But no matter how often you do it during a 
>> dialog you still might miss the moment your gruu becomes invalid and the 
>> other end of the dialog sends you a message. (Extremely unlikely, but 
>> possible.)
>>
>> I think the other existing solutions get "pretty close", and OPTIONS 
>> doesn't significantly improve on that.
> 
> I don't like requiring the registrar to return a timestamp because it's
> yet another piece of state that it has to keep for every registration.
> 
> I don't think it actually helps all that much - going back to your
> original problem statements:
> 
>> 3) the UA may attempt to refresh the registration, but be too slow,
>>     so that the reregistration arrives after the prior one expires
>> 4) some other UA may deregister all the registrations, including
>>     the one by this UA. (When the UA decides it should refresh the
>>     registration it will actually create a new one.)
>> 5) the registrar may remove the registration administratively
>> 6) the registrar may crash and restart, losing all its registrations
> 
> Case 3 can be avoided by just not waiting too long (overlap by a few
> minutes instead of a few seconds - if that's still problematic then you
> probably have other problems that are worse anyway).
> 
> Cases 4,5, and 6 are essentially all the same from the point of view of
> the UA; your registration state at the server is lost because of some
> event you are not aware of.  This is true of any registration, and is
> not specific to temp-gruu or public gruu; it can happen with a Contact
> value you provide without gruu at all.  A gruu is slightly worse in that
> loosing the state can affect in-dialog requests for established dialogs,
> but I think that's just an unavoidable chance that you take when you
> choose to use a gruu of either kind.  Put another way, if you use a
> gruu, you accept the fact that you're relying on an additional service
> for in-dialog connectivity - it's either worth the benefits or it isn't.
> 
> 
> 
> 

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From sip-bounces@ietf.org Tue Dec 05 16:20:36 2006
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On Tue, 2006-12-05 at 16:07 -0500, Paul Kyzivat wrote:
> Scott,
> 
> To paraphrase, it seems that you think we have enough mechanism now, and 
> don't need any more to deal with the issues raised in this thread. Is 
> that right?

Yes, that's correct.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/


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Let's go back to the original topic - what should the draft say about the 
transport protocol in relationship with outbound

The major use case for outbound is UAs behind NAT / firewalls. Many of such 
devices have known issues with UDP fragmentation, so for many environments 
TCP will need to be used for messages larger than the path MTU (often ~1460 
bytes in ethernet environments, RFC3261 puts the limit at 1300).

If you start from the requirement "MUST only use UDP", then there is little 
room for constructive discussions. For UDP-only to work you will need to 
ensure that messages either stay below path MTUs, or disallow NAT. This 
implies significant limitations for your deployment environment. In any 
case, this is not a setup that outbound should focus on.

In the general case, there are environments that have to deal with NAT and 
messages larger than path MTU. This is what outbound was written for. This 
also implies a UA that supports TCP. Given that the UA supports TCP and the 
fact that presence of devices like stateful firewalls cannot reliably be 
detected, the UA SHOULD use TCP (or another connection oriented transport). 
If in addition the UA supports TLS, it SHOULD use TLS.

I think this is sufficient

Regards,
Jeroen

Fredrik Thulin wrote:
> Hisham Khartabil wrote:
>>
>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
> ...
>>> I personally believe going TCP-only would actually help deployment
>>> rather than hinder it, but I'd be really interested to be convinced
>>> otherwise.
>>
>> I could ask you the converse. I'd really be interested to be
>> convinced how TCP deployment improves scalability.
>
> To name two things where TCP would provide better scalability than UDP
> for a SIP service provider :
>
>   1) Say that your services maximum capacity for non-INVITEs is 1000
>      requests per second. If you get 2000 in a single second, you
>      won't only have to deal with all those 2000 requests, but also
>      with quite a few retransmissions sent after 500ms. With TCP, the
>      wall isn't as dense.
>
>   2) UDP sockets have a shared receive buffer, most commonly around
>      64k I believe (even though the maximum seems to be 256k on
>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>      average, you have to be able to move messages out of that buffer
>      pretty fast to keep it from filling up. A full buffer means
>      dropped packets. I've also heard that on for example Linux,
>      every packet actually occupies an even 2k (or was it 4k) in the
>      buffer, so 64k buffer is actually at most 32 packets at a time.
>
> And then there are of course other aspects than scalability where UDP
> is an inferior solution, most notably security related things like for
> example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
>
> /Fredrik
>
>
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From sip-bounces@ietf.org Tue Dec 05 16:50:34 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 23:50:00 +0200
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Hi,

On 05 Dec 2006, Dean Willis wrote:=20
>The simplest answer, which we've discussed here previously, is=20
>that a naive implementation of a proxy using TCP is limited by=20
>the number of sockets provided by the OS, which is usually=20
>something less than the number of file handles and tends to be=20
>less than 64k. This caps the "fan out" of a proxy" at a number=20
>smaller than some operators would like. Of course, it's=20

but it's also good to notice that many/most modern operating systems
do not have these limitations. For instance, on most systems the=20
64k limit you mention, only applies to outbound connections (due to the
available port range), and does not limit for instance how many incoming
TCP=20
flows a SIP proxy can handle. I believe the unofficial record for a
single=20
x86 box is still the 1.6M TCP connections (on FreeBSD, from 2003) [1],
but=20
this of course is not the case in general. But tens of thousands of TCP=20
connections is surely not a problem on any modern system.

And of course, it's true a naive implementation will have problems, but=20
I can assure the limit on TCP connections is not the only place where
a naive implementation will hit the limits. ;)

There are numerous interesting papers and resources on the topic [2-4]
(and
note that most of these are quite old papers), and many indicate=20
that TCP scalability is not _that_ big of an issue. Especially, in
the case of SIP, if we consider that UDP flows will require more
frequent=20
keepalives, it is not obvious that a system using persistent TCP flows=20
from the clients, will scale worse than a similar system utilizing UDP.

[1] http://kerneltrap.org/node/277 (search for "1.6 Million")
[2] http://www.kegel.com/c10k.html
[3]
http://bcr2.uwaterloo.ca/~brecht/papers/getpaper.php?file=3Dols-2004.pdf
[4] http://haproxy.1wt.eu/

>possible to write a more sophisticated proxy using select/poll=20
>that doesn't have this limitation, but only on OS that support it.

Actually you shouldn't be using select/poll if you want to=20
write a modern proxy -- see [2,3] above. ;)=20

>One other thing that UDP makes easier is fault tolerance=20
>across a proxy farm with replicated state. Of course, we've=20
>also learned how to do this with TCP (and even with TLS) at=20
>reasonable levels of efficiency.

Yes, agreed. This is a real concern, but also a problem that
has been solved in many existing TCP/TLS based services.

>In other words, to the best of my knowledge, the only good=20
>reason to keep using UDP is because you have it deployed=20
>already and don't want to change, because change is expensive=20
>and disruptive. Of course, if one is going to phase in=20

Yes, agreed.=20

But it's also good to note that a big reason why UDP is so widely
used as the default transport, is the not-so-great state of standards
support=20
for using TCP/TLS in SIP clients (how TCP/TLS flows are managed/reused
in clients,
how server elements can reuse these flows -> leading to need=20
for ugly, but currently necessary, real-world hacks as described=20
for instance in [5]). The sip-outbound draft may be useful for UDP, but=20
it is absolutely critical for getting TCP/TLS more widely used.=20

[5] sect 3.4.1 of
    http://tools.ietf.org/wg/sipping/draft-ietf-sipping-sbc-funcs/

--=20
first.surname@nokia.com (Kai Vehmanen)
Networking Technologies Laboratory, Nokia Research Center

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From sip-bounces@ietf.org Tue Dec 05 17:44:56 2006
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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 23:43:52 +0100
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Not to mention the differences in transaction timers. For UDP transaction 
state (in particular non-INVITE server transactions) is retained much longer 
than for TCP. For the proxy-2-proxy case (i.e. small fan-in, few rx 
buffers ) TCP is often a better choice than UDP, especially when memory is 
your bottleneck.

Drawbacks for TCP are message boundary detection overhead and some byte 
copying overhead, which add some delay and some CPU load (though probably 
insignificant). Implementation complexity is slightly higher for TCP too.

Though interesting, IMHO all of this is not so relevant for the outbound 
discussion. Proxies must already implement TCP, the UA is better of using 
TCP if it can.

Regards,
Jeroen

Kai.Vehmanen@nokia.com wrote:
> Hi,
>
> On 05 Dec 2006, Dean Willis wrote:
>> The simplest answer, which we've discussed here previously, is
>> that a naive implementation of a proxy using TCP is limited by
>> the number of sockets provided by the OS, which is usually
>> something less than the number of file handles and tends to be
>> less than 64k. This caps the "fan out" of a proxy" at a number
>> smaller than some operators would like. Of course, it's
>
> but it's also good to notice that many/most modern operating systems
> do not have these limitations. For instance, on most systems the
> 64k limit you mention, only applies to outbound connections (due to
> the available port range), and does not limit for instance how many
> incoming TCP
> flows a SIP proxy can handle. I believe the unofficial record for a
> single
> x86 box is still the 1.6M TCP connections (on FreeBSD, from 2003) [1],
> but
> this of course is not the case in general. But tens of thousands of
> TCP connections is surely not a problem on any modern system.
>
> And of course, it's true a naive implementation will have problems,
> but I can assure the limit on TCP connections is not the only place
> where
> a naive implementation will hit the limits. ;)
>
> There are numerous interesting papers and resources on the topic [2-4]
> (and
> note that most of these are quite old papers), and many indicate
> that TCP scalability is not _that_ big of an issue. Especially, in
> the case of SIP, if we consider that UDP flows will require more
> frequent
> keepalives, it is not obvious that a system using persistent TCP flows
> from the clients, will scale worse than a similar system utilizing
> UDP.
>
> [1] http://kerneltrap.org/node/277 (search for "1.6 Million")
> [2] http://www.kegel.com/c10k.html
> [3]
> http://bcr2.uwaterloo.ca/~brecht/papers/getpaper.php?file=ols-2004.pdf
> [4] http://haproxy.1wt.eu/
>
>> possible to write a more sophisticated proxy using select/poll
>> that doesn't have this limitation, but only on OS that support it.
>
> Actually you shouldn't be using select/poll if you want to
> write a modern proxy -- see [2,3] above. ;)
>
>> One other thing that UDP makes easier is fault tolerance
>> across a proxy farm with replicated state. Of course, we've
>> also learned how to do this with TCP (and even with TLS) at
>> reasonable levels of efficiency.
>
> Yes, agreed. This is a real concern, but also a problem that
> has been solved in many existing TCP/TLS based services.
>
>> In other words, to the best of my knowledge, the only good
>> reason to keep using UDP is because you have it deployed
>> already and don't want to change, because change is expensive
>> and disruptive. Of course, if one is going to phase in
>
> Yes, agreed.
>
> But it's also good to note that a big reason why UDP is so widely
> used as the default transport, is the not-so-great state of standards
> support
> for using TCP/TLS in SIP clients (how TCP/TLS flows are managed/reused
> in clients,
> how server elements can reuse these flows -> leading to need
> for ugly, but currently necessary, real-world hacks as described
> for instance in [5]). The sip-outbound draft may be useful for UDP,
> but it is absolutely critical for getting TCP/TLS more widely used.
>
> [5] sect 3.4.1 of
>     http://tools.ietf.org/wg/sipping/draft-ietf-sipping-sbc-funcs/ 


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From sip-bounces@ietf.org Tue Dec 05 18:11:13 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 17:09:43 -0600
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Thanks for bringning back the original topic.

It seems to me that we do not have any evidence that when Outbound
keep-alive is
used because of the presence of a NAT, that there are any advantage to
using
UDP instead of TCP. It seems that all the perceived advantages of UDP
versus TCP
vanish in the presence of a NAT (in fact some have argued that TCP will
actually
scale better). Worst, all the evidence is that UDP will have=20
problems with fragmentation and that it will therefore not be suitable.

Futhermore, if the most important use case for outbound keep-alive is
TCP=20
and not UDP, "optimizing" the solution for UDP is not a good idea. I'd
rather
we have a solution optimized for TCP (and TLS). If we have a solution
for UDP or
not is of little interest to me: I just don't think it will be very
popular and it
will be just one more useless feature to implement.

Nit-pick: the other major use case for outbound is TLS support, but that
does
not require the keepalive. So what you really meant in the second
paragraph is
"The major use case for outbound keep-alive is ...".

> -----Original Message-----
> From: Jeroen van Bemmel [mailto:jbemmel@zonnet.nl]=20
> Sent: Tuesday, December 05, 2006 13:44
> To: Fredrik Thulin; Hisham Khartabil
> Cc: fluffy@cisco.com; rohan@ekabal.com; sip@ietf.org;=20
> Markus.Isomaki@nokia.com; peter.leis@siemens.com; aki.niemi@nokia.com
> Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
> Let's go back to the original topic - what should the draft=20
> say about the transport protocol in relationship with outbound
>=20
> The major use case for outbound is UAs behind NAT /=20
> firewalls. Many of such devices have known issues with UDP=20
> fragmentation, so for many environments TCP will need to be=20
> used for messages larger than the path MTU (often ~1460 bytes=20
> in ethernet environments, RFC3261 puts the limit at 1300).
>=20
> If you start from the requirement "MUST only use UDP", then=20
> there is little room for constructive discussions. For=20
> UDP-only to work you will need to ensure that messages either=20
> stay below path MTUs, or disallow NAT. This implies=20
> significant limitations for your deployment environment. In=20
> any case, this is not a setup that outbound should focus on.
>=20
> In the general case, there are environments that have to deal=20
> with NAT and messages larger than path MTU. This is what=20
> outbound was written for. This also implies a UA that=20
> supports TCP. Given that the UA supports TCP and the fact=20
> that presence of devices like stateful firewalls cannot=20
> reliably be detected, the UA SHOULD use TCP (or another=20
> connection oriented transport).=20
> If in addition the UA supports TLS, it SHOULD use TLS.
>=20
> I think this is sufficient
>=20
> Regards,
> Jeroen
>=20
> Fredrik Thulin wrote:
> > Hisham Khartabil wrote:
> >>
> >> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
> > ...
> >>> I personally believe going TCP-only would actually help=20
> deployment=20
> >>> rather than hinder it, but I'd be really interested to be=20
> convinced=20
> >>> otherwise.
> >>
> >> I could ask you the converse. I'd really be interested to be=20
> >> convinced how TCP deployment improves scalability.
> >
> > To name two things where TCP would provide better=20
> scalability than UDP=20
> > for a SIP service provider :
> >
> >   1) Say that your services maximum capacity for non-INVITEs is 1000
> >      requests per second. If you get 2000 in a single second, you
> >      won't only have to deal with all those 2000 requests, but also
> >      with quite a few retransmissions sent after 500ms.=20
> With TCP, the
> >      wall isn't as dense.
> >
> >   2) UDP sockets have a shared receive buffer, most commonly around
> >      64k I believe (even though the maximum seems to be 256k on
> >      FreeBSD at least). With SIP messages of perhaps 700 bytes on
> >      average, you have to be able to move messages out of=20
> that buffer
> >      pretty fast to keep it from filling up. A full buffer means
> >      dropped packets. I've also heard that on for example Linux,
> >      every packet actually occupies an even 2k (or was it 4k) in the
> >      buffer, so 64k buffer is actually at most 32 packets at a time.
> >
> > And then there are of course other aspects than scalability=20
> where UDP=20
> > is an inferior solution, most notably security related=20
> things like for=20
> > example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
> >
> > /Fredrik
> >
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
>=20
>=20
> _______________________________________________
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>=20

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From sip-bounces@ietf.org Tue Dec 05 19:39:32 2006
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I just submitted a new version of this I-D.  Until it appears in the
archives, a copy can be found here:

http://marc.petit-huguenin.org/draft-petithuguenin-sip-fragmentation-responses-01.txt

The new version was rewritten to be more readable and an example was
added.  This version also solves a problem when multiple hops would
trigger response fragmentation.

I am interested in any comments or questions on this I-D, and especially
to hear if people are interested in this work.

Thanks.

-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Tue, 5 Dec 2006 21:48:51 -0600
To: Hisham Khartabil <hisham.khartabil@telio.no>
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On Dec 5, 2006, at 1:22 PM, Hisham Khartabil wrote:

>
> I could ask you the converse. I'd really be interested to be  
> convinced how TCP deployment improves scalability.
>

That's easy. Imagine that the Internet is a set of sewer pipes, down  
which someone periodically flushes whole cantaloupe. The cantaloupe  
have the side effect of blocking up other drains, causing "back up"  
problems in other people's plumbing fictures.

UDP fragmentation of large responses (or large messages, if one is so  
tacky as to send them that way) will cause bursts of non-flow- 
controlled traffic to be inserted into the network. This clogs up  
buffers, reduces efficient network flow, and generally makes SIP a  
bad network neighbor -- much like the moron upstairs who flushes the  
cantaloupe.

TCP reduces the congestion impact of large messages substantially.  
The net effect is like running the whole cantaloupe through a  
blender, flushing it a cup at a time, and not flushing the next cup  
until you've asked everyone in the building to make sure their  
plumbing is working.

Sure, we could re-invent all of the TCP flow-control using UDP, but  
why bother? We'd eventually just have TCP again.

Factor in the detail that we're well on the way towards having the  
average message large enough to throw at least one fragment (and  
possibly 3 or 4, especially on smaller MTU networks), and I think TCP- 
only SIP starts to become very attractive.


--
Dean 

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From sip-bounces@ietf.org Wed Dec 06 07:58:37 2006
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Thread-Topic: Issue related to sigcomp
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Hello,

I have encountered one unclarity in usage of SigComp as specified by RFC =
3486. In section "4. Sending a Request to a Server" there is said that =
if client wants to receive subsequent requests with compression then =
"this client SHOULD add the parameter comp=3Dsigcomp to the URI in the =
Contact header field". Later in the specification there are examples =
that are in conflict with this statement.

(1) INVITE UAS
    Via: UAC;comp=3Dsigcomp
    Route: P1;comp=3Dsigcomp
    Contact: UAC;comp=3Dsigcomp

The contact header adds the comp parameter, but not as the URI parameter =
but as the header parameter.

I have found another example in 3GPP TS 24.141 v7.2.0 where on page 24 =
there is the Contact header in the form

Contact: <sip:[5555::aaa:bbb:ccc:ddd]:1357;comp=3Dsigcomp>

This header has no header parameter and its value is sip uri with =
parameter. Later in that document, there is SIP NOTIFY message where the =
SIP URI from the Contact header is used in Request-URI in exactly the =
same format.

I have got few questions:
1. Where shall be the comp parameter used - as uri parameter or as =
header parameter?
2. If it is uri parameter, are the examples in RFC 3486 invalid?
3. If it is header parameter, then if client issues comp parameter as =
uri parameter does it mean that it is willing to receive compressed =
content?

br, Martin Hynar

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From sip-bounces@ietf.org Wed Dec 06 08:27:23 2006
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Subject: Re: [Sip] Issue related to sigcomp
Date: Wed, 6 Dec 2006 14:26:31 +0100
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Martin,

The text above the example in RFC3486 explicitly says "We have not used a 
correct format for these header fields."

comp=sigcomp is a URI parameter, not a header parameter

Regards,
Jeroen

Martin.Hynar@tietoenator.com wrote:
> Hello,
>
> I have encountered one unclarity in usage of SigComp as specified by
> RFC 3486. In section "4. Sending a Request to a Server" there is said
> that if client wants to receive subsequent requests with compression
> then "this client SHOULD add the parameter comp=sigcomp to the URI in
> the Contact header field". Later in the specification there are
> examples that are in conflict with this statement.
>
> (1) INVITE UAS
>     Via: UAC;comp=sigcomp
>     Route: P1;comp=sigcomp
>     Contact: UAC;comp=sigcomp
>
> The contact header adds the comp parameter, but not as the URI
> parameter but as the header parameter.
>
> I have found another example in 3GPP TS 24.141 v7.2.0 where on page
> 24 there is the Contact header in the form
>
> Contact: <sip:[5555::aaa:bbb:ccc:ddd]:1357;comp=sigcomp>
>
> This header has no header parameter and its value is sip uri with
> parameter. Later in that document, there is SIP NOTIFY message where
> the SIP URI from the Contact header is used in Request-URI in exactly
> the same format.
>
> I have got few questions:
> 1. Where shall be the comp parameter used - as uri parameter or as
> header parameter?
> 2. If it is uri parameter, are the examples in RFC 3486 invalid?
> 3. If it is header parameter, then if client issues comp parameter as
> uri parameter does it mean that it is willing to receive compressed
> content?
>
> br, Martin Hynar
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


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From: Sean Olson <seanol@exchange.microsoft.com>
To: Fredrik Thulin <ft@it.su.se>, Hisham Khartabil <hisham.khartabil@telio.no>
Date: Tue, 5 Dec 2006 12:31:45 -0800
Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Thread-Topic: [Sip] Re: issue with oubound and the used transport protocol
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My two cents:  it's all about re-transmissions and security.  Would be curi=
ous to hear deployment experience around DTLS and whether or not that has a=
ctually helped the security of UDP deployments.

-----Original Message-----
From: Fredrik Thulin [mailto:ft@it.su.se]
Sent: Tuesday, December 05, 2006 12:24 PM
To: Hisham Khartabil
Cc: fluffy@cisco.com; rohan@ekabal.com; sip@ietf.org; Markus.Isomaki@nokia.=
com; peter.leis@siemens.com; aki.niemi@nokia.com
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol

Hisham Khartabil wrote:
>
> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
...
>> I personally believe going TCP-only would actually help deployment
>> rather than hinder it, but I'd be really interested to be convinced
>> otherwise.
>
> I could ask you the converse. I'd really be interested to be convinced
> how TCP deployment improves scalability.

To name two things where TCP would provide better scalability than UDP
for a SIP service provider :

   1) Say that your services maximum capacity for non-INVITEs is 1000
      requests per second. If you get 2000 in a single second, you won't
      only have to deal with all those 2000 requests, but also with quite
      a few retransmissions sent after 500ms. With TCP, the wall isn't as
      dense.

   2) UDP sockets have a shared receive buffer, most commonly around 64k
      I believe (even though the maximum seems to be 256k on FreeBSD at
      least). With SIP messages of perhaps 700 bytes on average, you
      have to be able to move messages out of that buffer pretty fast to
      keep it from filling up. A full buffer means dropped packets. I've
      also heard that on for example Linux, every packet actually
      occupies an even 2k (or was it 4k) in the buffer, so 64k buffer is
      actually at most 32 packets at a time.

And then there are of course other aspects than scalability where UDP is
an inferior solution, most notably security related things like for
example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).

/Fredrik


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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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   From: Hisham Khartabil <hisham.khartabil@telio.no>

   Besides scalability issues, there are no technical issues.

Naively, the scaling of UDP-with-outbound doesn't seem to be any
better than TCP-with-outbound.  Can you elucidate the scalability
issues that are causing you problems?

   The issues are with having to throw all the infrastructure we
   currently have out of the window and start all over again. Having
   110k+ customers live does not make is an easy and justifiable task.

Naively, to implement outbound will require changing the software in
the proxies and UAs, regardless of whether it uses UDP or TCP.  I
suppose this can be called "throwing infrastructure out of the
window", but it seems to be a routine matter.  Can you elucidate why
UDP-with-outbound reduces this burden compared to implementing
TCP-with-outbound?

   Why do we prefer to run away from the problem and ignore it
   pretending it does not "really" exist instead of trying to fix it?

Can you elucidate what "the problem" is that you are referring to?

Dale

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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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> Sure, we could re-invent all of the TCP flow-control using UDP, but why 
> bother? We'd eventually just have TCP again.

More likely, SCTP.

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From sip-bounces@ietf.org Wed Dec 06 19:25:57 2006
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Subject: [Sip] I-D: Preventing Fragmentation for Client Initiated
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I followed a little bit the discussion about the usage of UDP and TCP
with Outbound, and my opinion is that both UDP and TCP need to be supported.

Obviously UDP alone cannot be used with Outbound because if there is a
potential fragmentation in a request sent to the UA or in a response
sent from the UA, there is no way to switch to a congestion controlled
transport.

So we (Packet8) did some tests by comparing UDP and TCP, and we found
that, even by using advanced coding techniques, UDP uses less resources
than TCP.

So we tried a combination of temporary TCP and UDP.  The idea is that
TCP is really needed only for few types of transactions on a real life
endpoint.  One of this transactions is an INVITE, because the size added
by the SDP, especially when carrying ICE attributes and audio+video+text
sessions, nearly always results in a packet size that is over the MTU.
Another case is the full state NOTIFY for the reg-event that is to be
received after sending a SUBSCRIBE (partial state NOTIFY are OK).
REGISTER, even fully loaded with extensions are generally OK.

We retried our tests and found that using temporary TCP for the INVITE
and full state NOTIFY transaction with UDP for all the other
transactions was using more ressources than UDP alone, but still less
than using only TCP.

So I wrote this I-D that describe a solution to implement this mixed
UDP/TCP as an extension to Outbound.  This is really a first draft, as
there is a lot missing (e.g. how an edge proxy knows that the UA
supports this, etc...), but I wanted to know if there is interest in
working on this direction.

Until the draft appears in the archives, a copy can be found here:

http://marc.petit-huguenin.org/draft-petithuguenin-sip-outbound-fragmentation-00.txt

Questions and comments are welcome.

Thanks.

-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Wed, 6 Dec 2006 16:33:15 -0800
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I find this whole conversation way off base. If people would like to  
deprecate UDP, go work on a document to explain that and feel free to  
bring it to the WG. In the mean time, SIP is specified over both UDP  
and TCP. Most deployments only use UDP, some only use TCP, and I  
suspect there are some that do both but I am not aware of them.  
Outbound as it is seems very consistent with 3261 in that it allows a  
system to use TCP or UDP. Keeping this in mind few comments inline ....

On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:

> Let's go back to the original topic - what should the draft say  
> about the transport protocol in relationship with outbound
>
> The major use case for outbound is UAs behind NAT / firewalls. Many  
> of such devices have known issues with UDP fragmentation, so for  
> many environments TCP will need to be used for messages larger than  
> the path MTU (often ~1460 bytes in ethernet environments, RFC3261  
> puts the limit at 1300).

These known issues with fragmentation might be less common than you  
think however, outbound does not have anything to do with MTU issues.  
It does not try and solve them and it does not make them worse.

>
> If you start from the requirement "MUST only use UDP", then there  
> is little room for constructive discussions. For UDP-only to work  
> you will need to ensure that messages either stay below path MTUs,  
> or disallow NAT. This implies significant limitations for your  
> deployment environment. In any case, this is not a setup that  
> outbound should focus on.

MTU is a problem regardless of NATs or Firewalls. Fragmented packets  
do manager to traverse most NATs and Firewalls.

>
> In the general case, there are environments that have to deal with  
> NAT and messages larger than path MTU. This is what outbound was  
> written for. This also implies a UA that supports TCP. Given that  
> the UA supports TCP and the fact that presence of devices like  
> stateful firewalls cannot reliably be detected, the UA SHOULD use  
> TCP (or another connection oriented transport). If in addition the  
> UA supports TLS, it SHOULD use TLS.

Again, this topic is not really about outbound. If you think a 3261  
UA that supports both UDP and TCP SHOULD use TCP, that is a fine  
argument to go make in the discussion about use of TCP and UDP but it  
seems to be an independent choice from anything to do with outbound.


Cullen <with my individual hat on>

>
> I think this is sufficient
>
> Regards,
> Jeroen
>
> Fredrik Thulin wrote:
>> Hisham Khartabil wrote:
>>>
>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>> ...
>>>> I personally believe going TCP-only would actually help deployment
>>>> rather than hinder it, but I'd be really interested to be convinced
>>>> otherwise.
>>>
>>> I could ask you the converse. I'd really be interested to be
>>> convinced how TCP deployment improves scalability.
>>
>> To name two things where TCP would provide better scalability than  
>> UDP
>> for a SIP service provider :
>>
>>   1) Say that your services maximum capacity for non-INVITEs is 1000
>>      requests per second. If you get 2000 in a single second, you
>>      won't only have to deal with all those 2000 requests, but also
>>      with quite a few retransmissions sent after 500ms. With TCP, the
>>      wall isn't as dense.
>>
>>   2) UDP sockets have a shared receive buffer, most commonly around
>>      64k I believe (even though the maximum seems to be 256k on
>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>      average, you have to be able to move messages out of that buffer
>>      pretty fast to keep it from filling up. A full buffer means
>>      dropped packets. I've also heard that on for example Linux,
>>      every packet actually occupies an even 2k (or was it 4k) in the
>>      buffer, so 64k buffer is actually at most 32 packets at a time.
>>
>> And then there are of course other aspects than scalability where UDP
>> is an inferior solution, most notably security related things like  
>> for
>> example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
>>
>> /Fredrik
>>
>>
>> _______________________________________________
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>> Use sip-implementors@cs.columbia.edu for questions on current sip
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From: "Francois Audet" <audet@nortel.com>
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I don't think this is naive at all. I think you hit the nail on the=20
head very sqarely.

=20

> -----Original Message-----
> From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]=20
> Sent: Wednesday, December 06, 2006 15:36
> To: sip@ietf.org
> Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
>    From: Hisham Khartabil <hisham.khartabil@telio.no>
>=20
>    Besides scalability issues, there are no technical issues.
>=20
> Naively, the scaling of UDP-with-outbound doesn't seem to be=20
> any better than TCP-with-outbound.  Can you elucidate the=20
> scalability issues that are causing you problems?
>=20
>    The issues are with having to throw all the infrastructure we
>    currently have out of the window and start all over again. Having
>    110k+ customers live does not make is an easy and justifiable task.
>=20
> Naively, to implement outbound will require changing the=20
> software in the proxies and UAs, regardless of whether it=20
> uses UDP or TCP.  I suppose this can be called "throwing=20
> infrastructure out of the window", but it seems to be a=20
> routine matter.  Can you elucidate why UDP-with-outbound=20
> reduces this burden compared to implementing TCP-with-outbound?
>=20
>    Why do we prefer to run away from the problem and ignore it
>    pretending it does not "really" exist instead of trying to fix it?
>=20
> Can you elucidate what "the problem" is that you are referring to?
>=20
> Dale
>=20
> _______________________________________________
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> Use sipping@ietf.org for new developments on the application of sip
>=20

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Subject: RE: [Sip] I-D: Preventing Fragmentation for Client Initiated
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Maybe I'm missing something in this keep-alive and UDP/TCP debate.

I find the obsession with performance out of place.

All the evidence that I've seen is that in a commercial product, that
the server is=20
really not spending much of it's cycles doing keep-alives. The evidence
is also that
if you are doing keep-alives, the transport will matter even less.

We keep going back to the same issues, with the same conclusions
regarding
performance: the keep-alive mechanism doesn't really matter, and the
transport=20
doesn't matter either.

Yet somehow, we keep agonising over this. In the real world, people
keep on using PING/OPTIONS/short-lived-REGISTER/CRLF/STUN and whatever.

Let's pick something and move on.

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   From: "Francois Audet" <audet@nortel.com>

   I don't think this is naive at all. I think you hit the nail on the
   head very sqarely.

Well, I know I'm not up-to-date on all these issues.  I suspect that
Hisham knows of certain problems that I don't, but we need more
information about them to be able to discuss the situation usefully.

Dale

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Francois Audet wrote:
> Maybe I'm missing something in this keep-alive and UDP/TCP debate.
> 
> I find the obsession with performance out of place.

Performance is important at least for service providers, because they
have to buy the hardware to run the servers, pay the licenses per CPU,
pay the datacenters cost, pay the sysadmins to run the servers and pay
the managers to manage the sysadmins.

So yes, performance is important, but perhaps not for the hardware
vendors, the software vendors and the sysadmins.

> 
> All the evidence that I've seen is that in a commercial product, that
> the server is 
> really not spending much of it's cycles doing keep-alives. The evidence
> is also that
> if you are doing keep-alives, the transport will matter even less.
> 
> We keep going back to the same issues, with the same conclusions
> regarding
> performance: the keep-alive mechanism doesn't really matter, and the
> transport 
> doesn't matter either.
> 
> Yet somehow, we keep agonising over this. In the real world, people
> keep on using PING/OPTIONS/short-lived-REGISTER/CRLF/STUN and whatever.
> 
> Let's pick something and move on.
> 


-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
> I find this whole conversation way off base. If people would like to 
> deprecate UDP, go work on a document to explain that and feel free to 
> bring it to the WG. In the mean time, SIP is specified over both UDP 
> and TCP.

To be more specific, SIP is currently specified so it can use UDP for 
requests smaller than 1300 bytes.  It is specified so it can use TCP 
for requests of any size.

Some people have suggested that user agents SHOULD be able to send and 
receive requests larger than 1300 bytes when using the outbound 
mechanism.  This seems like a logical requirement.  Are you objecting 
to this requirement Cullen?

For the sake of this discussion, lets assume that we have consensus on 
this requirement.  What are the implications?  Well, most UAs 
implementing outbound are doing so because they are behind a NAT or a 
firewall and will not be able to receive inbound TCP flows.  These UAs 
will need to establish a TCP flow if they want to meet this requirement 
of receiving requests larger than 1300 bytes. The logical normative 
statement is to say a UA which cannot (or suspects that it cannot) 
receive an inbound TCP flow from its outbound proxies SHOULD open at 
least one TCP or TLS flow to a server in its outbound-proxy-set.

A few people on this mailing list have suggested that for any user 
agent that actually bothers to open a TCP flow, there is no reason 
whatsoever to open a corresponding UDP flow. I wholeheartedly agree 
with them.

The problem is that this thread strikes right at the heart of our 
community's dilemma. We have a large body of people who have deployed 
UDP almost exclusively and effectively ignore the MUST use (and several 
even ignore the MUST implement) TCP statements in our core spec.  
Ironically, one of the problems which prevents realistic interoperable 
deployment of TCP-based SIP implementations is the lack of a 
standardized mechanism for doing outbound.  [Dean also identified some 
other problems often faced when deploying TCP with programs that use 
"obvious" (Dean actually said "naive") textbook implementation 
techniques.]

I think that a large number of the core spec contributors and seasoned 
implementors now recognize that there are many non-academic problems 
with SIP over UDP and that some day we will want to either deprecate or 
at least seriously curtail the use of UDP.  Like a very heavy drinker 
we know it will be good for us to kick the habit.  We are not quitting 
today. The question before us is are we going to do something in 
outbound that makes it harder for us to quit in the future.

My concrete proposal is the following:
- make CRLF the only keepalive mechanism for TCP and TLS over TCP flow 
(retain STUN for UDP flows)
- make outbound registrations over UDP flows optional for UAs
- include the statement that its is RECOMMENDED that the 
outbound-proxy-set is configured to result in registrations over TCP or 
TLS whenever the UA can't accept incoming TCP flows (the 3GPP 
exception).  explain that for historical purposes a lot of deployments 
still only use UDP, but that we expect/hope those deployments to 
eventually migrate to TCP when doing outbound.

What's in it for the TCP crowd?  They get to implement outbound with no 
UDP if they so desire.  They don't have to implement STUN over TCP. The 
draft clearly explains why TCP is the technically preferred solution.

What's in it for the UDP crowd? They get outbound behavior (at least 
for requests <1300 bytes) specified in the core draft for UDP.  They 
can keep doing what they are doing, but with STUN keepalives if the UA 
and edge proxy both support it.

What's not to like?
thanks,
-rohan


> Most deployments only use UDP, some only use TCP, and I suspect there 
> are some that do both but I am not aware of them. Outbound as it is 
> seems very consistent with 3261 in that it allows a system to use TCP 
> or UDP. Keeping this in mind few comments inline ....
>
> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
>
>> Let's go back to the original topic - what should the draft say about 
>> the transport protocol in relationship with outbound
>>
>> The major use case for outbound is UAs behind NAT / firewalls. Many 
>> of such devices have known issues with UDP fragmentation, so for many 
>> environments TCP will need to be used for messages larger than the 
>> path MTU (often ~1460 bytes in ethernet environments, RFC3261 puts 
>> the limit at 1300).
>
> These known issues with fragmentation might be less common than you 
> think however, outbound does not have anything to do with MTU issues. 
> It does not try and solve them and it does not make them worse.
>
>>
>> If you start from the requirement "MUST only use UDP", then there is 
>> little room for constructive discussions. For UDP-only to work you 
>> will need to ensure that messages either stay below path MTUs, or 
>> disallow NAT. This implies significant limitations for your 
>> deployment environment. In any case, this is not a setup that 
>> outbound should focus on.
>
> MTU is a problem regardless of NATs or Firewalls. Fragmented packets 
> do manager to traverse most NATs and Firewalls.
>
>>
>> In the general case, there are environments that have to deal with 
>> NAT and messages larger than path MTU. This is what outbound was 
>> written for. This also implies a UA that supports TCP. Given that the 
>> UA supports TCP and the fact that presence of devices like stateful 
>> firewalls cannot reliably be detected, the UA SHOULD use TCP (or 
>> another connection oriented transport). If in addition the UA 
>> supports TLS, it SHOULD use TLS.
>
> Again, this topic is not really about outbound. If you think a 3261 UA 
> that supports both UDP and TCP SHOULD use TCP, that is a fine argument 
> to go make in the discussion about use of TCP and UDP but it seems to 
> be an independent choice from anything to do with outbound.
>
>
> Cullen <with my individual hat on>
>
>>
>> I think this is sufficient
>>
>> Regards,
>> Jeroen
>>
>> Fredrik Thulin wrote:
>>> Hisham Khartabil wrote:
>>>>
>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>>> ...
>>>>> I personally believe going TCP-only would actually help deployment
>>>>> rather than hinder it, but I'd be really interested to be convinced
>>>>> otherwise.
>>>>
>>>> I could ask you the converse. I'd really be interested to be
>>>> convinced how TCP deployment improves scalability.
>>>
>>> To name two things where TCP would provide better scalability than 
>>> UDP
>>> for a SIP service provider :
>>>
>>>   1) Say that your services maximum capacity for non-INVITEs is 1000
>>>      requests per second. If you get 2000 in a single second, you
>>>      won't only have to deal with all those 2000 requests, but also
>>>      with quite a few retransmissions sent after 500ms. With TCP, the
>>>      wall isn't as dense.
>>>
>>>   2) UDP sockets have a shared receive buffer, most commonly around
>>>      64k I believe (even though the maximum seems to be 256k on
>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>>      average, you have to be able to move messages out of that buffer
>>>      pretty fast to keep it from filling up. A full buffer means
>>>      dropped packets. I've also heard that on for example Linux,
>>>      every packet actually occupies an even 2k (or was it 4k) in the
>>>      buffer, so 64k buffer is actually at most 32 packets at a time.
>>>
>>> And then there are of course other aspects than scalability where UDP
>>> is an inferior solution, most notably security related things like 
>>> for
>>> example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
>>>
>>> /Fredrik
>>>
>>>
>>> _______________________________________________
>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>> This list is for NEW development of the core SIP Protocol
>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>> Use sipping@ietf.org for new developments on the application of sip


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From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
To: Rohan Mahy <rohan@ekabal.com>, Cullen Jennings <fluffy@cisco.com>
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I like this proposal. It makes it very attractive to implement outbound for
TCP, whilst still having the possibility of doing UDP if you really want to.
Don't forget that one of the use cases for outbound is TLS, where there is
no certificate deployed at the UA. For this case, outbound is needed
regardless of the presence of NAT/firewall. Having to do STUN keep-alive for
this case just because it is needed for the UDP NAT/firewall case is rather
burdensome.

John 

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com] 
> Sent: 07 December 2006 05:16
> To: Cullen Jennings
> Cc: Rohan Mahy; Fredrik Thulin; sip@ietf.org; 
> Markus.Isomaki@nokia.com; peter.leis@siemens.com; aki.niemi@nokia.com
> Subject: Re: [Sip] Re: issue with oubound and the used 
> transport protocol
> 
> 
> On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
> > I find this whole conversation way off base. If people 
> would like to 
> > deprecate UDP, go work on a document to explain that and 
> feel free to 
> > bring it to the WG. In the mean time, SIP is specified over 
> both UDP 
> > and TCP.
> 
> To be more specific, SIP is currently specified so it can use UDP for 
> requests smaller than 1300 bytes.  It is specified so it can use TCP 
> for requests of any size.
> 
> Some people have suggested that user agents SHOULD be able to 
> send and 
> receive requests larger than 1300 bytes when using the outbound 
> mechanism.  This seems like a logical requirement.  Are you objecting 
> to this requirement Cullen?
> 
> For the sake of this discussion, lets assume that we have 
> consensus on 
> this requirement.  What are the implications?  Well, most UAs 
> implementing outbound are doing so because they are behind a NAT or a 
> firewall and will not be able to receive inbound TCP flows.  
> These UAs 
> will need to establish a TCP flow if they want to meet this 
> requirement 
> of receiving requests larger than 1300 bytes. The logical normative 
> statement is to say a UA which cannot (or suspects that it cannot) 
> receive an inbound TCP flow from its outbound proxies SHOULD open at 
> least one TCP or TLS flow to a server in its outbound-proxy-set.
> 
> A few people on this mailing list have suggested that for any user 
> agent that actually bothers to open a TCP flow, there is no reason 
> whatsoever to open a corresponding UDP flow. I wholeheartedly agree 
> with them.
> 
> The problem is that this thread strikes right at the heart of our 
> community's dilemma. We have a large body of people who have deployed 
> UDP almost exclusively and effectively ignore the MUST use 
> (and several 
> even ignore the MUST implement) TCP statements in our core spec.  
> Ironically, one of the problems which prevents realistic 
> interoperable 
> deployment of TCP-based SIP implementations is the lack of a 
> standardized mechanism for doing outbound.  [Dean also 
> identified some 
> other problems often faced when deploying TCP with programs that use 
> "obvious" (Dean actually said "naive") textbook implementation 
> techniques.]
> 
> I think that a large number of the core spec contributors and 
> seasoned 
> implementors now recognize that there are many non-academic problems 
> with SIP over UDP and that some day we will want to either 
> deprecate or 
> at least seriously curtail the use of UDP.  Like a very heavy drinker 
> we know it will be good for us to kick the habit.  We are not 
> quitting 
> today. The question before us is are we going to do something in 
> outbound that makes it harder for us to quit in the future.
> 
> My concrete proposal is the following:
> - make CRLF the only keepalive mechanism for TCP and TLS over 
> TCP flow 
> (retain STUN for UDP flows)
> - make outbound registrations over UDP flows optional for UAs
> - include the statement that its is RECOMMENDED that the 
> outbound-proxy-set is configured to result in registrations 
> over TCP or 
> TLS whenever the UA can't accept incoming TCP flows (the 3GPP 
> exception).  explain that for historical purposes a lot of 
> deployments 
> still only use UDP, but that we expect/hope those deployments to 
> eventually migrate to TCP when doing outbound.
> 
> What's in it for the TCP crowd?  They get to implement 
> outbound with no 
> UDP if they so desire.  They don't have to implement STUN 
> over TCP. The 
> draft clearly explains why TCP is the technically preferred solution.
> 
> What's in it for the UDP crowd? They get outbound behavior (at least 
> for requests <1300 bytes) specified in the core draft for UDP.  They 
> can keep doing what they are doing, but with STUN keepalives 
> if the UA 
> and edge proxy both support it.
> 
> What's not to like?
> thanks,
> -rohan
> 
> 
> > Most deployments only use UDP, some only use TCP, and I 
> suspect there 
> > are some that do both but I am not aware of them. Outbound as it is 
> > seems very consistent with 3261 in that it allows a system 
> to use TCP 
> > or UDP. Keeping this in mind few comments inline ....
> >
> > On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
> >
> >> Let's go back to the original topic - what should the 
> draft say about 
> >> the transport protocol in relationship with outbound
> >>
> >> The major use case for outbound is UAs behind NAT / 
> firewalls. Many 
> >> of such devices have known issues with UDP fragmentation, 
> so for many 
> >> environments TCP will need to be used for messages larger than the 
> >> path MTU (often ~1460 bytes in ethernet environments, RFC3261 puts 
> >> the limit at 1300).
> >
> > These known issues with fragmentation might be less common than you 
> > think however, outbound does not have anything to do with 
> MTU issues. 
> > It does not try and solve them and it does not make them worse.
> >
> >>
> >> If you start from the requirement "MUST only use UDP", 
> then there is 
> >> little room for constructive discussions. For UDP-only to work you 
> >> will need to ensure that messages either stay below path MTUs, or 
> >> disallow NAT. This implies significant limitations for your 
> >> deployment environment. In any case, this is not a setup that 
> >> outbound should focus on.
> >
> > MTU is a problem regardless of NATs or Firewalls. 
> Fragmented packets 
> > do manager to traverse most NATs and Firewalls.
> >
> >>
> >> In the general case, there are environments that have to deal with 
> >> NAT and messages larger than path MTU. This is what outbound was 
> >> written for. This also implies a UA that supports TCP. 
> Given that the 
> >> UA supports TCP and the fact that presence of devices like 
> stateful 
> >> firewalls cannot reliably be detected, the UA SHOULD use TCP (or 
> >> another connection oriented transport). If in addition the UA 
> >> supports TLS, it SHOULD use TLS.
> >
> > Again, this topic is not really about outbound. If you 
> think a 3261 UA 
> > that supports both UDP and TCP SHOULD use TCP, that is a 
> fine argument 
> > to go make in the discussion about use of TCP and UDP but 
> it seems to 
> > be an independent choice from anything to do with outbound.
> >
> >
> > Cullen <with my individual hat on>
> >
> >>
> >> I think this is sufficient
> >>
> >> Regards,
> >> Jeroen
> >>
> >> Fredrik Thulin wrote:
> >>> Hisham Khartabil wrote:
> >>>>
> >>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
> >>> ...
> >>>>> I personally believe going TCP-only would actually help 
> deployment
> >>>>> rather than hinder it, but I'd be really interested to 
> be convinced
> >>>>> otherwise.
> >>>>
> >>>> I could ask you the converse. I'd really be interested to be
> >>>> convinced how TCP deployment improves scalability.
> >>>
> >>> To name two things where TCP would provide better 
> scalability than 
> >>> UDP
> >>> for a SIP service provider :
> >>>
> >>>   1) Say that your services maximum capacity for 
> non-INVITEs is 1000
> >>>      requests per second. If you get 2000 in a single second, you
> >>>      won't only have to deal with all those 2000 
> requests, but also
> >>>      with quite a few retransmissions sent after 500ms. 
> With TCP, the
> >>>      wall isn't as dense.
> >>>
> >>>   2) UDP sockets have a shared receive buffer, most 
> commonly around
> >>>      64k I believe (even though the maximum seems to be 256k on
> >>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
> >>>      average, you have to be able to move messages out of 
> that buffer
> >>>      pretty fast to keep it from filling up. A full buffer means
> >>>      dropped packets. I've also heard that on for example Linux,
> >>>      every packet actually occupies an even 2k (or was it 
> 4k) in the
> >>>      buffer, so 64k buffer is actually at most 32 packets 
> at a time.
> >>>
> >>> And then there are of course other aspects than 
> scalability where UDP
> >>> is an inferior solution, most notably security related 
> things like 
> >>> for
> >>> example spoofed CANCELs (which can't be challenged, 
> RFC3261 #22.1).
> >>>
> >>> /Fredrik
> >>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol
> >>> Use sip-implementors@cs.columbia.edu for questions on current sip
> >>> Use sipping@ietf.org for new developments on the 
> application of sip
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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>> I personally believe going TCP-only would actually help deployment=20
>> rather than hinder it, but I'd be really interested to be convinced=20
>> otherwise.
>
>I could ask you the converse. I'd really be interested to be=20
>convinced how TCP deployment improves scalability.

That's not what I said. I said using TCP helps *deployment*, namely
gives more robust NAT/FW traversal, enables security and just generally
reduces complexity (e.g., makes transport-switching unnecessary).

My thesis here is that going TCP-only makes deployment easier without
incurring any noticeable performance penalty regarding server-side
scalability. And we've done some experiments that show this to be
accurate.

Cheers,
Aki

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From sip-bounces@ietf.org Thu Dec 07 07:48:52 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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>> Not knowing much about the subject, my guess would be that TCP has 
>> overheads, in that one must keep significant state about each 
>> connection and perhaps send keepalive packets periodically.  But 
>> outbound seems to incorporate those already; I don't see how 
>TCP could 
>> be more burdensome than UDP-with-outbound.
>>
>
>The simplest answer, which we've discussed here previously, is 
>that a naive implementation of a proxy using TCP is limited by 
>the number of sockets provided by the OS, which is usually 
>something less than the number of file handles and tends to be 
>less than 64k. This caps the "fan out" of a proxy" at a number 
>smaller than some operators would like. Of course, it's 
>possible to write a more sophisticated proxy using select/poll 
>that doesn't have this limitation, but only on OS that support it.

Let me also chime in here a bit. 

We've done experiments on Linux using a proxy server built on top of our
sofia-sip stack, and have noticed no significant slowdown on an ordinary
desktop PC handling 100s of thousands of registrations simultaneously using
TCP. The stack's Linux implementation uses epoll, which is even more
efficient than select() or poll(). Now, this is not very portable, although
other OSs have similar schemes, like kqueue in BSD. 

Also, if one wanted to write their proxy in Python, the numbers would
probably look a lot different; then again high performance probably isn't
what one is looking for with Python in the first place. Similarly, the
performance of a proxy can suck due to numerous other reasons as well, e.g.,
a frequent dip in a hugely expensive database or RPC backend. We didn't
really touch up on those issues in our experiments, but I suspect many of
these other issues have fairly easy (and expensive) solutions to them. :)

So once and for all, I'd really like to bust the myth, break the urban
legend, and stop the folklore that using TCP wrecks proxy performance. On
modern OSs it really *does not*.

Cheers,
Aki


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AAA7

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From: "Drage, Keith \(Keith\)" <drage@alcatel-lucent.com>
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If any of you have your SIP WG chair in your address books, as opposed
to using the alias provided by IETF, please note that my email address
has changed to that shown below.

While the old address will continue to work until end February, now
would be a good time to update your address books.

The IETF web pages and email lists are now all showing/using the new
address.

Regards

Keith

Keith Drage
drage@alcatel-lucent.com
tel: +44 1793 776249

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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	"Cullen Jennings" <fluffy@cisco.com>
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Hi,=20

>>I find this whole conversation way off base. If people would like to=20
>>deprecate UDP, go work on a document to explain that and=20
>>feel free to bring it to the WG. In the mean time, SIP is specified
over=20
>>both UDP and TCP.
>=20
>To be more specific, SIP is currently specified so it can use=20
>UDP for requests smaller than 1300 bytes.  It is specified so=20
>it can use TCP for requests of any size.
>=20
>Some people have suggested that user agents SHOULD be able to=20
>send and receive requests larger than 1300 bytes when using=20
>the outbound mechanism.  This seems like a logical=20
>requirement.  Are you objecting to this requirement Cullen?

Isn't that an implementation issue?

What if a client "knows" that it will never send or receive requests
larger than 1300 bytes?

>For the sake of this discussion, lets assume that we have=20
>consensus on this requirement.  What are the implications? =20
>Well, most UAs implementing outbound are doing so because=20
>they are behind a NAT or a firewall and will not be able to=20
>receive inbound TCP flows. These UAs will need to establish=20
>a TCP flow if they want to meet this requirement of receiving=20
>requests larger than 1300 bytes. The logical normative=20
>statement is to say a UA which cannot (or suspects that it=20
>cannot) receive an inbound TCP flow from its outbound proxies=20
>SHOULD open at least one TCP or TLS flow to a server in its=20
>outbound-proxy-set.

I think it's quite clear that a client located behdind a NAT/Firewall,
wanting to be able to receive inbound TCP flows, should take actions to
allow that. But, do we need to say more than that?
=20
>A few people on this mailing list have suggested that for any=20
>user agent that actually bothers to open a TCP flow, there is=20
>no reason whatsoever to open a corresponding UDP flow. I=20
>wholeheartedly agree with them.

That's probably true. But, again, is that something we need to specify,
or is it an implementation issue?
=20
>The problem is that this thread strikes right at the heart of=20
>our community's dilemma. We have a large body of people who=20
>have deployed UDP almost exclusively and effectively ignore=20
>the MUST use (and several even ignore the MUST implement) TCP=20
>statements in our core spec. =20
>Ironically, one of the problems which prevents realistic=20
>interoperable deployment of TCP-based SIP implementations is=20
>the lack of a standardized mechanism for doing outbound. =20
>[Dean also identified some other problems often faced when=20
>deploying TCP with programs that use "obvious" (Dean actually=20
>said "naive") textbook implementation techniques.]
>=20
>I think that a large number of the core spec contributors and=20
>seasoned implementors now recognize that there are many=20
>non-academic problems with SIP over UDP and that some day we=20
>will want to either deprecate or at least seriously curtail=20
>the use of UDP.  Like a very heavy drinker we know it will be=20
>good for us to kick the habit.  We are not quitting today.=20
>The question before us is are we going to do something in=20
>outbound that makes it harder for us to quit in the future.

First people wanted to use outbound to "promote" STUN, now to get rid of
UDP. Is that really what we are supposed to be doing?

I still fail to see what the technical problems with outbound and STUN
is. It may be more complex than TCP, but are there really any open
issues to be solved?
=20
>My concrete proposal is the following:
>- make CRLF the only keepalive mechanism for TCP and TLS over=20
>TCP flow (retain STUN for UDP flows)
>- make outbound registrations over UDP flows optional for UAs
>- include the statement that its is RECOMMENDED that the=20
>outbound-proxy-set is configured to result in registrations=20
>over TCP or TLS whenever the UA can't accept incoming TCP=20
>flows (the 3GPP exception).  explain that for historical=20
>purposes a lot of deployments still only use UDP, but that we=20
>expect/hope those deployments to eventually migrate to TCP=20
>when doing outbound.

I don't understand why a standards spec shall talk about what we hope
and expect... As Cullen said, UDP is a valid transport protocol for SIP,
so I don't understand why we need to "justify" anything...
=20
>What's in it for the TCP crowd?  They get to implement=20
>outbound with no UDP if they so desire.  They don't have to=20
>implement STUN over TCP. The draft clearly explains why TCP=20
>is the technically preferred solution.
>=20
>What's in it for the UDP crowd? They get outbound behavior=20
>(at least for requests <1300 bytes) specified in the core=20
>draft for UDP.  They can keep doing what they are doing, but=20
>with STUN keepalives if the UA and edge proxy both support it.

Just to clarify: the edge proxy is still required to implement STUN, in
case someone establishes UDP flows with it, right?

Regards,

Christer






>=20
> > Most deployments only use UDP, some only use TCP, and I=20
> suspect there=20
> > are some that do both but I am not aware of them. Outbound as it is=20
> > seems very consistent with 3261 in that it allows a system=20
> to use TCP=20
> > or UDP. Keeping this in mind few comments inline ....
> >
> > On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
> >
> >> Let's go back to the original topic - what should the=20
> draft say about=20
> >> the transport protocol in relationship with outbound
> >>
> >> The major use case for outbound is UAs behind NAT /=20
> firewalls. Many=20
> >> of such devices have known issues with UDP fragmentation,=20
> so for many=20
> >> environments TCP will need to be used for messages larger than the=20
> >> path MTU (often ~1460 bytes in ethernet environments, RFC3261 puts=20
> >> the limit at 1300).
> >
> > These known issues with fragmentation might be less common than you=20
> > think however, outbound does not have anything to do with=20
> MTU issues.
> > It does not try and solve them and it does not make them worse.
> >
> >>
> >> If you start from the requirement "MUST only use UDP",=20
> then there is=20
> >> little room for constructive discussions. For UDP-only to work you=20
> >> will need to ensure that messages either stay below path MTUs, or=20
> >> disallow NAT. This implies significant limitations for your=20
> >> deployment environment. In any case, this is not a setup that=20
> >> outbound should focus on.
> >
> > MTU is a problem regardless of NATs or Firewalls.=20
> Fragmented packets=20
> > do manager to traverse most NATs and Firewalls.
> >
> >>
> >> In the general case, there are environments that have to deal with=20
> >> NAT and messages larger than path MTU. This is what outbound was=20
> >> written for. This also implies a UA that supports TCP.=20
> Given that the=20
> >> UA supports TCP and the fact that presence of devices like=20
> stateful=20
> >> firewalls cannot reliably be detected, the UA SHOULD use TCP (or=20
> >> another connection oriented transport). If in addition the UA=20
> >> supports TLS, it SHOULD use TLS.
> >
> > Again, this topic is not really about outbound. If you=20
> think a 3261 UA=20
> > that supports both UDP and TCP SHOULD use TCP, that is a=20
> fine argument=20
> > to go make in the discussion about use of TCP and UDP but=20
> it seems to=20
> > be an independent choice from anything to do with outbound.
> >
> >
> > Cullen <with my individual hat on>
> >
> >>
> >> I think this is sufficient
> >>
> >> Regards,
> >> Jeroen
> >>
> >> Fredrik Thulin wrote:
> >>> Hisham Khartabil wrote:
> >>>>
> >>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
> >>> ...
> >>>>> I personally believe going TCP-only would actually help=20
> deployment=20
> >>>>> rather than hinder it, but I'd be really interested to be=20
> >>>>> convinced otherwise.
> >>>>
> >>>> I could ask you the converse. I'd really be interested to be=20
> >>>> convinced how TCP deployment improves scalability.
> >>>
> >>> To name two things where TCP would provide better=20
> scalability than=20
> >>> UDP for a SIP service provider :
> >>>
> >>>   1) Say that your services maximum capacity for=20
> non-INVITEs is 1000
> >>>      requests per second. If you get 2000 in a single second, you
> >>>      won't only have to deal with all those 2000=20
> requests, but also
> >>>      with quite a few retransmissions sent after 500ms.=20
> With TCP, the
> >>>      wall isn't as dense.
> >>>
> >>>   2) UDP sockets have a shared receive buffer, most=20
> commonly around
> >>>      64k I believe (even though the maximum seems to be 256k on
> >>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
> >>>      average, you have to be able to move messages out of=20
> that buffer
> >>>      pretty fast to keep it from filling up. A full buffer means
> >>>      dropped packets. I've also heard that on for example Linux,
> >>>      every packet actually occupies an even 2k (or was it=20
> 4k) in the
> >>>      buffer, so 64k buffer is actually at most 32 packets=20
> at a time.
> >>>
> >>> And then there are of course other aspects than scalability where=20
> >>> UDP is an inferior solution, most notably security related things=20
> >>> like for example spoofed CANCELs (which can't be=20
> challenged, RFC3261=20
> >>> #22.1).
> >>>
> >>> /Fredrik
> >>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Thu Dec 07 10:19:12 2006
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From: "Vijay K. Gurbani" <vkg@alcatel-lucent.com>
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To: Aki Niemi <aki.niemi@nokia.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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Aki Niemi wrote:
> We've done experiments on Linux using a proxy server built on top of our
> sofia-sip stack, and have noticed no significant slowdown on an ordinary
> desktop PC handling 100s of thousands of registrations simultaneously using
> TCP. The stack's Linux implementation uses epoll, which is even more
> efficient than select() or poll(). Now, this is not very portable, although
> other OSs have similar schemes, like kqueue in BSD. 

You can actually hide all of this behind canonical APIs by using
libevent (http://monkey.org/~provos/libevent/).  Your code is
extremely portable then, unless of course, your target is a
RTOS system of some sort where libevent is not compilable (and I
know a couple of the RTOS where this is the case).

Maybe this thread should now migrate to sip-implementors ;-)

- vijay
-- 
Vijay K. Gurbani, Bell Laboratories, Alcatel-Lucent
2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA)
Email: vkg@{alcatel-lucent.com,bell-labs.com,acm.org}
URL:   http://www.alcatel-lucent.com/bell-labs

_______________________________________________
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<DIV><FONT face=3DArial size=3D2>Within step back power army? Gay =
marriage, among=20
ballot, issuesnew judge. Ballot issuesnew judge expels! Will iran, stop =
its!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>September wed sep baghdad expelled =
hussein his and.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Power army general sonthi.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Us, congress, americans vote next. =
Newsmiami zoo=20
hosts poop says, nkorea.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Said about three after pyongyang =
staged!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Wednesday september, wed, sep baghdad, =
expelled=20
hussein his and.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Polls will iran stop its, uranium =
enrichment?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Off, protest government sacked, chief =
throwing=20
monthold? Test abortion issues san francisco apart deciding whether. Gay =

marriage among ballot issuesnew judge.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>No main feeds latest newsmiami, zoo. No =
main feeds=20
latest, newsmiami? Has agreed stalled sixparty on ending =
programmes.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tuesday october, tue oct. Thai bangkok =
thailands,=20
military leaders choose.</FONT></DIV></BODY></HTML>

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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Thu, 7 Dec 2006 09:01:26 -0800
To: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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On Dec 7, 2006, at 6:59, Christer Holmberg ((JO/LMF)) wrote:
> Hi,
>
>>> I find this whole conversation way off base. If people would like to
>>> deprecate UDP, go work on a document to explain that and
>>> feel free to bring it to the WG. In the mean time, SIP is specified
> over
>>> both UDP and TCP.
>>
>> To be more specific, SIP is currently specified so it can use
>> UDP for requests smaller than 1300 bytes.  It is specified so
>> it can use TCP for requests of any size.
>>
>> Some people have suggested that user agents SHOULD be able to
>> send and receive requests larger than 1300 bytes when using
>> the outbound mechanism.  This seems like a logical
>> requirement.  Are you objecting to this requirement Cullen?
>
> Isn't that an implementation issue?

No, it is an interoperability issue which is consistent with our core 
spec.

> What if a client "knows" that it will never send or receive requests
> larger than 1300 bytes?

For most UAs the client will never be able to "know" that.  In fact 
most UAs will be reasonably sure that they WILL be getting requests 
larger than this size.  The only reason this is not a MUST is that we 
have some environments (like 3GPP) that are trying hard to keep 
requests under MTU size and where requests larger than this size are 
rare.

>> For the sake of this discussion, lets assume that we have
>> consensus on this requirement.  What are the implications?
>> Well, most UAs implementing outbound are doing so because
>> they are behind a NAT or a firewall and will not be able to
>> receive inbound TCP flows. These UAs will need to establish
>> a TCP flow if they want to meet this requirement of receiving
>> requests larger than 1300 bytes. The logical normative
>> statement is to say a UA which cannot (or suspects that it
>> cannot) receive an inbound TCP flow from its outbound proxies
>> SHOULD open at least one TCP or TLS flow to a server in its
>> outbound-proxy-set.
>
> I think it's quite clear that a client located behdind a NAT/Firewall,
> wanting to be able to receive inbound TCP flows, should take actions to
> allow that. But, do we need to say more than that?

For most UAs we do.  Therefore a SHOULD is appropriate.

>> A few people on this mailing list have suggested that for any
>> user agent that actually bothers to open a TCP flow, there is
>> no reason whatsoever to open a corresponding UDP flow. I
>> wholeheartedly agree with them.
>
> That's probably true. But, again, is that something we need to specify,
> or is it an implementation issue?

We have normative language in our specifications that recommend all 
sorts of things that are far less important and significant than this.

>> The problem is that this thread strikes right at the heart of
>> our community's dilemma. We have a large body of people who
>> have deployed UDP almost exclusively and effectively ignore
>> the MUST use (and several even ignore the MUST implement) TCP
>> statements in our core spec.
>> Ironically, one of the problems which prevents realistic
>> interoperable deployment of TCP-based SIP implementations is
>> the lack of a standardized mechanism for doing outbound.
>> [Dean also identified some other problems often faced when
>> deploying TCP with programs that use "obvious" (Dean actually
>> said "naive") textbook implementation techniques.]
>>
>> I think that a large number of the core spec contributors and
>> seasoned implementors now recognize that there are many
>> non-academic problems with SIP over UDP and that some day we
>> will want to either deprecate or at least seriously curtail
>> the use of UDP.  Like a very heavy drinker we know it will be
>> good for us to kick the habit.  We are not quitting today.
>> The question before us is are we going to do something in
>> outbound that makes it harder for us to quit in the future.
>
> First people wanted to use outbound to "promote" STUN, now to get rid 
> of
> UDP. Is that really what we are supposed to be doing?

No, we already have several good reasons to get rid of UDP in the 
future.  What I am proposing is that we don't make it harder for us to 
do so when the time comes.

> I still fail to see what the technical problems with outbound and STUN
> is. It may be more complex than TCP, but are there really any open
> issues to be solved?
>> My concrete proposal is the following:
>> - make CRLF the only keepalive mechanism for TCP and TLS over
>> TCP flow (retain STUN for UDP flows)
>> - make outbound registrations over UDP flows optional for UAs
>> - include the statement that its is RECOMMENDED that the
>> outbound-proxy-set is configured to result in registrations
>> over TCP or TLS whenever the UA can't accept incoming TCP
>> flows (the 3GPP exception).  explain that for historical
>> purposes a lot of deployments still only use UDP, but that we
>> expect/hope those deployments to eventually migrate to TCP
>> when doing outbound.
>
> I don't understand why a standards spec shall talk about what we hope
> and expect... As Cullen said, UDP is a valid transport protocol for 
> SIP,
> so I don't understand why we need to "justify" anything...

If we have a SHOULD, we need to justify why someone might not do the 
SHOULD.  In some ways the bar is even higher here since we are 
justifying why this wasn't a MUST.

>> What's in it for the TCP crowd?  They get to implement
>> outbound with no UDP if they so desire.  They don't have to
>> implement STUN over TCP. The draft clearly explains why TCP
>> is the technically preferred solution.
>>
>> What's in it for the UDP crowd? They get outbound behavior
>> (at least for requests <1300 bytes) specified in the core
>> draft for UDP.  They can keep doing what they are doing, but
>> with STUN keepalives if the UA and edge proxy both support it.
>
> Just to clarify: the edge proxy is still required to implement STUN, in
> case someone establishes UDP flows with it, right?

That would be part of my plan.

thanks,
-rohan

> Regards,
>
> Christer
>
>
>
>
>
>
>>
>>> Most deployments only use UDP, some only use TCP, and I
>> suspect there
>>> are some that do both but I am not aware of them. Outbound as it is
>>> seems very consistent with 3261 in that it allows a system
>> to use TCP
>>> or UDP. Keeping this in mind few comments inline ....
>>>
>>> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
>>>
>>>> Let's go back to the original topic - what should the
>> draft say about
>>>> the transport protocol in relationship with outbound
>>>>
>>>> The major use case for outbound is UAs behind NAT /
>> firewalls. Many
>>>> of such devices have known issues with UDP fragmentation,
>> so for many
>>>> environments TCP will need to be used for messages larger than the
>>>> path MTU (often ~1460 bytes in ethernet environments, RFC3261 puts
>>>> the limit at 1300).
>>>
>>> These known issues with fragmentation might be less common than you
>>> think however, outbound does not have anything to do with
>> MTU issues.
>>> It does not try and solve them and it does not make them worse.
>>>
>>>>
>>>> If you start from the requirement "MUST only use UDP",
>> then there is
>>>> little room for constructive discussions. For UDP-only to work you
>>>> will need to ensure that messages either stay below path MTUs, or
>>>> disallow NAT. This implies significant limitations for your
>>>> deployment environment. In any case, this is not a setup that
>>>> outbound should focus on.
>>>
>>> MTU is a problem regardless of NATs or Firewalls.
>> Fragmented packets
>>> do manager to traverse most NATs and Firewalls.
>>>
>>>>
>>>> In the general case, there are environments that have to deal with
>>>> NAT and messages larger than path MTU. This is what outbound was
>>>> written for. This also implies a UA that supports TCP.
>> Given that the
>>>> UA supports TCP and the fact that presence of devices like
>> stateful
>>>> firewalls cannot reliably be detected, the UA SHOULD use TCP (or
>>>> another connection oriented transport). If in addition the UA
>>>> supports TLS, it SHOULD use TLS.
>>>
>>> Again, this topic is not really about outbound. If you
>> think a 3261 UA
>>> that supports both UDP and TCP SHOULD use TCP, that is a
>> fine argument
>>> to go make in the discussion about use of TCP and UDP but
>> it seems to
>>> be an independent choice from anything to do with outbound.
>>>
>>>
>>> Cullen <with my individual hat on>
>>>
>>>>
>>>> I think this is sufficient
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>> Fredrik Thulin wrote:
>>>>> Hisham Khartabil wrote:
>>>>>>
>>>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>>>>> ...
>>>>>>> I personally believe going TCP-only would actually help
>> deployment
>>>>>>> rather than hinder it, but I'd be really interested to be
>>>>>>> convinced otherwise.
>>>>>>
>>>>>> I could ask you the converse. I'd really be interested to be
>>>>>> convinced how TCP deployment improves scalability.
>>>>>
>>>>> To name two things where TCP would provide better
>> scalability than
>>>>> UDP for a SIP service provider :
>>>>>
>>>>>   1) Say that your services maximum capacity for
>> non-INVITEs is 1000
>>>>>      requests per second. If you get 2000 in a single second, you
>>>>>      won't only have to deal with all those 2000
>> requests, but also
>>>>>      with quite a few retransmissions sent after 500ms.
>> With TCP, the
>>>>>      wall isn't as dense.
>>>>>
>>>>>   2) UDP sockets have a shared receive buffer, most
>> commonly around
>>>>>      64k I believe (even though the maximum seems to be 256k on
>>>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>>>>      average, you have to be able to move messages out of
>> that buffer
>>>>>      pretty fast to keep it from filling up. A full buffer means
>>>>>      dropped packets. I've also heard that on for example Linux,
>>>>>      every packet actually occupies an even 2k (or was it
>> 4k) in the
>>>>>      buffer, so 64k buffer is actually at most 32 packets
>> at a time.
>>>>>
>>>>> And then there are of course other aspects than scalability where
>>>>> UDP is an inferior solution, most notably security related things
>>>>> like for example spoofed CANCELs (which can't be
>> challenged, RFC3261
>>>>> #22.1).
>>>>>
>>>>> /Fredrik
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol Use
>>>>> sip-implementors@cs.columbia.edu for questions on current sip Use
>>>>> sipping@ietf.org for new developments on the application of sip
>>
>>
>> _______________________________________________
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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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Rohan Mahy wrote:
>> What if a client "knows" that it will never send or receive requests
>> larger than 1300 bytes?
>
> For most UAs the client will never be able to "know" that.  In fact 
> most UAs will be reasonably sure that they WILL be getting requests 
> larger than this size.  The only reason this is not a MUST is that we 
> have some environments (like 3GPP) that are trying hard to keep 
> requests under MTU size and where requests larger than this size are 
> rare.

And the minute that they interconnect to something that hasn't been
trying, the assumption gets blown clean out of the water, right? That
is the nature of an internet, right? I guess I don't see the point of
special casing something for the range of the moment. It seems like
the SHOULD isn't really a SHOULD at all because we know that
those assumptions will eventually bite because the SIP wg sure isn't
putting any energy in making that true for the Internet at large.

       Mike

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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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I really like this proposal and support it.

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan@ekabal.com]=20
> Sent: Wednesday, December 06, 2006 21:16
> To: Cullen Jennings
> Cc: Rohan Mahy; Fredrik Thulin; sip@ietf.org;=20
> Markus.Isomaki@nokia.com; peter.leis@siemens.com; aki.niemi@nokia.com
> Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport protocol
>
> My concrete proposal is the following:
> - make CRLF the only keepalive mechanism for TCP and TLS over=20
> TCP flow (retain STUN for UDP flows)
> - make outbound registrations over UDP flows optional for UAs
> - include the statement that its is RECOMMENDED that the=20
> outbound-proxy-set is configured to result in registrations=20
> over TCP or TLS whenever the UA can't accept incoming TCP=20
> flows (the 3GPP exception).  explain that for historical=20
> purposes a lot of deployments still only use UDP, but that we=20
> expect/hope those deployments to eventually migrate to TCP=20
> when doing outbound.
>=20
> What's in it for the TCP crowd?  They get to implement=20
> outbound with no UDP if they so desire.  They don't have to=20
> implement STUN over TCP. The draft clearly explains why TCP=20
> is the technically preferred solution.
>=20
> What's in it for the UDP crowd? They get outbound behavior=20
> (at least for requests <1300 bytes) specified in the core=20
> draft for UDP.  They can keep doing what they are doing, but=20
> with STUN keepalives if the UA and edge proxy both support it.
>=20
> What's not to like?
> thanks,
> -rohan
>=20
>=20
> > Most deployments only use UDP, some only use TCP, and I=20
> suspect there=20
> > are some that do both but I am not aware of them. Outbound as it is=20
> > seems very consistent with 3261 in that it allows a system=20
> to use TCP=20
> > or UDP. Keeping this in mind few comments inline ....
> >
> > On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
> >
> >> Let's go back to the original topic - what should the=20
> draft say about=20
> >> the transport protocol in relationship with outbound
> >>
> >> The major use case for outbound is UAs behind NAT /=20
> firewalls. Many=20
> >> of such devices have known issues with UDP fragmentation,=20
> so for many=20
> >> environments TCP will need to be used for messages larger than the=20
> >> path MTU (often ~1460 bytes in ethernet environments, RFC3261 puts=20
> >> the limit at 1300).
> >
> > These known issues with fragmentation might be less common than you=20
> > think however, outbound does not have anything to do with=20
> MTU issues.
> > It does not try and solve them and it does not make them worse.
> >
> >>
> >> If you start from the requirement "MUST only use UDP",=20
> then there is=20
> >> little room for constructive discussions. For UDP-only to work you=20
> >> will need to ensure that messages either stay below path MTUs, or=20
> >> disallow NAT. This implies significant limitations for your=20
> >> deployment environment. In any case, this is not a setup that=20
> >> outbound should focus on.
> >
> > MTU is a problem regardless of NATs or Firewalls.=20
> Fragmented packets=20
> > do manager to traverse most NATs and Firewalls.
> >
> >>
> >> In the general case, there are environments that have to deal with=20
> >> NAT and messages larger than path MTU. This is what outbound was=20
> >> written for. This also implies a UA that supports TCP.=20
> Given that the=20
> >> UA supports TCP and the fact that presence of devices like=20
> stateful=20
> >> firewalls cannot reliably be detected, the UA SHOULD use TCP (or=20
> >> another connection oriented transport). If in addition the UA=20
> >> supports TLS, it SHOULD use TLS.
> >
> > Again, this topic is not really about outbound. If you=20
> think a 3261 UA=20
> > that supports both UDP and TCP SHOULD use TCP, that is a=20
> fine argument=20
> > to go make in the discussion about use of TCP and UDP but=20
> it seems to=20
> > be an independent choice from anything to do with outbound.
> >
> >
> > Cullen <with my individual hat on>
> >
> >>
> >> I think this is sufficient
> >>
> >> Regards,
> >> Jeroen
> >>
> >> Fredrik Thulin wrote:
> >>> Hisham Khartabil wrote:
> >>>>
> >>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
> >>> ...
> >>>>> I personally believe going TCP-only would actually help=20
> deployment=20
> >>>>> rather than hinder it, but I'd be really interested to be=20
> >>>>> convinced otherwise.
> >>>>
> >>>> I could ask you the converse. I'd really be interested to be=20
> >>>> convinced how TCP deployment improves scalability.
> >>>
> >>> To name two things where TCP would provide better=20
> scalability than=20
> >>> UDP for a SIP service provider :
> >>>
> >>>   1) Say that your services maximum capacity for=20
> non-INVITEs is 1000
> >>>      requests per second. If you get 2000 in a single second, you
> >>>      won't only have to deal with all those 2000=20
> requests, but also
> >>>      with quite a few retransmissions sent after 500ms.=20
> With TCP, the
> >>>      wall isn't as dense.
> >>>
> >>>   2) UDP sockets have a shared receive buffer, most=20
> commonly around
> >>>      64k I believe (even though the maximum seems to be 256k on
> >>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
> >>>      average, you have to be able to move messages out of=20
> that buffer
> >>>      pretty fast to keep it from filling up. A full buffer means
> >>>      dropped packets. I've also heard that on for example Linux,
> >>>      every packet actually occupies an even 2k (or was it=20
> 4k) in the
> >>>      buffer, so 64k buffer is actually at most 32 packets=20
> at a time.
> >>>
> >>> And then there are of course other aspects than scalability where=20
> >>> UDP is an inferior solution, most notably security related things=20
> >>> like for example spoofed CANCELs (which can't be=20
> challenged, RFC3261=20
> >>> #22.1).
> >>>
> >>> /Fredrik
> >>>
> >>>
> >>> _______________________________________________
> >>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>> This list is for NEW development of the core SIP Protocol Use=20
> >>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
> >>> sipping@ietf.org for new developments on the application of sip
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Thu, 7 Dec 2006 10:22:38 -0800
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Dear chairs,

Rohan has always disagreed with our TCP keep alive mechanism. He has  
brought this up over many meetings. IT has been discussed many times  
and we have taken WG hums on it. This whole argument that he is  
creating is centered around trying to kill the way TCP keep alive are  
done. He is not alone, others agree with him. The basic argument  
against it is it is a pain to implement - I've implemented and did  
not find it a big deal but agree it is more complicated that  
implementing the Double CRLF approach. We did have reasonable good  
consensus for it in the last WG hum we took on it.

I would like the chairs to make a statement about if they want reopen  
the issue of how to do TCP Keep alives. If yes, I'm glad to present  
on it at next IETF, if not, I want to know that I can just ignore it.

Cullen (with my outbound editor hat on)


On Dec 6, 2006, at 9:15 PM, Rohan Mahy wrote:

>
> On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
>> I find this whole conversation way off base. If people would like  
>> to deprecate UDP, go work on a document to explain that and feel  
>> free to bring it to the WG. In the mean time, SIP is specified  
>> over both UDP and TCP.
>
> To be more specific, SIP is currently specified so it can use UDP  
> for requests smaller than 1300 bytes.  It is specified so it can  
> use TCP for requests of any size.
>
> Some people have suggested that user agents SHOULD be able to send  
> and receive requests larger than 1300 bytes when using the outbound  
> mechanism.  This seems like a logical requirement.  Are you  
> objecting to this requirement Cullen?
yes.

>
> For the sake of this discussion, lets assume that we have consensus  
> on this requirement.  What are the implications?  Well, most UAs  
> implementing outbound are doing so because they are behind a NAT or  
> a firewall and will not be able to receive inbound TCP flows.   
> These UAs will need to establish a TCP flow if they want to meet  
> this requirement of receiving requests larger than 1300 bytes. The  
> logical normative statement is to say a UA which cannot (or  
> suspects that it cannot) receive an inbound TCP flow from its  
> outbound proxies SHOULD open at least one TCP or TLS flow to a  
> server in its outbound-proxy-set.
>
> A few people on this mailing list have suggested that for any user  
> agent that actually bothers to open a TCP flow, there is no reason  
> whatsoever to open a corresponding UDP flow. I wholeheartedly agree  
> with them.
>
> The problem is that this thread strikes right at the heart of our  
> community's dilemma. We have a large body of people who have  
> deployed UDP almost exclusively and effectively ignore the MUST use  
> (and several even ignore the MUST implement) TCP statements in our  
> core spec.  Ironically, one of the problems which prevents  
> realistic interoperable deployment of TCP-based SIP implementations  
> is the lack of a standardized mechanism for doing outbound.  [Dean  
> also identified some other problems often faced when deploying TCP  
> with programs that use "obvious" (Dean actually said "naive")  
> textbook implementation techniques.]
>
> I think that a large number of the core spec contributors and  
> seasoned implementors now recognize that there are many non- 
> academic problems with SIP over UDP and that some day we will want  
> to either deprecate or at least seriously curtail the use of UDP.   
> Like a very heavy drinker we know it will be good for us to kick  
> the habit.  We are not quitting today. The question before us is  
> are we going to do something in outbound that makes it harder for  
> us to quit in the future.
>
> My concrete proposal is the following:
> - make CRLF the only keepalive mechanism for TCP and TLS over TCP  
> flow (retain STUN for UDP flows)
> - make outbound registrations over UDP flows optional for UAs
> - include the statement that its is RECOMMENDED that the outbound- 
> proxy-set is configured to result in registrations over TCP or TLS  
> whenever the UA can't accept incoming TCP flows (the 3GPP  
> exception).  explain that for historical purposes a lot of  
> deployments still only use UDP, but that we expect/hope those  
> deployments to eventually migrate to TCP when doing outbound.
>
> What's in it for the TCP crowd?  They get to implement outbound  
> with no UDP if they so desire.  They don't have to implement STUN  
> over TCP. The draft clearly explains why TCP is the technically  
> preferred solution.
>
> What's in it for the UDP crowd? They get outbound behavior (at  
> least for requests <1300 bytes) specified in the core draft for  
> UDP.  They can keep doing what they are doing, but with STUN  
> keepalives if the UA and edge proxy both support it.
>
> What's not to like?
> thanks,
> -rohan
>
>
>> Most deployments only use UDP, some only use TCP, and I suspect  
>> there are some that do both but I am not aware of them. Outbound  
>> as it is seems very consistent with 3261 in that it allows a  
>> system to use TCP or UDP. Keeping this in mind few comments  
>> inline ....
>>
>> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
>>
>>> Let's go back to the original topic - what should the draft say  
>>> about the transport protocol in relationship with outbound
>>>
>>> The major use case for outbound is UAs behind NAT / firewalls.  
>>> Many of such devices have known issues with UDP fragmentation, so  
>>> for many environments TCP will need to be used for messages  
>>> larger than the path MTU (often ~1460 bytes in ethernet  
>>> environments, RFC3261 puts the limit at 1300).
>>
>> These known issues with fragmentation might be less common than  
>> you think however, outbound does not have anything to do with MTU  
>> issues. It does not try and solve them and it does not make them  
>> worse.
>>
>>>
>>> If you start from the requirement "MUST only use UDP", then there  
>>> is little room for constructive discussions. For UDP-only to work  
>>> you will need to ensure that messages either stay below path  
>>> MTUs, or disallow NAT. This implies significant limitations for  
>>> your deployment environment. In any case, this is not a setup  
>>> that outbound should focus on.
>>
>> MTU is a problem regardless of NATs or Firewalls. Fragmented  
>> packets do manager to traverse most NATs and Firewalls.
>>
>>>
>>> In the general case, there are environments that have to deal  
>>> with NAT and messages larger than path MTU. This is what outbound  
>>> was written for. This also implies a UA that supports TCP. Given  
>>> that the UA supports TCP and the fact that presence of devices  
>>> like stateful firewalls cannot reliably be detected, the UA  
>>> SHOULD use TCP (or another connection oriented transport). If in  
>>> addition the UA supports TLS, it SHOULD use TLS.
>>
>> Again, this topic is not really about outbound. If you think a  
>> 3261 UA that supports both UDP and TCP SHOULD use TCP, that is a  
>> fine argument to go make in the discussion about use of TCP and  
>> UDP but it seems to be an independent choice from anything to do  
>> with outbound.
>>
>>
>> Cullen <with my individual hat on>
>>
>>>
>>> I think this is sufficient
>>>
>>> Regards,
>>> Jeroen
>>>
>>> Fredrik Thulin wrote:
>>>> Hisham Khartabil wrote:
>>>>>
>>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>>>> ...
>>>>>> I personally believe going TCP-only would actually help  
>>>>>> deployment
>>>>>> rather than hinder it, but I'd be really interested to be  
>>>>>> convinced
>>>>>> otherwise.
>>>>>
>>>>> I could ask you the converse. I'd really be interested to be
>>>>> convinced how TCP deployment improves scalability.
>>>>
>>>> To name two things where TCP would provide better scalability  
>>>> than UDP
>>>> for a SIP service provider :
>>>>
>>>>   1) Say that your services maximum capacity for non-INVITEs is  
>>>> 1000
>>>>      requests per second. If you get 2000 in a single second, you
>>>>      won't only have to deal with all those 2000 requests, but also
>>>>      with quite a few retransmissions sent after 500ms. With  
>>>> TCP, the
>>>>      wall isn't as dense.
>>>>
>>>>   2) UDP sockets have a shared receive buffer, most commonly around
>>>>      64k I believe (even though the maximum seems to be 256k on
>>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>>>      average, you have to be able to move messages out of that  
>>>> buffer
>>>>      pretty fast to keep it from filling up. A full buffer means
>>>>      dropped packets. I've also heard that on for example Linux,
>>>>      every packet actually occupies an even 2k (or was it 4k) in  
>>>> the
>>>>      buffer, so 64k buffer is actually at most 32 packets at a  
>>>> time.
>>>>
>>>> And then there are of course other aspects than scalability  
>>>> where UDP
>>>> is an inferior solution, most notably security related things  
>>>> like for
>>>> example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
>>>>
>>>> /Fredrik
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Thu, 7 Dec 2006 10:22:23 -0800
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On Dec 6, 2006, at 9:15 PM, Rohan Mahy wrote:

> To be more specific, SIP is currently specified so it can use UDP  
> for requests smaller than 1300 bytes.  It is specified so it can  
> use TCP for requests of any size.

Agree with above. Note that you said SIP specified a way to do this  
not that deployments must use this.

>
> Some people have suggested that user agents SHOULD be able to send  
> and receive requests larger than 1300 bytes when using the outbound  
> mechanism.  This seems like a logical requirement.  Are you  
> objecting to this requirement Cullen?

yes - I object to this. I think the specification should specify how  
things work for large requests but that an UA should be allowed to  
choose to use only UDP if it wants to accept the limitations that  
this implies.

Again, I don't see why you are trying to tie up deprecating UDP with  
the outbound draft.


_______________________________________________
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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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Dear chairs,

Rohan has always disagreed with our TCP keep alive mechanism. He has  
brought this up over many meetings. IT has been discussed many times  
and we have taken WG hums on it. This whole argument that he is  
creating is centered around trying to kill the way TCP keep alive are  
done. He is not alone, others agree with him. The basic argument  
against it is it is a pain to implement - I've implemented and did  
not find it a big deal but agree it is more complicated that  
implementing the Double CRLF approach. We did have reasonable good  
consensus for it in the last WG hum we took on it.

I would like the chairs to make a statement about if they want reopen  
the issue of how to do TCP Keep alives. If yes, I'm glad to present  
on it at next IETF, if not, I want to know that I can just ignore it.

Cullen (with my outbound editor hat on)


On Dec 6, 2006, at 9:15 PM, Rohan Mahy wrote:

>
> On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
>> I find this whole conversation way off base. If people would like  
>> to deprecate UDP, go work on a document to explain that and feel  
>> free to bring it to the WG. In the mean time, SIP is specified  
>> over both UDP and TCP.
>
> To be more specific, SIP is currently specified so it can use UDP  
> for requests smaller than 1300 bytes.  It is specified so it can  
> use TCP for requests of any size.
>
> Some people have suggested that user agents SHOULD be able to send  
> and receive requests larger than 1300 bytes when using the outbound  
> mechanism.  This seems like a logical requirement.  Are you  
> objecting to this requirement Cullen?
yes.

>
> For the sake of this discussion, lets assume that we have consensus  
> on this requirement.  What are the implications?  Well, most UAs  
> implementing outbound are doing so because they are behind a NAT or  
> a firewall and will not be able to receive inbound TCP flows.   
> These UAs will need to establish a TCP flow if they want to meet  
> this requirement of receiving requests larger than 1300 bytes. The  
> logical normative statement is to say a UA which cannot (or  
> suspects that it cannot) receive an inbound TCP flow from its  
> outbound proxies SHOULD open at least one TCP or TLS flow to a  
> server in its outbound-proxy-set.
>
> A few people on this mailing list have suggested that for any user  
> agent that actually bothers to open a TCP flow, there is no reason  
> whatsoever to open a corresponding UDP flow. I wholeheartedly agree  
> with them.
>
> The problem is that this thread strikes right at the heart of our  
> community's dilemma. We have a large body of people who have  
> deployed UDP almost exclusively and effectively ignore the MUST use  
> (and several even ignore the MUST implement) TCP statements in our  
> core spec.  Ironically, one of the problems which prevents  
> realistic interoperable deployment of TCP-based SIP implementations  
> is the lack of a standardized mechanism for doing outbound.  [Dean  
> also identified some other problems often faced when deploying TCP  
> with programs that use "obvious" (Dean actually said "naive")  
> textbook implementation techniques.]
>
> I think that a large number of the core spec contributors and  
> seasoned implementors now recognize that there are many non- 
> academic problems with SIP over UDP and that some day we will want  
> to either deprecate or at least seriously curtail the use of UDP.   
> Like a very heavy drinker we know it will be good for us to kick  
> the habit.  We are not quitting today. The question before us is  
> are we going to do something in outbound that makes it harder for  
> us to quit in the future.
>
> My concrete proposal is the following:
> - make CRLF the only keepalive mechanism for TCP and TLS over TCP  
> flow (retain STUN for UDP flows)
> - make outbound registrations over UDP flows optional for UAs
> - include the statement that its is RECOMMENDED that the outbound- 
> proxy-set is configured to result in registrations over TCP or TLS  
> whenever the UA can't accept incoming TCP flows (the 3GPP  
> exception).  explain that for historical purposes a lot of  
> deployments still only use UDP, but that we expect/hope those  
> deployments to eventually migrate to TCP when doing outbound.
>
> What's in it for the TCP crowd?  They get to implement outbound  
> with no UDP if they so desire.  They don't have to implement STUN  
> over TCP. The draft clearly explains why TCP is the technically  
> preferred solution.
>
> What's in it for the UDP crowd? They get outbound behavior (at  
> least for requests <1300 bytes) specified in the core draft for  
> UDP.  They can keep doing what they are doing, but with STUN  
> keepalives if the UA and edge proxy both support it.
>
> What's not to like?
> thanks,
> -rohan
>
>
>> Most deployments only use UDP, some only use TCP, and I suspect  
>> there are some that do both but I am not aware of them. Outbound  
>> as it is seems very consistent with 3261 in that it allows a  
>> system to use TCP or UDP. Keeping this in mind few comments  
>> inline ....
>>
>> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
>>
>>> Let's go back to the original topic - what should the draft say  
>>> about the transport protocol in relationship with outbound
>>>
>>> The major use case for outbound is UAs behind NAT / firewalls.  
>>> Many of such devices have known issues with UDP fragmentation, so  
>>> for many environments TCP will need to be used for messages  
>>> larger than the path MTU (often ~1460 bytes in ethernet  
>>> environments, RFC3261 puts the limit at 1300).
>>
>> These known issues with fragmentation might be less common than  
>> you think however, outbound does not have anything to do with MTU  
>> issues. It does not try and solve them and it does not make them  
>> worse.
>>
>>>
>>> If you start from the requirement "MUST only use UDP", then there  
>>> is little room for constructive discussions. For UDP-only to work  
>>> you will need to ensure that messages either stay below path  
>>> MTUs, or disallow NAT. This implies significant limitations for  
>>> your deployment environment. In any case, this is not a setup  
>>> that outbound should focus on.
>>
>> MTU is a problem regardless of NATs or Firewalls. Fragmented  
>> packets do manager to traverse most NATs and Firewalls.
>>
>>>
>>> In the general case, there are environments that have to deal  
>>> with NAT and messages larger than path MTU. This is what outbound  
>>> was written for. This also implies a UA that supports TCP. Given  
>>> that the UA supports TCP and the fact that presence of devices  
>>> like stateful firewalls cannot reliably be detected, the UA  
>>> SHOULD use TCP (or another connection oriented transport). If in  
>>> addition the UA supports TLS, it SHOULD use TLS.
>>
>> Again, this topic is not really about outbound. If you think a  
>> 3261 UA that supports both UDP and TCP SHOULD use TCP, that is a  
>> fine argument to go make in the discussion about use of TCP and  
>> UDP but it seems to be an independent choice from anything to do  
>> with outbound.
>>
>>
>> Cullen <with my individual hat on>
>>
>>>
>>> I think this is sufficient
>>>
>>> Regards,
>>> Jeroen
>>>
>>> Fredrik Thulin wrote:
>>>> Hisham Khartabil wrote:
>>>>>
>>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>>>> ...
>>>>>> I personally believe going TCP-only would actually help  
>>>>>> deployment
>>>>>> rather than hinder it, but I'd be really interested to be  
>>>>>> convinced
>>>>>> otherwise.
>>>>>
>>>>> I could ask you the converse. I'd really be interested to be
>>>>> convinced how TCP deployment improves scalability.
>>>>
>>>> To name two things where TCP would provide better scalability  
>>>> than UDP
>>>> for a SIP service provider :
>>>>
>>>>   1) Say that your services maximum capacity for non-INVITEs is  
>>>> 1000
>>>>      requests per second. If you get 2000 in a single second, you
>>>>      won't only have to deal with all those 2000 requests, but also
>>>>      with quite a few retransmissions sent after 500ms. With  
>>>> TCP, the
>>>>      wall isn't as dense.
>>>>
>>>>   2) UDP sockets have a shared receive buffer, most commonly around
>>>>      64k I believe (even though the maximum seems to be 256k on
>>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>>>      average, you have to be able to move messages out of that  
>>>> buffer
>>>>      pretty fast to keep it from filling up. A full buffer means
>>>>      dropped packets. I've also heard that on for example Linux,
>>>>      every packet actually occupies an even 2k (or was it 4k) in  
>>>> the
>>>>      buffer, so 64k buffer is actually at most 32 packets at a  
>>>> time.
>>>>
>>>> And then there are of course other aspects than scalability  
>>>> where UDP
>>>> is an inferior solution, most notably security related things  
>>>> like for
>>>> example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
>>>>
>>>> /Fredrik
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip

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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Thu, 7 Dec 2006 10:22:23 -0800
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On Dec 6, 2006, at 9:15 PM, Rohan Mahy wrote:

> To be more specific, SIP is currently specified so it can use UDP  
> for requests smaller than 1300 bytes.  It is specified so it can  
> use TCP for requests of any size.

Agree with above. Note that you said SIP specified a way to do this  
not that deployments must use this.

>
> Some people have suggested that user agents SHOULD be able to send  
> and receive requests larger than 1300 bytes when using the outbound  
> mechanism.  This seems like a logical requirement.  Are you  
> objecting to this requirement Cullen?

yes - I object to this. I think the specification should specify how  
things work for large requests but that an UA should be allowed to  
choose to use only UDP if it wants to accept the limitations that  
this implies.

Again, I don't see why you are trying to tie up deprecating UDP with  
the outbound draft.


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------=_NextPart_000_0005_01C71A3D.55A607A0--




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From: "Rohan Mahy" <rohan.mahy@gmail.com>
To: "Cullen Jennings" <fluffy@cisco.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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Hi,

- The argument for using STUN over TCP was "for consistency with UDP".
 However the code necessary to demux STUN and SIP over TCP is pretty
much orthogonal to the code to demux STUN and SIP over UDP.
- In addition to being simpler, CRLF over TCP is backwards compatible
with RFC 3261 implementations.

I believe the WG made the decision it did based on the erroneous
assumption that STUN over TCP would be easier since implementations
would have to implement STUN for UDP anyway.  I would like to reopen
this decision because I believe it was based on faulty data.  In
addition, if we allow UAs to only implement outbound over TCP, this
woudl be yet another assumption that would no longer hold true.

thanks,
-rohan

On 12/7/06, Cullen Jennings <fluffy@cisco.com> wrote:
> Dear chairs,
>
> Rohan has always disagreed with our TCP keep alive mechanism. He has
> brought this up over many meetings. IT has been discussed many times
> and we have taken WG hums on it. This whole argument that he is
> creating is centered around trying to kill the way TCP keep alive are
> done. He is not alone, others agree with him. The basic argument
> against it is it is a pain to implement - I've implemented and did
> not find it a big deal but agree it is more complicated that
> implementing the Double CRLF approach. We did have reasonable good
> consensus for it in the last WG hum we took on it.
>
> I would like the chairs to make a statement about if they want reopen
> the issue of how to do TCP Keep alives. If yes, I'm glad to present
> on it at next IETF, if not, I want to know that I can just ignore it.
>
> Cullen (with my outbound editor hat on)

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Thu Dec 07 17:04:33 2006
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Date: Thu, 7 Dec 2006 14:04:14 -0800
From: "Rohan Mahy" <rohan.mahy@gmail.com>
To: "Cullen Jennings" <fluffy@cisco.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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Hi Cullen,

Please read the following definition carefully and then tell me why
you think my proposed statement ("user agents SHOULD be able to send
and receive requests larger than 1300 bytes when using the outbound
mechanism") is inappropriate.

>From RFC 2119:
>3. SHOULD   This word, or the adjective "RECOMMENDED", mean that there
>   may exist valid reasons in particular circumstances to ignore a
>   particular item, but the full implications must be understood and
>   carefully weighed before choosing a different course.

thanks,
-rohan


On 12/7/06, Cullen Jennings <fluffy@cisco.com> wrote:
> On Dec 6, 2006, at 9:15 PM, Rohan Mahy wrote:
>
> > To be more specific, SIP is currently specified so it can use UDP
> > for requests smaller than 1300 bytes.  It is specified so it can
> > use TCP for requests of any size.
>
> Agree with above. Note that you said SIP specified a way to do this
> not that deployments must use this.
>
> >
> > Some people have suggested that user agents SHOULD be able to send
> > and receive requests larger than 1300 bytes when using the outbound
> > mechanism.  This seems like a logical requirement.  Are you
> > objecting to this requirement Cullen?
>
> yes - I object to this. I think the specification should specify how
> things work for large requests but that an UA should be allowed to
> choose to use only UDP if it wants to accept the limitations that
> this implies.
>
> Again, I don't see why you are trying to tie up deprecating UDP with
> the outbound draft.
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
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From sip-bounces@ietf.org Thu Dec 07 17:27:05 2006
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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Thu, 7 Dec 2006 23:25:57 +0100
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Cc: Rohan Mahy <rohan@ekabal.com>, Fredrik Thulin <ft@it.su.se>,
	SIP <sip@ietf.org>,
	"Isomaki Markus \(Nokia-TP/Espoo\)" <Markus.Isomaki@nokia.com>,
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hum. This is not about deprecating UDP, it's about avoiding unnecessary 
complexity for TCP (and TLS)

Regards,
Jeroen

Rohan Mahy wrote:
> Hi,
>
> - The argument for using STUN over TCP was "for consistency with UDP".
> However the code necessary to demux STUN and SIP over TCP is pretty
> much orthogonal to the code to demux STUN and SIP over UDP.
> - In addition to being simpler, CRLF over TCP is backwards compatible
> with RFC 3261 implementations.
>
> I believe the WG made the decision it did based on the erroneous
> assumption that STUN over TCP would be easier since implementations
> would have to implement STUN for UDP anyway.  I would like to reopen
> this decision because I believe it was based on faulty data.  In
> addition, if we allow UAs to only implement outbound over TCP, this
> woudl be yet another assumption that would no longer hold true.
>
> thanks,
> -rohan
>
> On 12/7/06, Cullen Jennings <fluffy@cisco.com> wrote:
>> Dear chairs,
>>
>> Rohan has always disagreed with our TCP keep alive mechanism. He has
>> brought this up over many meetings. IT has been discussed many times
>> and we have taken WG hums on it. This whole argument that he is
>> creating is centered around trying to kill the way TCP keep alive are
>> done. He is not alone, others agree with him. The basic argument
>> against it is it is a pain to implement - I've implemented and did
>> not find it a big deal but agree it is more complicated that
>> implementing the Double CRLF approach. We did have reasonable good
>> consensus for it in the last WG hum we took on it.
>>
>> I would like the chairs to make a statement about if they want reopen
>> the issue of how to do TCP Keep alives. If yes, I'm glad to present
>> on it at next IETF, if not, I want to know that I can just ignore it.
>>
>> Cullen (with my outbound editor hat on)
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip 


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From sip-bounces@ietf.org Fri Dec 08 00:36:32 2006
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To: "Rohan Mahy" <rohan.mahy@gmail.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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i support rohan's proposal to use CRLF over TCP.

-- juha

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From sip-bounces@ietf.org Fri Dec 08 00:42:45 2006
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Date: Fri, 08 Dec 2006 14:42:23 +0900
From: Eanny Bae <eanny.bae@samsung.com>
To: 'mmusic' <mmusic@ietf.org>, 'sipping' <sipping@ietf.org>,
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Cc: 
Subject: [Sip] The support of multiple connection line 'c=' in SIP UA. 
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Dear all SIP experts,

 

Related with the usage of this multiple connections in SDP, I'd like to get
a clarification from all your SIP experts. 

The IETF RFC 4317 provides some examples for multiple connection lines (i.e.
'c=') in SDP offer and SDP answer. 

Even though IETF RFC 4566 (SDP), there is no clear definition of the usage
of multiple connection lines in SDP. Can it be expected that all SIP UA
support media transmission to/from multiple IP addresses in a single
session. i.e. Can supporting multiple connection('c=') lines be regarded as
a basic capability of the SIP UA?

 

Best Regards,

Eanny Bae 

 

 

-----------------------------------------------------
Eanny Bae (Ms.)
 
Engineer
Samsung Electronics Co., Ltd.
Tel) +82-31-279-5886
Mobile) +82-16-9530-5886
E-Mail)  <mailto:eanny.bae@samsung.com> eanny.bae@samsung.com
-----------------------------------------------------

 


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<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 color=navy
face=Arial><span lang=EN-US style='font-size:10.0pt;font-family:Arial;
color:navy'>Dear all SIP experts,</span></font><font face=Arial><span
lang=EN-GB style='font-family:Arial'><o:p></o:p></span></font></p>

<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 face=Arial><span
lang=EN-GB style='font-size:10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 color=navy
face=Arial><span lang=EN-GB style='font-size:10.0pt;font-family:Arial;
color:navy'>Related with the usage of this multiple connections in SDP,
I&#8217;d like to get a clarification from all your SIP experts. <o:p></o:p></span></font></p>

<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 color=navy
face=Arial><span lang=EN-GB style='font-size:10.0pt;font-family:Arial;
color:navy'>The IETF RFC 4317 provides some examples for multiple connection
lines (i.e. &#8216;c=&#8217;) in SDP offer and SDP answer. <o:p></o:p></span></font></p>

<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 color=navy
face=Arial><span lang=EN-GB style='font-size:10.0pt;font-family:Arial;
color:navy'>Even though IETF RFC 4566 (SDP), there is no clear definition of
the usage of multiple connection lines in SDP. Can it be expected that all SIP
UA support media transmission to/from multiple IP addresses in a single
session. i.e. Can supporting multiple connection(&#8216;c=&#8217;) lines be
regarded as a basic capability of the SIP UA?<o:p></o:p></span></font></p>

<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 color=navy
face=Arial><span lang=EN-US style='font-size:10.0pt;font-family:Arial;
color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 color=navy
face=Arial><span lang=EN-US style='font-size:10.0pt;font-family:Arial;
color:navy'>Best Regards,</span></font><font face=Arial><span lang=EN-GB
style='font-family:Arial'><o:p></o:p></span></font></p>

<p class=MsoNormal align=left style='margin-bottom:6.0pt;text-align:left;
text-autospace:ideograph-other;word-break:keep-all'><font size=2 color=navy
face=Arial><span lang=EN-US style='font-size:10.0pt;font-family:Arial;
color:navy'>Eanny Bae </span></font><font face=Arial><span lang=EN-US
style='font-family:Arial'><o:p></o:p></span></font></p>

<p class=MsoNormal><font size=2 face=Arial><span lang=EN-US style='font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><font size=2 face=&#44404;&#47548;><span lang=EN-US
style='font-size:10.0pt;font-family:Gulim'><o:p>&nbsp;</o:p></span></font></p>

<p class=MsoNormal><strong><b><font size=2 face="Times New Roman"><span
lang=EN-US style='font-size:10.0pt'>-----------------------------------------------------</span></font></b></strong><b><font
face="Times New Roman"><span lang=EN-US style='font-family:"Times New Roman";
font-weight:bold'><br>
<strong><b><font face="Times New Roman">Eanny Bae (Ms.)</font></b></strong><br>
<strong><b><font face="Times New Roman">&nbsp;</font></b></strong><br>
<strong><b><font face="Times New Roman">Engineer</font></b></strong><br>
<strong><b><font face="Times New Roman">Samsung Electronics Co., Ltd.</font></b></strong><br>
<strong><b><font face="Times New Roman">Tel) +82-31-279-5886</font></b></strong><br>
<st1:City w:st="on"><st1:place w:st="on"><strong><b><font face="Times New Roman">Mobile</font></b></strong></st1:place></st1:City><strong><b><font
face="Times New Roman">) +82-16-9530-5886</font></b></strong><br>
<strong><b><font face="Times New Roman">E-Mail) </font></b></strong></span></font></b><span
lang=EN-US><a href="mailto:eanny.bae@samsung.com"><strong><b><font color=black
face="Times New Roman"><span style='color:windowtext;text-decoration:none'>eanny.bae@samsung.com</span></font></b></strong></a><br>
</span><strong><b><font face="Times New Roman"><span lang=EN-US>-----------------------------------------------------</span></font></b></strong><span
lang=EN-US><o:p></o:p></span></p>

<p class=MsoNormal><font size=2 face=&#48148;&#53461;><span lang=EN-US
style='font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></p>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============2145044307==--




From sip-bounces@ietf.org Fri Dec 08 02:13:28 2006
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Subject: RE: [Sip] The support of multiple connection line 'c=' in SIP UA.
Date: Fri, 8 Dec 2006 06:56:28 -0000
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Thread-Topic: [Sip] The support of multiple connection line 'c=' in SIP UA. 
thread-index: Accai6Kiu//xFrVaQ9GydDWZblmaOAACJ/Et
From: "Attila Sipos" <Attila.Sipos@vegastream.com>
To: "Eanny Bae" <eanny.bae@samsung.com>, "mmusic" <mmusic@ietf.org>,
	"sipping" <sipping@ietf.org>, "SIP" <sip@ietf.org>
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1361814038==--



From sip-bounces@ietf.org Fri Dec 08 02:27:23 2006
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Date: Fri, 08 Dec 2006 16:26:12 +0900
From: Eanny Bae <eanny.bae@samsung.com>
Subject: RE: [Sip] The support of multiple connection line 'c=' in SIP UA.
In-reply-to: <8F1F2062CA7D754A838F0E66097D23A0012A3682@veg-brk-svr-pri.vegastream>
To: 'Attila Sipos' <Attila.Sipos@vegastream.com>, 'mmusic' <mmusic@ietf.org>, 
	'sipping' <sipping@ietf.org>, 'SIP' <sip@ietf.org>
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Dear Attila Sipos, 

Thanks for quick answer. 
>From your kindly explanation, I got to clarify that all SIP UA can support
SDP including multiple media description having a different connection
('c=') value per media type. Furthermore, I believe this capability is a
basic capability of SIP UA. 

Best Regards,
Eanny Bae 

-----------------------------------------------------
Eanny Bae (Ms.)
 
Engineer
Samsung Electronics Co., Ltd.
Tel) +82-31-279-5886
Mobile) +82-16-9530-5886
E-Mail) eanny.bae@samsung.com
-----------------------------------------------------

-----Original Message-----
From: Attila Sipos [mailto:Attila.Sipos@vegastream.com] 
Sent: Friday, December 08, 2006 3:56 PM
To: Eanny Bae; mmusic; sipping; SIP
Subject: RE: [Sip] The support of multiple connection line 'c=' in SIP UA.

RFC 4566 says...
 
5.7.  Connection Data ("c=")

      c=<nettype> <addrtype> <connection-address>

   The "c=" field contains connection data.

   A session description MUST contain either at least one "c=" field in
   each media description or a single "c=" field at the session level.
   It MAY contain a single session-level "c=" field and additional "c="
   field(s) per media description, in which case the per-media values
   override the session-level settings for the respective media.

The "session description" above  is the bit before any m= lines.
The "media descriptions" start with an m= line.
So an SDP has 1 session description section and can have
multiple media descriptions.
 
so as examples or positioning of c= lines, you can have:
 
1. There is one c= line at the session description
    Each media description uses the connection information from
    that top c= line.
c=....                     // session description
m=....                    // 1st media description
a=..... 
m=....                    // 2nd media description
a=.....
 
or
 
 
2. Each media description has its own c= line
    (There is no session description c= line):
 
 
                                // session description (no c= line)
m=....                    // 1st media description
c=....
a=..... 
m=....                    // 2nd media description
c=....
a=.....
 
 
or
 
3. There is a session description c= line but the first
    media description has its own c= line , so the first
    media description's c= line overrides the session description c= line.
   The 2nd media description uses the c= line in the session description.
c=....                    // session description
m=....                    // 1st media description
c=....
a=..... 
m=....                    // 2nd media description
a=.....
 
 
Regards,
Attila

	-----Original Message----- 
	From: Eanny Bae [mailto:eanny.bae@samsung.com] 
	Sent: Fri 08/12/2006 05:42 
	To: 'mmusic'; 'sipping'; 'SIP' 
	Cc: 
	Subject: [Sip] The support of multiple connection line 'c=' in SIP
UA. 
	
	

	Dear all SIP experts,

	 

	Related with the usage of this multiple connections in SDP, I'd like
to get a clarification from all your SIP experts. 

	The IETF RFC 4317 provides some examples for multiple connection
lines (i.e. 'c=') in SDP offer and SDP answer. 

	Even though IETF RFC 4566 (SDP), there is no clear definition of the
usage of multiple connection lines in SDP. Can it be expected that all SIP
UA support media transmission to/from multiple IP addresses in a single
session. i.e. Can supporting multiple connection('c=') lines be regarded as
a basic capability of the SIP UA?

	 

	Best Regards,

	Eanny Bae 

	 

	 

	-----------------------------------------------------
	Eanny Bae (Ms.)
	 
	Engineer
	Samsung Electronics Co., Ltd.
	Tel) +82-31-279-5886
	Mobile) +82-16-9530-5886
	E-Mail) eanny.bae@samsung.com <mailto:eanny.bae@samsung.com> 
	-----------------------------------------------------

	 



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Fri Dec 08 03:25:07 2006
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Subject: RE: [Sip] The support of multiple connection line 'c=' in SIP UA.
Date: Fri, 8 Dec 2006 08:17:49 -0000
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Thread-Topic: [Sip] The support of multiple connection line 'c=' in SIP UA.
thread-index: Accai6Kiu//xFrVaQ9GydDWZblmaOAACJ/EtAAEUOGAAAh3oQA==
From: "Attila Sipos" <Attila.Sipos@vegastream.com>
To: "Eanny Bae" <eanny.bae@samsung.com>, "mmusic" <mmusic@ietf.org>,
	"sipping" <sipping@ietf.org>, "SIP" <sip@ietf.org>
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By the way, for interop, most UA's like the style in
example 1.  I have found some UA's that don't like
example 2 where there is no c=3D line in the session description.

(I have not actually tried example 3 with any UA)

Regards,
Attila


-----Original Message-----
From: Eanny Bae [mailto:eanny.bae@samsung.com]
Sent: 08 December 2006 07:26
To: Attila Sipos; 'mmusic'; 'sipping'; 'SIP'
Subject: RE: [Sip] The support of multiple connection line 'c=3D' in SIP
UA.


Dear Attila Sipos,=20

Thanks for quick answer.=20
>From your kindly explanation, I got to clarify that all SIP UA can =
support
SDP including multiple media description having a different connection
('c=3D') value per media type. Furthermore, I believe this capability is =
a
basic capability of SIP UA.=20

Best Regards,
Eanny Bae=20

-----------------------------------------------------
Eanny Bae (Ms.)
=20
Engineer
Samsung Electronics Co., Ltd.
Tel) +82-31-279-5886
Mobile) +82-16-9530-5886
E-Mail) eanny.bae@samsung.com
-----------------------------------------------------

-----Original Message-----
From: Attila Sipos [mailto:Attila.Sipos@vegastream.com]=20
Sent: Friday, December 08, 2006 3:56 PM
To: Eanny Bae; mmusic; sipping; SIP
Subject: RE: [Sip] The support of multiple connection line 'c=3D' in SIP =
UA.

RFC 4566 says...
=20
5.7.  Connection Data ("c=3D")

      c=3D<nettype> <addrtype> <connection-address>

   The "c=3D" field contains connection data.

   A session description MUST contain either at least one "c=3D" field =
in
   each media description or a single "c=3D" field at the session level.
   It MAY contain a single session-level "c=3D" field and additional =
"c=3D"
   field(s) per media description, in which case the per-media values
   override the session-level settings for the respective media.

The "session description" above  is the bit before any m=3D lines.
The "media descriptions" start with an m=3D line.
So an SDP has 1 session description section and can have
multiple media descriptions.
=20
so as examples or positioning of c=3D lines, you can have:
=20
1. There is one c=3D line at the session description
    Each media description uses the connection information from
    that top c=3D line.
c=3D....                     // session description
m=3D....                    // 1st media description
a=3D.....=20
m=3D....                    // 2nd media description
a=3D.....
=20
or
=20
=20
2. Each media description has its own c=3D line
    (There is no session description c=3D line):
=20
=20
                                // session description (no c=3D line)
m=3D....                    // 1st media description
c=3D....
a=3D.....=20
m=3D....                    // 2nd media description
c=3D....
a=3D.....
=20
=20
or
=20
3. There is a session description c=3D line but the first
    media description has its own c=3D line , so the first
    media description's c=3D line overrides the session description c=3D =
line.
   The 2nd media description uses the c=3D line in the session =
description.
c=3D....                    // session description
m=3D....                    // 1st media description
c=3D....
a=3D.....=20
m=3D....                    // 2nd media description
a=3D.....
=20
=20
Regards,
Attila

	-----Original Message-----=20
	From: Eanny Bae [mailto:eanny.bae@samsung.com]=20
	Sent: Fri 08/12/2006 05:42=20
	To: 'mmusic'; 'sipping'; 'SIP'=20
	Cc:=20
	Subject: [Sip] The support of multiple connection line 'c=3D' in SIP
UA.=20
=09
=09

	Dear all SIP experts,

	=20

	Related with the usage of this multiple connections in SDP, I'd like
to get a clarification from all your SIP experts.=20

	The IETF RFC 4317 provides some examples for multiple connection
lines (i.e. 'c=3D') in SDP offer and SDP answer.=20

	Even though IETF RFC 4566 (SDP), there is no clear definition of the
usage of multiple connection lines in SDP. Can it be expected that all =
SIP
UA support media transmission to/from multiple IP addresses in a single
session. i.e. Can supporting multiple connection('c=3D') lines be =
regarded as
a basic capability of the SIP UA?

	=20

	Best Regards,

	Eanny Bae=20

	=20

	=20

	-----------------------------------------------------
	Eanny Bae (Ms.)
	=20
	Engineer
	Samsung Electronics Co., Ltd.
	Tel) +82-31-279-5886
	Mobile) +82-16-9530-5886
	E-Mail) eanny.bae@samsung.com <mailto:eanny.bae@samsung.com>=20
	-----------------------------------------------------

	=20



_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Hello, 

I know that SIP can carry signalling information and media information. Is it possible to make SIP to carry information related to configuration of the end devices or any other information that user wants to exchange. 

Thanks in advance 

with regards 
Raj.
 
---------------------------------
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<SPAN class=postbody>Hello, <BR><BR>I know that SIP can carry signalling information and media information. Is it possible to make SIP to carry information related to configuration of the end devices or any other information that user wants to exchange. <BR><BR>Thanks in advance <BR><BR>with regards <BR>Raj.</SPAN><p>&#32;

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From sip-bounces@ietf.org Fri Dec 08 08:50:47 2006
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 14:48:46 +0100
To: Rohan Mahy <rohan@ekabal.com>
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I like this proposal and support it.

Hisham

On Dec 7, 2006, at 6:15 AM, Rohan Mahy wrote:

>
> On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
>> I find this whole conversation way off base. If people would like to 
>> deprecate UDP, go work on a document to explain that and feel free to 
>> bring it to the WG. In the mean time, SIP is specified over both UDP 
>> and TCP.
>
> To be more specific, SIP is currently specified so it can use UDP for 
> requests smaller than 1300 bytes.  It is specified so it can use TCP 
> for requests of any size.
>
> Some people have suggested that user agents SHOULD be able to send and 
> receive requests larger than 1300 bytes when using the outbound 
> mechanism.  This seems like a logical requirement.  Are you objecting 
> to this requirement Cullen?
>
> For the sake of this discussion, lets assume that we have consensus on 
> this requirement.  What are the implications?  Well, most UAs 
> implementing outbound are doing so because they are behind a NAT or a 
> firewall and will not be able to receive inbound TCP flows.  These UAs 
> will need to establish a TCP flow if they want to meet this 
> requirement of receiving requests larger than 1300 bytes. The logical 
> normative statement is to say a UA which cannot (or suspects that it 
> cannot) receive an inbound TCP flow from its outbound proxies SHOULD 
> open at least one TCP or TLS flow to a server in its 
> outbound-proxy-set.
>
> A few people on this mailing list have suggested that for any user 
> agent that actually bothers to open a TCP flow, there is no reason 
> whatsoever to open a corresponding UDP flow. I wholeheartedly agree 
> with them.
>
> The problem is that this thread strikes right at the heart of our 
> community's dilemma. We have a large body of people who have deployed 
> UDP almost exclusively and effectively ignore the MUST use (and 
> several even ignore the MUST implement) TCP statements in our core 
> spec.  Ironically, one of the problems which prevents realistic 
> interoperable deployment of TCP-based SIP implementations is the lack 
> of a standardized mechanism for doing outbound.  [Dean also identified 
> some other problems often faced when deploying TCP with programs that 
> use "obvious" (Dean actually said "naive") textbook implementation 
> techniques.]
>
> I think that a large number of the core spec contributors and seasoned 
> implementors now recognize that there are many non-academic problems 
> with SIP over UDP and that some day we will want to either deprecate 
> or at least seriously curtail the use of UDP.  Like a very heavy 
> drinker we know it will be good for us to kick the habit.  We are not 
> quitting today. The question before us is are we going to do something 
> in outbound that makes it harder for us to quit in the future.
>
> My concrete proposal is the following:
> - make CRLF the only keepalive mechanism for TCP and TLS over TCP flow 
> (retain STUN for UDP flows)
> - make outbound registrations over UDP flows optional for UAs
> - include the statement that its is RECOMMENDED that the 
> outbound-proxy-set is configured to result in registrations over TCP 
> or TLS whenever the UA can't accept incoming TCP flows (the 3GPP 
> exception).  explain that for historical purposes a lot of deployments 
> still only use UDP, but that we expect/hope those deployments to 
> eventually migrate to TCP when doing outbound.
>
> What's in it for the TCP crowd?  They get to implement outbound with 
> no UDP if they so desire.  They don't have to implement STUN over TCP. 
> The draft clearly explains why TCP is the technically preferred 
> solution.
>
> What's in it for the UDP crowd? They get outbound behavior (at least 
> for requests <1300 bytes) specified in the core draft for UDP.  They 
> can keep doing what they are doing, but with STUN keepalives if the UA 
> and edge proxy both support it.
>
> What's not to like?
> thanks,
> -rohan
>
>
>> Most deployments only use UDP, some only use TCP, and I suspect there 
>> are some that do both but I am not aware of them. Outbound as it is 
>> seems very consistent with 3261 in that it allows a system to use TCP 
>> or UDP. Keeping this in mind few comments inline ....
>>
>> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
>>
>>> Let's go back to the original topic - what should the draft say 
>>> about the transport protocol in relationship with outbound
>>>
>>> The major use case for outbound is UAs behind NAT / firewalls. Many 
>>> of such devices have known issues with UDP fragmentation, so for 
>>> many environments TCP will need to be used for messages larger than 
>>> the path MTU (often ~1460 bytes in ethernet environments, RFC3261 
>>> puts the limit at 1300).
>>
>> These known issues with fragmentation might be less common than you 
>> think however, outbound does not have anything to do with MTU issues. 
>> It does not try and solve them and it does not make them worse.
>>
>>>
>>> If you start from the requirement "MUST only use UDP", then there is 
>>> little room for constructive discussions. For UDP-only to work you 
>>> will need to ensure that messages either stay below path MTUs, or 
>>> disallow NAT. This implies significant limitations for your 
>>> deployment environment. In any case, this is not a setup that 
>>> outbound should focus on.
>>
>> MTU is a problem regardless of NATs or Firewalls. Fragmented packets 
>> do manager to traverse most NATs and Firewalls.
>>
>>>
>>> In the general case, there are environments that have to deal with 
>>> NAT and messages larger than path MTU. This is what outbound was 
>>> written for. This also implies a UA that supports TCP. Given that 
>>> the UA supports TCP and the fact that presence of devices like 
>>> stateful firewalls cannot reliably be detected, the UA SHOULD use 
>>> TCP (or another connection oriented transport). If in addition the 
>>> UA supports TLS, it SHOULD use TLS.
>>
>> Again, this topic is not really about outbound. If you think a 3261 
>> UA that supports both UDP and TCP SHOULD use TCP, that is a fine 
>> argument to go make in the discussion about use of TCP and UDP but it 
>> seems to be an independent choice from anything to do with outbound.
>>
>>
>> Cullen <with my individual hat on>
>>
>>>
>>> I think this is sufficient
>>>
>>> Regards,
>>> Jeroen
>>>
>>> Fredrik Thulin wrote:
>>>> Hisham Khartabil wrote:
>>>>>
>>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>>>> ...
>>>>>> I personally believe going TCP-only would actually help deployment
>>>>>> rather than hinder it, but I'd be really interested to be 
>>>>>> convinced
>>>>>> otherwise.
>>>>>
>>>>> I could ask you the converse. I'd really be interested to be
>>>>> convinced how TCP deployment improves scalability.
>>>>
>>>> To name two things where TCP would provide better scalability than 
>>>> UDP
>>>> for a SIP service provider :
>>>>
>>>>   1) Say that your services maximum capacity for non-INVITEs is 1000
>>>>      requests per second. If you get 2000 in a single second, you
>>>>      won't only have to deal with all those 2000 requests, but also
>>>>      with quite a few retransmissions sent after 500ms. With TCP, 
>>>> the
>>>>      wall isn't as dense.
>>>>
>>>>   2) UDP sockets have a shared receive buffer, most commonly around
>>>>      64k I believe (even though the maximum seems to be 256k on
>>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>>>      average, you have to be able to move messages out of that 
>>>> buffer
>>>>      pretty fast to keep it from filling up. A full buffer means
>>>>      dropped packets. I've also heard that on for example Linux,
>>>>      every packet actually occupies an even 2k (or was it 4k) in the
>>>>      buffer, so 64k buffer is actually at most 32 packets at a time.
>>>>
>>>> And then there are of course other aspects than scalability where 
>>>> UDP
>>>> is an inferior solution, most notably security related things like 
>>>> for
>>>> example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
>>>>
>>>> /Fredrik
>>>>
>>>>
>>>> _______________________________________________
>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>> This list is for NEW development of the core SIP Protocol
>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 14:49:42 +0100
To: Cullen Jennings <fluffy@cisco.com>
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	"Isomaki Markus \(Nokia-TP/Espoo\)" <Markus.Isomaki@nokia.com>,
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On Dec 7, 2006, at 7:22 PM, Cullen Jennings wrote:

>
> Dear chairs,
>
> Rohan has always disagreed with our TCP keep alive mechanism. He has 
> brought this up over many meetings. IT has been discussed many times 
> and we have taken WG hums on it. This whole argument that he is 
> creating is centered around trying to kill the way TCP keep alive are 
> done. He is not alone, others agree with him. The basic argument 
> against it is it is a pain to implement - I've implemented and did not 
> find it a big deal but agree it is more complicated that implementing 
> the Double CRLF approach. We did have reasonable good consensus for it 
> in the last WG hum we took on it.
>
> I would like the chairs to make a statement about if they want reopen 
> the issue of how to do TCP Keep alives. If yes, I'm glad to present on 
> it at next IETF

Please, lets not wait until March to discuss this. We can take a hum 
here on the mailing list. My vote for TCP keepalive is CRLF.

Hisham


> , if not, I want to know that I can just ignore it.
>
> Cullen (with my outbound editor hat on)
>
>
> On Dec 6, 2006, at 9:15 PM, Rohan Mahy wrote:
>
>>
>> On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
>>> I find this whole conversation way off base. If people would like to 
>>> deprecate UDP, go work on a document to explain that and feel free 
>>> to bring it to the WG. In the mean time, SIP is specified over both 
>>> UDP and TCP.
>>
>> To be more specific, SIP is currently specified so it can use UDP for 
>> requests smaller than 1300 bytes.  It is specified so it can use TCP 
>> for requests of any size.
>>
>> Some people have suggested that user agents SHOULD be able to send 
>> and receive requests larger than 1300 bytes when using the outbound 
>> mechanism.  This seems like a logical requirement.  Are you objecting 
>> to this requirement Cullen?
> yes.
>
>>
>> For the sake of this discussion, lets assume that we have consensus 
>> on this requirement.  What are the implications?  Well, most UAs 
>> implementing outbound are doing so because they are behind a NAT or a 
>> firewall and will not be able to receive inbound TCP flows.  These 
>> UAs will need to establish a TCP flow if they want to meet this 
>> requirement of receiving requests larger than 1300 bytes. The logical 
>> normative statement is to say a UA which cannot (or suspects that it 
>> cannot) receive an inbound TCP flow from its outbound proxies SHOULD 
>> open at least one TCP or TLS flow to a server in its 
>> outbound-proxy-set.
>>
>> A few people on this mailing list have suggested that for any user 
>> agent that actually bothers to open a TCP flow, there is no reason 
>> whatsoever to open a corresponding UDP flow. I wholeheartedly agree 
>> with them.
>>
>> The problem is that this thread strikes right at the heart of our 
>> community's dilemma. We have a large body of people who have deployed 
>> UDP almost exclusively and effectively ignore the MUST use (and 
>> several even ignore the MUST implement) TCP statements in our core 
>> spec.  Ironically, one of the problems which prevents realistic 
>> interoperable deployment of TCP-based SIP implementations is the lack 
>> of a standardized mechanism for doing outbound.  [Dean also 
>> identified some other problems often faced when deploying TCP with 
>> programs that use "obvious" (Dean actually said "naive") textbook 
>> implementation techniques.]
>>
>> I think that a large number of the core spec contributors and 
>> seasoned implementors now recognize that there are many non-academic 
>> problems with SIP over UDP and that some day we will want to either 
>> deprecate or at least seriously curtail the use of UDP.  Like a very 
>> heavy drinker we know it will be good for us to kick the habit.  We 
>> are not quitting today. The question before us is are we going to do 
>> something in outbound that makes it harder for us to quit in the 
>> future.
>>
>> My concrete proposal is the following:
>> - make CRLF the only keepalive mechanism for TCP and TLS over TCP 
>> flow (retain STUN for UDP flows)
>> - make outbound registrations over UDP flows optional for UAs
>> - include the statement that its is RECOMMENDED that the 
>> outbound-proxy-set is configured to result in registrations over TCP 
>> or TLS whenever the UA can't accept incoming TCP flows (the 3GPP 
>> exception).  explain that for historical purposes a lot of 
>> deployments still only use UDP, but that we expect/hope those 
>> deployments to eventually migrate to TCP when doing outbound.
>>
>> What's in it for the TCP crowd?  They get to implement outbound with 
>> no UDP if they so desire.  They don't have to implement STUN over 
>> TCP. The draft clearly explains why TCP is the technically preferred 
>> solution.
>>
>> What's in it for the UDP crowd? They get outbound behavior (at least 
>> for requests <1300 bytes) specified in the core draft for UDP.  They 
>> can keep doing what they are doing, but with STUN keepalives if the 
>> UA and edge proxy both support it.
>>
>> What's not to like?
>> thanks,
>> -rohan
>>
>>
>>> Most deployments only use UDP, some only use TCP, and I suspect 
>>> there are some that do both but I am not aware of them. Outbound as 
>>> it is seems very consistent with 3261 in that it allows a system to 
>>> use TCP or UDP. Keeping this in mind few comments inline ....
>>>
>>> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
>>>
>>>> Let's go back to the original topic - what should the draft say 
>>>> about the transport protocol in relationship with outbound
>>>>
>>>> The major use case for outbound is UAs behind NAT / firewalls. Many 
>>>> of such devices have known issues with UDP fragmentation, so for 
>>>> many environments TCP will need to be used for messages larger than 
>>>> the path MTU (often ~1460 bytes in ethernet environments, RFC3261 
>>>> puts the limit at 1300).
>>>
>>> These known issues with fragmentation might be less common than you 
>>> think however, outbound does not have anything to do with MTU 
>>> issues. It does not try and solve them and it does not make them 
>>> worse.
>>>
>>>>
>>>> If you start from the requirement "MUST only use UDP", then there 
>>>> is little room for constructive discussions. For UDP-only to work 
>>>> you will need to ensure that messages either stay below path MTUs, 
>>>> or disallow NAT. This implies significant limitations for your 
>>>> deployment environment. In any case, this is not a setup that 
>>>> outbound should focus on.
>>>
>>> MTU is a problem regardless of NATs or Firewalls. Fragmented packets 
>>> do manager to traverse most NATs and Firewalls.
>>>
>>>>
>>>> In the general case, there are environments that have to deal with 
>>>> NAT and messages larger than path MTU. This is what outbound was 
>>>> written for. This also implies a UA that supports TCP. Given that 
>>>> the UA supports TCP and the fact that presence of devices like 
>>>> stateful firewalls cannot reliably be detected, the UA SHOULD use 
>>>> TCP (or another connection oriented transport). If in addition the 
>>>> UA supports TLS, it SHOULD use TLS.
>>>
>>> Again, this topic is not really about outbound. If you think a 3261 
>>> UA that supports both UDP and TCP SHOULD use TCP, that is a fine 
>>> argument to go make in the discussion about use of TCP and UDP but 
>>> it seems to be an independent choice from anything to do with 
>>> outbound.
>>>
>>>
>>> Cullen <with my individual hat on>
>>>
>>>>
>>>> I think this is sufficient
>>>>
>>>> Regards,
>>>> Jeroen
>>>>
>>>> Fredrik Thulin wrote:
>>>>> Hisham Khartabil wrote:
>>>>>>
>>>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>>>>> ...
>>>>>>> I personally believe going TCP-only would actually help 
>>>>>>> deployment
>>>>>>> rather than hinder it, but I'd be really interested to be 
>>>>>>> convinced
>>>>>>> otherwise.
>>>>>>
>>>>>> I could ask you the converse. I'd really be interested to be
>>>>>> convinced how TCP deployment improves scalability.
>>>>>
>>>>> To name two things where TCP would provide better scalability than 
>>>>> UDP
>>>>> for a SIP service provider :
>>>>>
>>>>>   1) Say that your services maximum capacity for non-INVITEs is 
>>>>> 1000
>>>>>      requests per second. If you get 2000 in a single second, you
>>>>>      won't only have to deal with all those 2000 requests, but also
>>>>>      with quite a few retransmissions sent after 500ms. With TCP, 
>>>>> the
>>>>>      wall isn't as dense.
>>>>>
>>>>>   2) UDP sockets have a shared receive buffer, most commonly around
>>>>>      64k I believe (even though the maximum seems to be 256k on
>>>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>>>>      average, you have to be able to move messages out of that 
>>>>> buffer
>>>>>      pretty fast to keep it from filling up. A full buffer means
>>>>>      dropped packets. I've also heard that on for example Linux,
>>>>>      every packet actually occupies an even 2k (or was it 4k) in 
>>>>> the
>>>>>      buffer, so 64k buffer is actually at most 32 packets at a 
>>>>> time.
>>>>>
>>>>> And then there are of course other aspects than scalability where 
>>>>> UDP
>>>>> is an inferior solution, most notably security related things like 
>>>>> for
>>>>> example spoofed CANCELs (which can't be challenged, RFC3261 #22.1).
>>>>>
>>>>> /Fredrik
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>> This list is for NEW development of the core SIP Protocol
>>>>> Use sip-implementors@cs.columbia.edu for questions on current sip
>>>>> Use sipping@ietf.org for new developments on the application of sip
>


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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip

From sip-bounces@ietf.org Fri Dec 08 08:58:43 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
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Hi,

A couple of questions:

1. What was the reason(s) the CRLF solution is not suitable for UDP?

2. Even for UDP it can be possible to use lower layer keep-alive
mechanisms, e.g. when UDP encapsulated IPSec, is used, so I think one
should be allowed to use that instead of STUN.=20

Regards,

Christer



> -----Original Message-----
> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no]=20
> Sent: 8. joulukuuta 2006 15:49
> To: Rohan Mahy
> Cc: SIP WG
> Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
> I like this proposal and support it.
>=20
> Hisham
>=20
> On Dec 7, 2006, at 6:15 AM, Rohan Mahy wrote:
>=20
> >
> > On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
> >> I find this whole conversation way off base. If people=20
> would like to=20
> >> deprecate UDP, go work on a document to explain that and=20
> feel free to=20
> >> bring it to the WG. In the mean time, SIP is specified=20
> over both UDP=20
> >> and TCP.
> >
> > To be more specific, SIP is currently specified so it can=20
> use UDP for=20
> > requests smaller than 1300 bytes.  It is specified so it=20
> can use TCP=20
> > for requests of any size.
> >
> > Some people have suggested that user agents SHOULD be able=20
> to send and=20
> > receive requests larger than 1300 bytes when using the outbound=20
> > mechanism.  This seems like a logical requirement.  Are you=20
> objecting=20
> > to this requirement Cullen?
> >
> > For the sake of this discussion, lets assume that we have=20
> consensus on=20
> > this requirement.  What are the implications?  Well, most UAs=20
> > implementing outbound are doing so because they are behind=20
> a NAT or a=20
> > firewall and will not be able to receive inbound TCP flows.=20
>  These UAs=20
> > will need to establish a TCP flow if they want to meet this=20
> > requirement of receiving requests larger than 1300 bytes.=20
> The logical=20
> > normative statement is to say a UA which cannot (or suspects that it
> > cannot) receive an inbound TCP flow from its outbound=20
> proxies SHOULD=20
> > open at least one TCP or TLS flow to a server in its=20
> > outbound-proxy-set.
> >
> > A few people on this mailing list have suggested that for any user=20
> > agent that actually bothers to open a TCP flow, there is no reason=20
> > whatsoever to open a corresponding UDP flow. I wholeheartedly agree=20
> > with them.
> >
> > The problem is that this thread strikes right at the heart of our=20
> > community's dilemma. We have a large body of people who=20
> have deployed=20
> > UDP almost exclusively and effectively ignore the MUST use (and=20
> > several even ignore the MUST implement) TCP statements in our core=20
> > spec.  Ironically, one of the problems which prevents realistic=20
> > interoperable deployment of TCP-based SIP implementations=20
> is the lack=20
> > of a standardized mechanism for doing outbound.  [Dean also=20
> identified=20
> > some other problems often faced when deploying TCP with=20
> programs that=20
> > use "obvious" (Dean actually said "naive") textbook implementation=20
> > techniques.]
> >
> > I think that a large number of the core spec contributors=20
> and seasoned=20
> > implementors now recognize that there are many non-academic=20
> problems=20
> > with SIP over UDP and that some day we will want to either=20
> deprecate=20
> > or at least seriously curtail the use of UDP.  Like a very heavy=20
> > drinker we know it will be good for us to kick the habit. =20
> We are not=20
> > quitting today. The question before us is are we going to=20
> do something=20
> > in outbound that makes it harder for us to quit in the future.
> >
> > My concrete proposal is the following:
> > - make CRLF the only keepalive mechanism for TCP and TLS=20
> over TCP flow=20
> > (retain STUN for UDP flows)
> > - make outbound registrations over UDP flows optional for UAs
> > - include the statement that its is RECOMMENDED that the=20
> > outbound-proxy-set is configured to result in registrations=20
> over TCP=20
> > or TLS whenever the UA can't accept incoming TCP flows (the 3GPP=20
> > exception).  explain that for historical purposes a lot of=20
> deployments=20
> > still only use UDP, but that we expect/hope those deployments to=20
> > eventually migrate to TCP when doing outbound.
> >
> > What's in it for the TCP crowd?  They get to implement=20
> outbound with=20
> > no UDP if they so desire.  They don't have to implement=20
> STUN over TCP.
> > The draft clearly explains why TCP is the technically preferred=20
> > solution.
> >
> > What's in it for the UDP crowd? They get outbound behavior=20
> (at least=20
> > for requests <1300 bytes) specified in the core draft for=20
> UDP.  They=20
> > can keep doing what they are doing, but with STUN=20
> keepalives if the UA=20
> > and edge proxy both support it.
> >
> > What's not to like?
> > thanks,
> > -rohan
> >
> >
> >> Most deployments only use UDP, some only use TCP, and I=20
> suspect there=20
> >> are some that do both but I am not aware of them. Outbound=20
> as it is=20
> >> seems very consistent with 3261 in that it allows a system=20
> to use TCP=20
> >> or UDP. Keeping this in mind few comments inline ....
> >>
> >> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
> >>
> >>> Let's go back to the original topic - what should the draft say=20
> >>> about the transport protocol in relationship with outbound
> >>>
> >>> The major use case for outbound is UAs behind NAT /=20
> firewalls. Many=20
> >>> of such devices have known issues with UDP fragmentation, so for=20
> >>> many environments TCP will need to be used for messages=20
> larger than=20
> >>> the path MTU (often ~1460 bytes in ethernet environments, RFC3261=20
> >>> puts the limit at 1300).
> >>
> >> These known issues with fragmentation might be less common=20
> than you=20
> >> think however, outbound does not have anything to do with=20
> MTU issues.
> >> It does not try and solve them and it does not make them worse.
> >>
> >>>
> >>> If you start from the requirement "MUST only use UDP",=20
> then there is=20
> >>> little room for constructive discussions. For UDP-only to=20
> work you=20
> >>> will need to ensure that messages either stay below path MTUs, or=20
> >>> disallow NAT. This implies significant limitations for your=20
> >>> deployment environment. In any case, this is not a setup that=20
> >>> outbound should focus on.
> >>
> >> MTU is a problem regardless of NATs or Firewalls.=20
> Fragmented packets=20
> >> do manager to traverse most NATs and Firewalls.
> >>
> >>>
> >>> In the general case, there are environments that have to=20
> deal with=20
> >>> NAT and messages larger than path MTU. This is what outbound was=20
> >>> written for. This also implies a UA that supports TCP. Given that=20
> >>> the UA supports TCP and the fact that presence of devices like=20
> >>> stateful firewalls cannot reliably be detected, the UA SHOULD use=20
> >>> TCP (or another connection oriented transport). If in=20
> addition the=20
> >>> UA supports TLS, it SHOULD use TLS.
> >>
> >> Again, this topic is not really about outbound. If you=20
> think a 3261=20
> >> UA that supports both UDP and TCP SHOULD use TCP, that is a fine=20
> >> argument to go make in the discussion about use of TCP and=20
> UDP but it=20
> >> seems to be an independent choice from anything to do with=20
> outbound.
> >>
> >>
> >> Cullen <with my individual hat on>
> >>
> >>>
> >>> I think this is sufficient
> >>>
> >>> Regards,
> >>> Jeroen
> >>>
> >>> Fredrik Thulin wrote:
> >>>> Hisham Khartabil wrote:
> >>>>>
> >>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
> >>>> ...
> >>>>>> I personally believe going TCP-only would actually help=20
> >>>>>> deployment rather than hinder it, but I'd be really=20
> interested to=20
> >>>>>> be convinced otherwise.
> >>>>>
> >>>>> I could ask you the converse. I'd really be interested to be=20
> >>>>> convinced how TCP deployment improves scalability.
> >>>>
> >>>> To name two things where TCP would provide better=20
> scalability than=20
> >>>> UDP for a SIP service provider :
> >>>>
> >>>>   1) Say that your services maximum capacity for=20
> non-INVITEs is 1000
> >>>>      requests per second. If you get 2000 in a single second, you
> >>>>      won't only have to deal with all those 2000=20
> requests, but also
> >>>>      with quite a few retransmissions sent after 500ms.=20
> With TCP,=20
> >>>> the
> >>>>      wall isn't as dense.
> >>>>
> >>>>   2) UDP sockets have a shared receive buffer, most=20
> commonly around
> >>>>      64k I believe (even though the maximum seems to be 256k on
> >>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
> >>>>      average, you have to be able to move messages out of that=20
> >>>> buffer
> >>>>      pretty fast to keep it from filling up. A full buffer means
> >>>>      dropped packets. I've also heard that on for example Linux,
> >>>>      every packet actually occupies an even 2k (or was=20
> it 4k) in the
> >>>>      buffer, so 64k buffer is actually at most 32=20
> packets at a time.
> >>>>
> >>>> And then there are of course other aspects than=20
> scalability where=20
> >>>> UDP is an inferior solution, most notably security=20
> related things=20
> >>>> like for example spoofed CANCELs (which can't be challenged,=20
> >>>> RFC3261 #22.1).
> >>>>
> >>>> /Fredrik
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >>>> This list is for NEW development of the core SIP Protocol Use=20
> >>>> sip-implementors@cs.columbia.edu for questions on=20
> current sip Use=20
> >>>> sipping@ietf.org for new developments on the application of sip
> >
>=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 07:33:39 -0800
To: Michael Thomas <mat@cisco.com>
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Hi,

Well in this case, the compression they are using still makes it rare 
to receive > 1300 byte requests over the last hop even when they get 
requests from something in the public Internet that is substantially 
larger. Even over TCP it is still acceptable for a proxy or UA to 
reject a request that is too big.

thanks,
-rohan


On Dec 7, 2006, at 9:25, Michael Thomas wrote:
> Rohan Mahy wrote:
>>> What if a client "knows" that it will never send or receive requests
>>> larger than 1300 bytes?
>>
>> For most UAs the client will never be able to "know" that.  In fact 
>> most UAs will be reasonably sure that they WILL be getting requests 
>> larger than this size.  The only reason this is not a MUST is that we 
>> have some environments (like 3GPP) that are trying hard to keep 
>> requests under MTU size and where requests larger than this size are 
>> rare.
>
> And the minute that they interconnect to something that hasn't been
> trying, the assumption gets blown clean out of the water, right? That
> is the nature of an internet, right? I guess I don't see the point of
> special casing something for the range of the moment. It seems like
> the SHOULD isn't really a SHOULD at all because we know that
> those assumptions will eventually bite because the SIP wg sure isn't
> putting any energy in making that true for the Internet at large.
>
>       Mike


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Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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From sip-bounces@ietf.org Fri Dec 08 10:40:25 2006
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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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On Dec 8, 2006, at 5:57, Christer Holmberg ((JO/LMF)) wrote:

>
> Hi,
>
> A couple of questions:
>
> 1. What was the reason(s) the CRLF solution is not suitable for UDP?

You can't detect that a NAT has rebooted and your binding has changed 
over UDP unless your message contains the reflected address.

> 2. Even for UDP it can be possible to use lower layer keep-alive
> mechanisms, e.g. when UDP encapsulated IPSec, is used, so I think one
> should be allowed to use that instead of STUN.

There is not MUST USE for keepalive=stun, so I think this is tacitly 
permitted anyway.

thanks,
-rohan

> Regards,
>
> Christer
>
>
>
>> -----Original Message-----
>> From: Hisham Khartabil [mailto:hisham.khartabil@telio.no]
>> Sent: 8. joulukuuta 2006 15:49
>> To: Rohan Mahy
>> Cc: SIP WG
>> Subject: Re: [Sip] Re: issue with oubound and the used
>> transport protocol
>>
>> I like this proposal and support it.
>>
>> Hisham
>>
>> On Dec 7, 2006, at 6:15 AM, Rohan Mahy wrote:
>>
>>>
>>> On Dec 6, 2006, at 16:33, Cullen Jennings wrote:
>>>> I find this whole conversation way off base. If people
>> would like to
>>>> deprecate UDP, go work on a document to explain that and
>> feel free to
>>>> bring it to the WG. In the mean time, SIP is specified
>> over both UDP
>>>> and TCP.
>>>
>>> To be more specific, SIP is currently specified so it can
>> use UDP for
>>> requests smaller than 1300 bytes.  It is specified so it
>> can use TCP
>>> for requests of any size.
>>>
>>> Some people have suggested that user agents SHOULD be able
>> to send and
>>> receive requests larger than 1300 bytes when using the outbound
>>> mechanism.  This seems like a logical requirement.  Are you
>> objecting
>>> to this requirement Cullen?
>>>
>>> For the sake of this discussion, lets assume that we have
>> consensus on
>>> this requirement.  What are the implications?  Well, most UAs
>>> implementing outbound are doing so because they are behind
>> a NAT or a
>>> firewall and will not be able to receive inbound TCP flows.
>>  These UAs
>>> will need to establish a TCP flow if they want to meet this
>>> requirement of receiving requests larger than 1300 bytes.
>> The logical
>>> normative statement is to say a UA which cannot (or suspects that it
>>> cannot) receive an inbound TCP flow from its outbound
>> proxies SHOULD
>>> open at least one TCP or TLS flow to a server in its
>>> outbound-proxy-set.
>>>
>>> A few people on this mailing list have suggested that for any user
>>> agent that actually bothers to open a TCP flow, there is no reason
>>> whatsoever to open a corresponding UDP flow. I wholeheartedly agree
>>> with them.
>>>
>>> The problem is that this thread strikes right at the heart of our
>>> community's dilemma. We have a large body of people who
>> have deployed
>>> UDP almost exclusively and effectively ignore the MUST use (and
>>> several even ignore the MUST implement) TCP statements in our core
>>> spec.  Ironically, one of the problems which prevents realistic
>>> interoperable deployment of TCP-based SIP implementations
>> is the lack
>>> of a standardized mechanism for doing outbound.  [Dean also
>> identified
>>> some other problems often faced when deploying TCP with
>> programs that
>>> use "obvious" (Dean actually said "naive") textbook implementation
>>> techniques.]
>>>
>>> I think that a large number of the core spec contributors
>> and seasoned
>>> implementors now recognize that there are many non-academic
>> problems
>>> with SIP over UDP and that some day we will want to either
>> deprecate
>>> or at least seriously curtail the use of UDP.  Like a very heavy
>>> drinker we know it will be good for us to kick the habit.
>> We are not
>>> quitting today. The question before us is are we going to
>> do something
>>> in outbound that makes it harder for us to quit in the future.
>>>
>>> My concrete proposal is the following:
>>> - make CRLF the only keepalive mechanism for TCP and TLS
>> over TCP flow
>>> (retain STUN for UDP flows)
>>> - make outbound registrations over UDP flows optional for UAs
>>> - include the statement that its is RECOMMENDED that the
>>> outbound-proxy-set is configured to result in registrations
>> over TCP
>>> or TLS whenever the UA can't accept incoming TCP flows (the 3GPP
>>> exception).  explain that for historical purposes a lot of
>> deployments
>>> still only use UDP, but that we expect/hope those deployments to
>>> eventually migrate to TCP when doing outbound.
>>>
>>> What's in it for the TCP crowd?  They get to implement
>> outbound with
>>> no UDP if they so desire.  They don't have to implement
>> STUN over TCP.
>>> The draft clearly explains why TCP is the technically preferred
>>> solution.
>>>
>>> What's in it for the UDP crowd? They get outbound behavior
>> (at least
>>> for requests <1300 bytes) specified in the core draft for
>> UDP.  They
>>> can keep doing what they are doing, but with STUN
>> keepalives if the UA
>>> and edge proxy both support it.
>>>
>>> What's not to like?
>>> thanks,
>>> -rohan
>>>
>>>
>>>> Most deployments only use UDP, some only use TCP, and I
>> suspect there
>>>> are some that do both but I am not aware of them. Outbound
>> as it is
>>>> seems very consistent with 3261 in that it allows a system
>> to use TCP
>>>> or UDP. Keeping this in mind few comments inline ....
>>>>
>>>> On Dec 5, 2006, at 1:44 PM, Jeroen van Bemmel wrote:
>>>>
>>>>> Let's go back to the original topic - what should the draft say
>>>>> about the transport protocol in relationship with outbound
>>>>>
>>>>> The major use case for outbound is UAs behind NAT /
>> firewalls. Many
>>>>> of such devices have known issues with UDP fragmentation, so for
>>>>> many environments TCP will need to be used for messages
>> larger than
>>>>> the path MTU (often ~1460 bytes in ethernet environments, RFC3261
>>>>> puts the limit at 1300).
>>>>
>>>> These known issues with fragmentation might be less common
>> than you
>>>> think however, outbound does not have anything to do with
>> MTU issues.
>>>> It does not try and solve them and it does not make them worse.
>>>>
>>>>>
>>>>> If you start from the requirement "MUST only use UDP",
>> then there is
>>>>> little room for constructive discussions. For UDP-only to
>> work you
>>>>> will need to ensure that messages either stay below path MTUs, or
>>>>> disallow NAT. This implies significant limitations for your
>>>>> deployment environment. In any case, this is not a setup that
>>>>> outbound should focus on.
>>>>
>>>> MTU is a problem regardless of NATs or Firewalls.
>> Fragmented packets
>>>> do manager to traverse most NATs and Firewalls.
>>>>
>>>>>
>>>>> In the general case, there are environments that have to
>> deal with
>>>>> NAT and messages larger than path MTU. This is what outbound was
>>>>> written for. This also implies a UA that supports TCP. Given that
>>>>> the UA supports TCP and the fact that presence of devices like
>>>>> stateful firewalls cannot reliably be detected, the UA SHOULD use
>>>>> TCP (or another connection oriented transport). If in
>> addition the
>>>>> UA supports TLS, it SHOULD use TLS.
>>>>
>>>> Again, this topic is not really about outbound. If you
>> think a 3261
>>>> UA that supports both UDP and TCP SHOULD use TCP, that is a fine
>>>> argument to go make in the discussion about use of TCP and
>> UDP but it
>>>> seems to be an independent choice from anything to do with
>> outbound.
>>>>
>>>>
>>>> Cullen <with my individual hat on>
>>>>
>>>>>
>>>>> I think this is sufficient
>>>>>
>>>>> Regards,
>>>>> Jeroen
>>>>>
>>>>> Fredrik Thulin wrote:
>>>>>> Hisham Khartabil wrote:
>>>>>>>
>>>>>>> On Dec 5, 2006, at 3:33 PM, <aki.niemi@nokia.com> wrote:
>>>>>> ...
>>>>>>>> I personally believe going TCP-only would actually help
>>>>>>>> deployment rather than hinder it, but I'd be really
>> interested to
>>>>>>>> be convinced otherwise.
>>>>>>>
>>>>>>> I could ask you the converse. I'd really be interested to be
>>>>>>> convinced how TCP deployment improves scalability.
>>>>>>
>>>>>> To name two things where TCP would provide better
>> scalability than
>>>>>> UDP for a SIP service provider :
>>>>>>
>>>>>>   1) Say that your services maximum capacity for
>> non-INVITEs is 1000
>>>>>>      requests per second. If you get 2000 in a single second, you
>>>>>>      won't only have to deal with all those 2000
>> requests, but also
>>>>>>      with quite a few retransmissions sent after 500ms.
>> With TCP,
>>>>>> the
>>>>>>      wall isn't as dense.
>>>>>>
>>>>>>   2) UDP sockets have a shared receive buffer, most
>> commonly around
>>>>>>      64k I believe (even though the maximum seems to be 256k on
>>>>>>      FreeBSD at least). With SIP messages of perhaps 700 bytes on
>>>>>>      average, you have to be able to move messages out of that
>>>>>> buffer
>>>>>>      pretty fast to keep it from filling up. A full buffer means
>>>>>>      dropped packets. I've also heard that on for example Linux,
>>>>>>      every packet actually occupies an even 2k (or was
>> it 4k) in the
>>>>>>      buffer, so 64k buffer is actually at most 32
>> packets at a time.
>>>>>>
>>>>>> And then there are of course other aspects than
>> scalability where
>>>>>> UDP is an inferior solution, most notably security
>> related things
>>>>>> like for example spoofed CANCELs (which can't be challenged,
>>>>>> RFC3261 #22.1).
>>>>>>
>>>>>> /Fredrik
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>>>>>> This list is for NEW development of the core SIP Protocol Use
>>>>>> sip-implementors@cs.columbia.edu for questions on
>> current sip Use
>>>>>> sipping@ietf.org for new developments on the application of sip
>>>
>>
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use
>> sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>>


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I agree with Attila. But I'm not sure from your comments if you are 
understanding fully, so let me try again.

An SDP can have one or more media descriptions, each beginning with an 
m= line. (Actually it can have zero media descriptions, but that isn't 
very interesting.)

Each media description needs connection data. If there is more than one 
media description, then the connection data for each may be different, 
or multiple media descriptions may have the same connection data.

If there is a c= line following an m= line, and before another m= line, 
then that provides the connection data for that media description. If 
there is no c= line following the m= line, then the media description 
uses the c= line from the session description - the one before the first 
m= line.

It is legal, for each media description to contain its own c= line, and 
for there also to be a c= line in the session description, even though 
in that case the one in the session description serves no purpose.

A UA must be prepared to receive SDP formatted in any of these ways.

	Paul

Eanny Bae wrote:
> Dear Attila Sipos, 
> 
> Thanks for quick answer. 
>>From your kindly explanation, I got to clarify that all SIP UA can support
> SDP including multiple media description having a different connection
> ('c=') value per media type. Furthermore, I believe this capability is a
> basic capability of SIP UA. 
> 
> Best Regards,
> Eanny Bae 
> 
> -----------------------------------------------------
> Eanny Bae (Ms.)
>  
> Engineer
> Samsung Electronics Co., Ltd.
> Tel) +82-31-279-5886
> Mobile) +82-16-9530-5886
> E-Mail) eanny.bae@samsung.com
> -----------------------------------------------------
> 
> -----Original Message-----
> From: Attila Sipos [mailto:Attila.Sipos@vegastream.com] 
> Sent: Friday, December 08, 2006 3:56 PM
> To: Eanny Bae; mmusic; sipping; SIP
> Subject: RE: [Sip] The support of multiple connection line 'c=' in SIP UA.
> 
> RFC 4566 says...
>  
> 5.7.  Connection Data ("c=")
> 
>       c=<nettype> <addrtype> <connection-address>
> 
>    The "c=" field contains connection data.
> 
>    A session description MUST contain either at least one "c=" field in
>    each media description or a single "c=" field at the session level.
>    It MAY contain a single session-level "c=" field and additional "c="
>    field(s) per media description, in which case the per-media values
>    override the session-level settings for the respective media.
> 
> The "session description" above  is the bit before any m= lines.
> The "media descriptions" start with an m= line.
> So an SDP has 1 session description section and can have
> multiple media descriptions.
>  
> so as examples or positioning of c= lines, you can have:
>  
> 1. There is one c= line at the session description
>     Each media description uses the connection information from
>     that top c= line.
> c=....                     // session description
> m=....                    // 1st media description
> a=..... 
> m=....                    // 2nd media description
> a=.....
>  
> or
>  
>  
> 2. Each media description has its own c= line
>     (There is no session description c= line):
>  
>  
>                                 // session description (no c= line)
> m=....                    // 1st media description
> c=....
> a=..... 
> m=....                    // 2nd media description
> c=....
> a=.....
>  
>  
> or
>  
> 3. There is a session description c= line but the first
>     media description has its own c= line , so the first
>     media description's c= line overrides the session description c= line.
>    The 2nd media description uses the c= line in the session description.
> c=....                    // session description
> m=....                    // 1st media description
> c=....
> a=..... 
> m=....                    // 2nd media description
> a=.....
>  
>  
> Regards,
> Attila
> 
> 	-----Original Message----- 
> 	From: Eanny Bae [mailto:eanny.bae@samsung.com] 
> 	Sent: Fri 08/12/2006 05:42 
> 	To: 'mmusic'; 'sipping'; 'SIP' 
> 	Cc: 
> 	Subject: [Sip] The support of multiple connection line 'c=' in SIP
> UA. 
> 	
> 	
> 
> 	Dear all SIP experts,
> 
> 	 
> 
> 	Related with the usage of this multiple connections in SDP, I'd like
> to get a clarification from all your SIP experts. 
> 
> 	The IETF RFC 4317 provides some examples for multiple connection
> lines (i.e. 'c=') in SDP offer and SDP answer. 
> 
> 	Even though IETF RFC 4566 (SDP), there is no clear definition of the
> usage of multiple connection lines in SDP. Can it be expected that all SIP
> UA support media transmission to/from multiple IP addresses in a single
> session. i.e. Can supporting multiple connection('c=') lines be regarded as
> a basic capability of the SIP UA?
> 
> 	 
> 
> 	Best Regards,
> 
> 	Eanny Bae 
> 
> 	 
> 
> 	 
> 
> 	-----------------------------------------------------
> 	Eanny Bae (Ms.)
> 	 
> 	Engineer
> 	Samsung Electronics Co., Ltd.
> 	Tel) +82-31-279-5886
> 	Mobile) +82-16-9530-5886
> 	E-Mail) eanny.bae@samsung.com <mailto:eanny.bae@samsung.com> 
> 	-----------------------------------------------------
> 
> 	 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

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From sip-bounces@ietf.org Fri Dec 08 12:00:46 2006
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I suspect what you need is=20

http://www.ietf.org/internet-drafts/draft-ietf-sipping-config-framework-0=
9.txt

Regards

Keith

P.S. If you ever intend using INFO for any application, I suggest you =
test the water with an internet-draft on the subject before doing so. =
Many proposed INFO usages are regarded as inappropriate to SIP.


> -----Original Message-----
> From: Emmanuel BUU [mailto:emmanuel.buu@ives.fr]=20
> Sent: 08 December 2006 14:04
> To: Raj
> Cc: SIP forum; SIP Proposal Investigation; SIP forum; SIP Resiprocate
> Subject: [Sipping] Re: [reSIProcate] Information carried by=20
> SIP Message -regarding
>=20
> There is an INFO message that can be used for this purpose
>=20
> Raj a =E9crit :
> > Hello,
> >
> > I know that SIP can carry signalling information and media=20
> > information. Is it possible to make SIP to carry=20
> information related=20
> > to configuration of the end devices or any other=20
> information that user=20
> > wants to exchange.
> >
> > Thanks in advance
> >
> > with regards
> > Raj.
> >
> >=20
> ----------------------------------------------------------------------
> > -- Access over 1 million songs - Yahoo! Music Unlimited.
> >=20
> =
<http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=3D36035/*http://music.y
> > ahoo.com/unlimited/>
> >
> >=20
> ----------------------------------------------------------------------
> > --
> >
> > _______________________________________________
> > resiprocate-devel mailing list
> > resiprocate-devel@list.resiprocate.org
> > https://list.resiprocate.org/mailman/listinfo/resiprocate-devel
>=20
>=20
> --
> logo
>=20
> Emmanuel BUU <mailto:emmanuel.buu@ives.fr> - G=E9rant / General=20
> Manager - IV=E8S <http://www.ives.fr/>
>=20
>=20
>=20
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP=20
> Use sip-implementors@cs.columbia.edu for questions on current=20
> sip Use sip@ietf.org for new developments of core SIP
>=20

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On Dec 8, 2006, at 5:49 AM, Hisham Khartabil wrote:

> Please, lets not wait until March to discuss this. We can take a  
> hum here on the mailing list

We had most of a room hum the other direction so, which did leave me  
thinking, assuming we don't get a single no on the the list, how many  
yes do think the chairs should be looking for to overturn this? I  
don't really want to know the answer to this as this is something  
that only the chairs need to figure out but it is an interesting  
question to think about when thinking about reopening this.

And just so people understand my position on this, if we reopen this  
issue, I will be voting for whichever path I think can get the draft  
to the IESG the soonest. I've implemented both and they are both  
trivial to implement. I really believe that what we need is to stop  
delaying this draft. 

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On Dec 7, 2006, at 2:25 PM, Jeroen van Bemmel wrote:

> hum. This is not about deprecating UDP, it's about avoiding  
> unnecessary complexity for TCP (and TLS)

I can understand the argument that what keep alive we use in TCP is  
about avoiding unnecessary complexity for TCP and I'm fine with  
discussing that (thus my email to the chairs) however, Rohan's  
proposal contained other things that are separate issues. And how  
keep-alives work is very relevant to this draft. What protocols  
operators choose to use is an issue orthogonal to this draft (and a  
very complicated issue).

>
> Regards,
> Jeroen
>
> Rohan Mahy wrote:
>> Hi,
>>
>> - The argument for using STUN over TCP was "for consistency with  
>> UDP".
>> However the code necessary to demux STUN and SIP over TCP is pretty
>> much orthogonal to the code to demux STUN and SIP over UDP.
>> - In addition to being simpler, CRLF over TCP is backwards compatible
>> with RFC 3261 implementations.
>>
>> I believe the WG made the decision it did based on the erroneous
>> assumption that STUN over TCP would be easier since implementations
>> would have to implement STUN for UDP anyway.  I would like to reopen
>> this decision because I believe it was based on faulty data.  In
>> addition, if we allow UAs to only implement outbound over TCP, this
>> woudl be yet another assumption that would no longer hold true.
>>
>> thanks,
>> -rohan
>>
>> On 12/7/06, Cullen Jennings <fluffy@cisco.com> wrote:
>>> Dear chairs,
>>>
>>> Rohan has always disagreed with our TCP keep alive mechanism. He has
>>> brought this up over many meetings. IT has been discussed many times
>>> and we have taken WG hums on it. This whole argument that he is
>>> creating is centered around trying to kill the way TCP keep alive  
>>> are
>>> done. He is not alone, others agree with him. The basic argument
>>> against it is it is a pain to implement - I've implemented and did
>>> not find it a big deal but agree it is more complicated that
>>> implementing the Double CRLF approach. We did have reasonable good
>>> consensus for it in the last WG hum we took on it.
>>>
>>> I would like the chairs to make a statement about if they want  
>>> reopen
>>> the issue of how to do TCP Keep alives. If yes, I'm glad to present
>>> on it at next IETF, if not, I want to know that I can just ignore  
>>> it.
>>>
>>> Cullen (with my outbound editor hat on)
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 10:10:17 -0800
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On Nov 30, 2006, at 9:45 AM, Francois Audet wrote:

> Didn't we agree way back NOT to use PING, CRLF, etc., and use STUN
> instead?
>
> Why is this debate being re-opened?
>
> Remember: I favored the non-STUN TCP approach. However, if we can't
> stick to a decision,
> we'll debate this forever. It already feels like it.

I was thinking, wow, we won't really repeat the exact same  
conversation over again. But then I read the thread below and  
realized that I can not imagine this conversation going any  
differently than the last several times we have done it.


On Dec 8, 2006, at 7:40 AM, Rohan Mahy wrote:
>
> On Dec 8, 2006, at 5:57, Christer Holmberg ((JO/LMF)) wrote:
>
>>
>> Hi,
>>
>> A couple of questions:
>>
>> 1. What was the reason(s) the CRLF solution is not suitable for UDP?
>
> You can't detect that a NAT has rebooted and your binding has  
> changed over UDP unless your message contains the reflected address.
>
>> 2. Even for UDP it can be possible to use lower layer keep-alive
>> mechanisms, e.g. when UDP encapsulated IPSec, is used, so I think one
>> should be allowed to use that instead of STUN.
>
> There is not MUST USE for keepalive=stun, so I think this is  
> tacitly permitted anyway.
>
> thanks,
> -rohan
>
>> Regards,
>>
>> Christer


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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From: Byron Campen <bcampen@estacado.net>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 16:21:44 -0600
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	This looks sensible to me. But, I would like clarification on one  
point; are you proposing that server implementations of outbound  
(registrar, edge proxy) must (still) support outbound over UDP? Or  
would it be optional for these elements as well?

Best regards,
Byron Campen


> My concrete proposal is the following:
> - make CRLF the only keepalive mechanism for TCP and TLS over TCP  
> flow (retain STUN for UDP flows)
> - make outbound registrations over UDP flows optional for UAs
> - include the statement that its is RECOMMENDED that the outbound- 
> proxy-set is configured to result in registrations over TCP or TLS  
> whenever the UA can't accept incoming TCP flows (the 3GPP  
> exception).  explain that for historical purposes a lot of  
> deployments still only use UDP, but that we expect/hope those  
> deployments to eventually migrate to TCP when doing outbound.
>
> What's in it for the TCP crowd?  They get to implement outbound  
> with no UDP if they so desire.  They don't have to implement STUN  
> over TCP. The draft clearly explains why TCP is the technically  
> preferred solution.
>
> What's in it for the UDP crowd? They get outbound behavior (at  
> least for requests <1300 bytes) specified in the core draft for  
> UDP.  They can keep doing what they are doing, but with STUN  
> keepalives if the UA and edge proxy both support it.
>
> What's not to like?
> thanks,
> -rohan


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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 18:31:12 -0600
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Thread-Topic: [Sip] Re: issue with oubound and the used transport protocol
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From: "Francois Audet" <audet@nortel.com>
To: "Cullen Jennings" <fluffy@cisco.com>,
	"Hisham Khartabil" <hisham.khartabil@telio.no>
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	"Isomaki Markus \(Nokia-TP/Espoo\)" <Markus.Isomaki@nokia.com>,
	Peter Leis <peter.leis@siemens.com>, Aki Niemi <aki.niemi@nokia.com>,
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Cullen,

My best guest on why this is happening is that people used to be
thinking more
about UDP than TCP and consequently expressed a preference for the STUN
approach.

It seems now that more people are paying attention to TCP because of a
bunch of issues
with UDP (including fragmentation, and TLS usage catching up since it
took us so long
to finalize this document). Many are reaching the conclusion that=20
the CRLF approach makes more sense for TCP.

It seems that Rohan's last proposal of using CRLF for TCP, and STUN for
UDP to be
reasonable and logical. And it seems to have lots of support on the
list.


> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: Friday, December 08, 2006 09:43
> To: Hisham Khartabil
> Cc: Rohan Mahy; Fredrik Thulin; SIP; Isomaki Markus=20
> (Nokia-TP/Espoo); Peter Leis; Aki Niemi; Dean Willis; Keith Drage
> Subject: Re: [Sip] Re: issue with oubound and the used=20
> transport protocol
>=20
>=20
> On Dec 8, 2006, at 5:49 AM, Hisham Khartabil wrote:
>=20
> > Please, lets not wait until March to discuss this. We can=20
> take a hum=20
> > here on the mailing list
>=20
> We had most of a room hum the other direction so, which did=20
> leave me thinking, assuming we don't get a single no on the=20
> the list, how many yes do think the chairs should be looking=20
> for to overturn this? I don't really want to know the answer=20
> to this as this is something that only the chairs need to=20
> figure out but it is an interesting question to think about=20
> when thinking about reopening this.
>=20
> And just so people understand my position on this, if we=20
> reopen this issue, I will be voting for whichever path I=20
> think can get the draft to the IESG the soonest. I've=20
> implemented both and they are both trivial to implement. I=20
> really believe that what we need is to stop delaying this draft.=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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From: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 20:01:31 -0600
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On Dec 8, 2006, at 11:42 AM, Cullen Jennings wrote:

>
> On Dec 8, 2006, at 5:49 AM, Hisham Khartabil wrote:
>
>> Please, lets not wait until March to discuss this. We can take a  
>> hum here on the mailing list
>
> We had most of a room hum the other direction so, which did leave  
> me thinking, assuming we don't get a single no on the the list, how  
> many yes do think the chairs should be looking for to overturn  
> this? I don't really want to know the answer to this as this is  
> something that only the chairs need to figure out but it is an  
> interesting question to think about when thinking about reopening  
> this.
>

As a chair, I'd be looking for "Yes" statements from a significant  
number of the implementors and folks who supported the STUN/TCP  
position in the previous hums. Continued calls for a revote that seem  
to have support only from the minority that supported CR/LF for TCP  
the first time around represent intransigence, not a change in  
working group position.

But hey, I'll Ask . . .


> And just so people understand my position on this, if we reopen  
> this issue, I will be voting for whichever path I think can get the  
> draft to the IESG the soonest. I've implemented both and they are  
> both trivial to implement. I really believe that what we need is to  
> stop delaying this draft.

Yea, verily!

--
Dean

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From sip-bounces@ietf.org Fri Dec 08 21:12:33 2006
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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Previously we had recorded consensus to use STUN for keepalive with  
both TCP and UDP. This was not a perfect consensus (there were some  
dissenters). There has been ongoing discussion notably instigated by  
the current editor of the Outbound draft, that some people feel may  
have resulted in a change of the consensus on this draft's technical  
direction.

So I'd like to ask people who care and think they understand the  
differences to answer the following questions, on-list and on-the  
record.


Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS  
over TCP flow (retain STUN for UDP flows).

Yes (accept the proposal and use CRLF for TCP) or No (reject the  
proposal and keep STUN for TCP)


Proposal 2: Make outbound registrations over UDP flows optional for UAs.

Yes (Accept the proposal) or No (Reject the proposal).


Proposal 3: Include (in Outbound) the statement that its is  
RECOMMENDED that the outbound-proxy-set is configured to result in  
registrations over TCP or TLS whenever the UA can't accept incoming  
TCP flows. Further, include text that explain that for historical  
purposes a lot of deployments still only use UDP, but that we expect/ 
hope those deployments to eventually migrate to TCP when doing outbound.

Yes (Accept the proposal and add the text) or No (Reject the proposal  
and keep the text as-is).


--
Dean Willis (wearing his chair sombrero)

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Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
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   From: Rohan Mahy <rohan@ekabal.com>

   Well in this case, the compression they are using still makes it rare 
   to receive > 1300 byte requests over the last hop even when they get 
   requests from something in the public Internet that is substantially 
   larger. Even over TCP it is still acceptable for a proxy or UA to 
   reject a request that is too big.

I suppose it's possible, but it seems to be strongly discouraged.  RFC
3261 says that an implementation must be willing to accept 64k-1 byte
UDP requests:

   18.1.1.  However, implementations MUST be able to handle messages
   up to the maximum datagram packet size.  For UDP, this size is
   65,535 bytes, including IP and UDP headers.

Dale

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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Fri, 8 Dec 2006 21:54:00 -0800
To: Byron Campen <bcampen@estacado.net>
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Edge Proxies and Registrars would still need to support outbound with 
UDP.

thanks,
-rohan

On Dec 8, 2006, at 14:21, Byron Campen wrote:
> 	This looks sensible to me. But, I would like clarification on one 
> point; are you proposing that server implementations of outbound 
> (registrar, edge proxy) must (still) support outbound over UDP? Or 
> would it be optional for these elements as well?
>
> Best regards,
> Byron Campen
>
>
>> My concrete proposal is the following:
>> - make CRLF the only keepalive mechanism for TCP and TLS over TCP 
>> flow (retain STUN for UDP flows)
>> - make outbound registrations over UDP flows optional for UAs
>> - include the statement that its is RECOMMENDED that the 
>> outbound-proxy-set is configured to result in registrations over TCP 
>> or TLS whenever the UA can't accept incoming TCP flows (the 3GPP 
>> exception).  explain that for historical purposes a lot of 
>> deployments still only use UDP, but that we expect/hope those 
>> deployments to eventually migrate to TCP when doing outbound.
>>
>> What's in it for the TCP crowd?  They get to implement outbound with 
>> no UDP if they so desire.  They don't have to implement STUN over 
>> TCP. The draft clearly explains why TCP is the technically preferred 
>> solution.
>>
>> What's in it for the UDP crowd? They get outbound behavior (at least 
>> for requests <1300 bytes) specified in the core draft for UDP.  They 
>> can keep doing what they are doing, but with STUN keepalives if the 
>> UA and edge proxy both support it.
>>
>> What's not to like?
>> thanks,
>> -rohan
>


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Subject: Re: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
Date: Sat, 9 Dec 2006 12:01:25 +0100
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Yes on all accounts

For (3) (TCP/TLS) I think we will need to say more to support the transition 
path. Currently many UAs dont support TLS, does that mean the outbound set 
should have both SIP and SIPS URIs, and the UA gets to pick?

Jeroen van Bemmel

Dean Willis wrote:
> Previously we had recorded consensus to use STUN for keepalive with
> both TCP and UDP. This was not a perfect consensus (there were some
> dissenters). There has been ongoing discussion notably instigated by
> the current editor of the Outbound draft, that some people feel may
> have resulted in a change of the consensus on this draft's technical
> direction.
>
> So I'd like to ask people who care and think they understand the
> differences to answer the following questions, on-list and on-the
> record.
>
>
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS
> over TCP flow (retain STUN for UDP flows).
>
> Yes (accept the proposal and use CRLF for TCP) or No (reject the
> proposal and keep STUN for TCP)
>
>
> Proposal 2: Make outbound registrations over UDP flows optional for
> UAs.
> Yes (Accept the proposal) or No (Reject the proposal).
>
>
> Proposal 3: Include (in Outbound) the statement that its is
> RECOMMENDED that the outbound-proxy-set is configured to result in
> registrations over TCP or TLS whenever the UA can't accept incoming
> TCP flows. Further, include text that explain that for historical
> purposes a lot of deployments still only use UDP, but that we expect/
> hope those deployments to eventually migrate to TCP when doing
> outbound.
> Yes (Accept the proposal and add the text) or No (Reject the proposal
> and keep the text as-is). 


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From:	"warte" <RULES@anytownarizona.org>
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Subject: aktuelle Wetter Guantanamo
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src=3D"cid:000301c71cc2$36eeeed0$00000000@Niels" align=3Dbaseline=20
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<DIV><FONT face=3DArial size=3D2>Freitags, und die redwas, katrin krank =
im mit dem?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Wondered why hollywood stars always =
look, bot=20
destroy. Osten hm wieso senfi halts.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Buck snag schiurr actelion ltd =
cash.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Xine kostenlos verglichen subjektiv? =
Pommes hau=20
wieder, hin astronaut uaaahh. Katrinlg nicole padder einknfte bjoern =
melde nun=20
folgen ausdauer. Computer watch savings womens crystal bezel =
tachometer.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Wurscht welches, layout target mai dazu =
enfernt=20
wurde hng. Fllt berleben rtl abgeraucht beiden fllen.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Erlebt geht schief aufstehen donpedro, =
weise elf bleibt.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Predict rise, credit anderson toomey, =
launches,=20
institute logistics profiles. Chinahelp items relations release careers =
join.=20
Microsoft overview quick learning.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Directors message complaint jobs=20
licensee.</FONT></DIV></BODY></HTML>

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From sip-bounces@ietf.org Mon Dec 11 01:03:59 2006
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From: Jari Urpalainen <jari.urpalainen@nokia.com>
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Hi all!

I've submitted an event package i-d 
<http://www.ietf.org/internet-drafts/draft-urpalainen-sip-xcap-diff-event-00.txt> 
about syncing xcap resources with xcap-diff format. The event package 
part is pretty much similar what Jonathan had in xcap-package before it 
was "moved" into the config-framework. The biggest difference is 
probably an OMA requirement to have subscriptions to XCAP components 
also. Any comments ?
thanks, Jari

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From sip-bounces@ietf.org Mon Dec 11 03:09:42 2006
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Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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Thread-Topic: [Sip] Poll: Proposal relating to keepalive, TCP,
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Hi,

Strong yes to all those three proposals. Please see the
details of my reasons below, I believe they are important
for the your consideration.


In IETF#65 meeting I was one of those who voted for STUN
as keepalive mechanism over TCP. I did that due to consistency
reason: as STUN was selected as a keepalive mechanism instead
of SIP PING for UDP, it appeared to me that using STUN over TCP
as well would be the most straightforward way to go. However I
have changed my opinion due to the following pieces of normative
text which have been added to Outbound draft AFTER the Dallas hum:

1. Usage of UDP flows now appear as a bad idea in general,
   thus my original "consistency reason" has lost its value.
   UA could just set up TCP flows with CRLF keepalives and
   NEVER use STUN for keepalives related to Outbound:

   Use of outbound with UDP flows typically prevents messages larger
   than 1300 bytes from being delivered using TCP as described in
   Section 18.1.1 RFC 3261.  Use of outbound in conjunction with UDP
   flows in environments where requests are expected to be larger than
   this size is therefore NOT RECOMMENDED.

2. Mechanisms to for the UA to validate if the proxy supports=20
   STUN or not appear more complex than hoped. Using OPTIONS for
   the check also consumes an extra round-trip. No such validation
   is needed for CRLF keepalives, which are totally RFC3261 compatible:

   Because sending and receiving binary STUN data on the same ports used
   for SIP is a significant and non-backwards compatible change to RFC
   3261, this section requires a number of checks before sending STUN
   messages to a SIP node.  If a SIP node sends STUN requests (for
   example due to misconfiguration) despite these warnings, the node
   could be blacklisted for UDP traffic, or cause its TCP server to
   loose framing over its connection.  For each target node (as
   determined by IP address, address family, and port number), the
   sender needs to determine if that destination is validated to support
   STUN, that it does not support STUN, or that it needs to be
   validated.

   A SIP node MUST NOT send STUN requests over a flow unless it has an
   explicit indication that the target next hop SIP server claims to
   support STUN.  For example, automatic or manual configuration of an
   outbound-proxy-set which contains the keepalive=3Dstun parameter is
   considered sufficient explicit indication.  Note that UACs MUST NOT
   use an ambiguous configuration option such as "Work through NATs?" or
   "Do Keepalives?" to imply next hop STUN support.  A SIP node MAY also
   probe the next hop using a SIP OPTIONS request to check for support
   of the 'sip-stun' option tag in a Supported header field.

3. If the UA does not support TCP keepalive and the proxy appears not
   to support STUN, then no keepalive over TCP is possible ! If UA could
   simply send CRLF keepalives this would NOT be a problem:

   Futhermore, even with explicit indication of next hop STUN support, a
   SIP node needs to validate support for STUN the first time it sends
   traffic to a specific unvalidated target destination.  If an explicit
   probe indicates support for the 'sip-stun' option-tag, that
   destination is validated for STUN support. If an explicit probe does
   not indicate support for the 'sip-stun' option-tag, the target
   destination does not support STUN request, and the UAC MUST NOT send
   further STUN requests to this destination.  A SIP node MAY send one
   STUN request and its retransmissions to an unvalidated destination.
   If a STUN request ever succeeds to a destination, that destination is
   thereafter validated for STUN support.  If this initial STUN request
   does not result in a STUN response, the SIP node MUST NOT send
   additional STUN requests over this flow, unless and until a next-hop
   probe later validates the destination.  In addition, the SIP node
   SHOULD remember unvalidated destination nodes that have been used
   within one hour and SHOULD NOT send additional STUN messages to any
   of these destinations.  Note that until STUN support has been
   verified, an initial STUN failure over UDP is not considered a flow
   failure.  For UDP flows, an unvalidated flow can still be reused for
   SIP traffic, however for unvalidated TCP or TLS over TCP flows, the
   connection over which STUN requests were sent MUST be closed.

4. Interaction of SIP, STUN and SigComp over TCP clearly wastes
   bandwidth, exactly on cases where SigComp would otherwise be
   useful to save bandwidth. Instead of sending a couple of bytes
   (CRLF keepalive) each keepalive would consist of more than 30 bytes
   (13 bytes of SigComp shim header plus 20 bytes of STUN header):

   For TCP-based flows, SigComp requires that all messages are processed
   by the SigComp compressor to facilitate framing.  For these
   transports, STUN messages are sent encapsulated in the SigComp "well-
   known shim header" as described in Section 11 of [I-D.ietf-rohc-
   sigcomp-impl-guide].


For an UA that would support Outbound only over TCP using
CRLF keepalives, all that complicated processing together
with waste of round-trips and bandwidth would be easily avoided.


>Previously we had recorded consensus to use STUN for keepalive with =20
>both TCP and UDP.=20

In my understanding that concensus was reached by hums in
IETF#65 meeting in Dallas. My observation from the meeting was
that the discussion was focusing to a selection between STUN
and SIP PING as a global keepalive mechanism, which raised a
hectic email rally before the meeting. CRLF was touched but
the debate was really about the choice between STUN and PING.
There were two votes in the meeting, the first one for UDP
keepalive mechanism and the second for TCP. My feeling was that
the meeting participants focused their thoughts very much on
the STUN vs. PING issue so that STUN vs. CRLF was not
considered too carefully before the votes. And as demonstrated
above, the decision made in Dallas brought a number of
complications to Outbound in a way which could not be
anticipated yet in that meeting.

Regards,

Erkki=20

>-----Original Message-----
>From: ext Dean Willis [mailto:dean.willis@softarmor.com]=20
>Sent: 09.December.2006 04:12
>To: SIP IETF
>Subject: [Sip] Poll: Proposal relating to keepalive, TCP,and=20
>UDP usage in draft-ietf-sip-outbound
>
>Previously we had recorded consensus to use STUN for keepalive with =20
>both TCP and UDP. This was not a perfect consensus (there were some =20
>dissenters). There has been ongoing discussion notably instigated by =20
>the current editor of the Outbound draft, that some people feel may =20
>have resulted in a change of the consensus on this draft's technical =20
>direction.
>
>So I'd like to ask people who care and think they understand the =20
>differences to answer the following questions, on-list and on-the =20
>record.
>
>
>Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS =20
>over TCP flow (retain STUN for UDP flows).
>
>Yes (accept the proposal and use CRLF for TCP) or No (reject the =20
>proposal and keep STUN for TCP)
>
>
>Proposal 2: Make outbound registrations over UDP flows=20
>optional for UAs.
>
>Yes (Accept the proposal) or No (Reject the proposal).
>
>
>Proposal 3: Include (in Outbound) the statement that its is =20
>RECOMMENDED that the outbound-proxy-set is configured to result in =20
>registrations over TCP or TLS whenever the UA can't accept incoming =20
>TCP flows. Further, include text that explain that for historical =20
>purposes a lot of deployments still only use UDP, but that we expect/=20
>hope those deployments to eventually migrate to TCP when doing=20
>outbound.
>
>Yes (Accept the proposal and add the text) or No (Reject the proposal =20
>and keep the text as-is).
>
>
>--
>Dean Willis (wearing his chair sombrero)
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
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From sip-bounces@ietf.org Mon Dec 11 03:21:30 2006
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Date: Mon, 11 Dec 2006 08:20:45 +0000
From: "Elwell, John" <john.elwell@siemens.com>
Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usag	e in draft-ietf-sip-outbound
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Yes to all.

John 

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com] 
> Sent: 09 December 2006 02:12
> To: SIP IETF
> Subject: [Sip] Poll: Proposal relating to keepalive, TCP, and 
> UDP usage in draft-ietf-sip-outbound
> 
> Previously we had recorded consensus to use STUN for keepalive with  
> both TCP and UDP. This was not a perfect consensus (there were some  
> dissenters). There has been ongoing discussion notably instigated by  
> the current editor of the Outbound draft, that some people feel may  
> have resulted in a change of the consensus on this draft's technical  
> direction.
> 
> So I'd like to ask people who care and think they understand the  
> differences to answer the following questions, on-list and on-the  
> record.
> 
> 
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS  
> over TCP flow (retain STUN for UDP flows).
> 
> Yes (accept the proposal and use CRLF for TCP) or No (reject the  
> proposal and keep STUN for TCP)
> 
> 
> Proposal 2: Make outbound registrations over UDP flows 
> optional for UAs.
> 
> Yes (Accept the proposal) or No (Reject the proposal).
> 
> 
> Proposal 3: Include (in Outbound) the statement that its is  
> RECOMMENDED that the outbound-proxy-set is configured to result in  
> registrations over TCP or TLS whenever the UA can't accept incoming  
> TCP flows. Further, include text that explain that for historical  
> purposes a lot of deployments still only use UDP, but that we expect/ 
> hope those deployments to eventually migrate to TCP when 
> doing outbound.
> 
> Yes (Accept the proposal and add the text) or No (Reject the 
> proposal  
> and keep the text as-is).
> 
> 
> --
> Dean Willis (wearing his chair sombrero)
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Mon Dec 11 03:30:35 2006
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Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
Date: Mon, 11 Dec 2006 09:29:52 +0100
Message-ID: <7374777208BDC7449D5620EF94232567E13F70@esealmw113.eemea.ericsson.se>
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Thread-Index: AccbN5coW0aYGBPqQBSD++qTR1PRdgBS1gVA
From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Dean Willis" <dean.willis@softarmor.com>,
	"SIP IETF" <sip@ietf.org>
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Hi,

There are a few issues I'd like to clarify on proposal 3.

First, I think the "default" recommendation should only cover TCP. The
text can say that, if the terminal supports TLS, the recommendation also
covers TLS. However, in some environments other security mechanisms may
be used, so I don't think clients in such environments should even be
recommended to implement TLS, nor should the whatever-outbound-proxy-set
configuration-mechanism is used in those environments be recommended to
return TLS registrations in such environements.

Second, as I said before, I don't think we shall talk about UDP and its
"history". UDP is a valid transport for SIP. IF someone really wants
this history text I propose we separate it into proposal 4.

Third, since it is only recommended to return TCP information, I guess
it means we will NOT mandate (eventhough we would, if propsoal 3 is
accepted, recommend it) a UA to establish TCP registrations? If so, I
think we could clarify that (either as part of option 2 or 3).=20

Regards,

Christer


=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 9. joulukuuta 2006 4:12
> To: SIP IETF
> Subject: [Sip] Poll: Proposal relating to keepalive, TCP,and=20
> UDP usage in draft-ietf-sip-outbound
>=20
> Previously we had recorded consensus to use STUN for=20
> keepalive with both TCP and UDP. This was not a perfect=20
> consensus (there were some dissenters). There has been=20
> ongoing discussion notably instigated by the current editor=20
> of the Outbound draft, that some people feel may have=20
> resulted in a change of the consensus on this draft's=20
> technical direction.
>=20
> So I'd like to ask people who care and think they understand=20
> the differences to answer the following questions, on-list=20
> and on-the record.
>=20
>=20
> Proposal 1: Make CRLF the only keepalive mechanism for TCP=20
> and TLS over TCP flow (retain STUN for UDP flows).
>=20
> Yes (accept the proposal and use CRLF for TCP) or No (reject=20
> the proposal and keep STUN for TCP)
>=20
>=20
> Proposal 2: Make outbound registrations over UDP flows=20
> optional for UAs.
>=20
> Yes (Accept the proposal) or No (Reject the proposal).
>=20
>=20
> Proposal 3: Include (in Outbound) the statement that its is=20
> RECOMMENDED that the outbound-proxy-set is configured to=20
> result in registrations over TCP or TLS whenever the UA can't=20
> accept incoming TCP flows. Further, include text that explain=20
> that for historical purposes a lot of deployments still only=20
> use UDP, but that we expect/ hope those deployments to=20
> eventually migrate to TCP when doing outbound.
>=20
> Yes (Accept the proposal and add the text) or No (Reject the=20
> proposal and keep the text as-is).
>=20
>=20
> --
> Dean Willis (wearing his chair sombrero)
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

From sip-bounces@ietf.org Mon Dec 11 03:30:35 2006
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From: Fredrik Thulin <ft@it.su.se>
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To: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
References: <B72D5D3A4BF30243B53DEEF2961613B0026EE4D9@esebe106.NOE.Nokia.com><f42ad5b1999ba54be6657b9cf36a9449@telio.no>	<4575D55B.8040200@it.su.se>	<00b801c718b6$84185490$0601a8c0@BEMBUSTER>	<3B1E01D3-F436-48D5-B625-864C7F9F54BA@cisco.com>	<ce53650ebacf8e3c6621b8136e01dd89@ekabal.com>	<8F8614DA-EA7F-407B-A5F0-F6578E7CB5E2@estacado.net>
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Rohan Mahy wrote:
> Edge Proxies and Registrars would still need to support outbound with UDP.

Hi

Before answering Dean's poll, I would like to know more about this.

Have you figured out a (reasonable) way for the edge proxy/authoritative 
proxy (or the combination thereof) to switch from UDP to TCP for large 
messages destined for a UAC that uses Outbound?

I'm thinking of the problems in doing that that we discussed briefly in 
these two messages :

   http://www1.ietf.org/mail-archive/web/sip/current/msg17418.html
   http://www1.ietf.org/mail-archive/web/sip/current/msg17420.html

Thanks

/Fredrik

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





From sip-bounces@ietf.org Mon Dec 11 03:30:35 2006
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Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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Hi,

There are a few issues I'd like to clarify on proposal 3.

First, I think the "default" recommendation should only cover TCP. The
text can say that, if the terminal supports TLS, the recommendation also
covers TLS. However, in some environments other security mechanisms may
be used, so I don't think clients in such environments should even be
recommended to implement TLS, nor should the whatever-outbound-proxy-set
configuration-mechanism is used in those environments be recommended to
return TLS registrations in such environements.

Second, as I said before, I don't think we shall talk about UDP and its
"history". UDP is a valid transport for SIP. IF someone really wants
this history text I propose we separate it into proposal 4.

Third, since it is only recommended to return TCP information, I guess
it means we will NOT mandate (eventhough we would, if propsoal 3 is
accepted, recommend it) a UA to establish TCP registrations? If so, I
think we could clarify that (either as part of option 2 or 3).=20

Regards,

Christer


=20

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: 9. joulukuuta 2006 4:12
> To: SIP IETF
> Subject: [Sip] Poll: Proposal relating to keepalive, TCP,and=20
> UDP usage in draft-ietf-sip-outbound
>=20
> Previously we had recorded consensus to use STUN for=20
> keepalive with both TCP and UDP. This was not a perfect=20
> consensus (there were some dissenters). There has been=20
> ongoing discussion notably instigated by the current editor=20
> of the Outbound draft, that some people feel may have=20
> resulted in a change of the consensus on this draft's=20
> technical direction.
>=20
> So I'd like to ask people who care and think they understand=20
> the differences to answer the following questions, on-list=20
> and on-the record.
>=20
>=20
> Proposal 1: Make CRLF the only keepalive mechanism for TCP=20
> and TLS over TCP flow (retain STUN for UDP flows).
>=20
> Yes (accept the proposal and use CRLF for TCP) or No (reject=20
> the proposal and keep STUN for TCP)
>=20
>=20
> Proposal 2: Make outbound registrations over UDP flows=20
> optional for UAs.
>=20
> Yes (Accept the proposal) or No (Reject the proposal).
>=20
>=20
> Proposal 3: Include (in Outbound) the statement that its is=20
> RECOMMENDED that the outbound-proxy-set is configured to=20
> result in registrations over TCP or TLS whenever the UA can't=20
> accept incoming TCP flows. Further, include text that explain=20
> that for historical purposes a lot of deployments still only=20
> use UDP, but that we expect/ hope those deployments to=20
> eventually migrate to TCP when doing outbound.
>=20
> Yes (Accept the proposal and add the text) or No (Reject the=20
> proposal and keep the text as-is).
>=20
>=20
> --
> Dean Willis (wearing his chair sombrero)
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip

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To: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
References: <B72D5D3A4BF30243B53DEEF2961613B0026EE4D9@esebe106.NOE.Nokia.com><f42ad5b1999ba54be6657b9cf36a9449@telio.no>	<4575D55B.8040200@it.su.se>	<00b801c718b6$84185490$0601a8c0@BEMBUSTER>	<3B1E01D3-F436-48D5-B625-864C7F9F54BA@cisco.com>	<ce53650ebacf8e3c6621b8136e01dd89@ekabal.com>	<8F8614DA-EA7F-407B-A5F0-F6578E7CB5E2@estacado.net>
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Rohan Mahy wrote:
> Edge Proxies and Registrars would still need to support outbound with UDP.

Hi

Before answering Dean's poll, I would like to know more about this.

Have you figured out a (reasonable) way for the edge proxy/authoritative 
proxy (or the combination thereof) to switch from UDP to TCP for large 
messages destined for a UAC that uses Outbound?

I'm thinking of the problems in doing that that we discussed briefly in 
these two messages :

   http://www1.ietf.org/mail-archive/web/sip/current/msg17418.html
   http://www1.ietf.org/mail-archive/web/sip/current/msg17420.html

Thanks

/Fredrik

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip





From sip-bounces@ietf.org Mon Dec 11 03:30:40 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Mon, 11 Dec 2006 09:29:56 +0100
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Thread-Topic: [Sip] Re: issue with oubound and the used transport protocol
Thread-Index: Acca8JHlmXPTIQRjQnm9Xf7SWjUsogANxesgAFeGaFA=
From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: "Francois Audet" <audet@nortel.com>, "Cullen Jennings" <fluffy@cisco.com>,
	"Hisham Khartabil" <hisham.khartabil@telio.no>
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	SIP <sip@ietf.org>,
	"Isomaki Markus \(Nokia-TP/Espoo\)" <Markus.Isomaki@nokia.com>,
	Peter Leis <peter.leis@siemens.com>, Aki Niemi <aki.niemi@nokia.com>,
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=20

>My best guest on why this is happening is that people used to=20
>be thinking more about UDP than TCP and consequently=20
>expressed a preference for the STUN approach.

I think people thought more about STUN than either UDP or TCP ;)

Regards,

Christer




> > -----Original Message-----
> > From: Cullen Jennings [mailto:fluffy@cisco.com]
> > Sent: Friday, December 08, 2006 09:43
> > To: Hisham Khartabil
> > Cc: Rohan Mahy; Fredrik Thulin; SIP; Isomaki Markus=20
> (Nokia-TP/Espoo);=20
> > Peter Leis; Aki Niemi; Dean Willis; Keith Drage
> > Subject: Re: [Sip] Re: issue with oubound and the used transport=20
> > protocol
> >=20
> >=20
> > On Dec 8, 2006, at 5:49 AM, Hisham Khartabil wrote:
> >=20
> > > Please, lets not wait until March to discuss this. We can
> > take a hum
> > > here on the mailing list
> >=20
> > We had most of a room hum the other direction so, which did=20
> leave me=20
> > thinking, assuming we don't get a single no on the the=20
> list, how many=20
> > yes do think the chairs should be looking for to overturn this? I=20
> > don't really want to know the answer to this as this is=20
> something that=20
> > only the chairs need to figure out but it is an interesting=20
> question=20
> > to think about when thinking about reopening this.
> >=20
> > And just so people understand my position on this, if we=20
> reopen this=20
> > issue, I will be voting for whichever path I think can get=20
> the draft=20
> > to the IESG the soonest. I've implemented both and they are both=20
> > trivial to implement. I really believe that what we need is to stop=20
> > delaying this draft.
> >=20
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol Use=20
> > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > sipping@ietf.org for new developments on the application of sip
> >=20
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Mon Dec 11 03:50:24 2006
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From: "Aki Niemi" <aki.niemi@nokia.com>
To: "'ext Dean Willis'" <dean.willis@softarmor.com>,
	"'SIP IETF'" <sip@ietf.org>
Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
Date: Mon, 11 Dec 2006 10:49:30 +0200
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Yes, yes, and yes.

Cheers,
Aki 

>-----Original Message-----
>From: ext Dean Willis [mailto:dean.willis@softarmor.com] 
>Sent: 09 December, 2006 04:12
>To: SIP IETF
>Subject: [Sip] Poll: Proposal relating to keepalive, TCP,and 
>UDP usage in draft-ietf-sip-outbound
>
>Previously we had recorded consensus to use STUN for keepalive 
>with both TCP and UDP. This was not a perfect consensus (there 
>were some dissenters). There has been ongoing discussion 
>notably instigated by the current editor of the Outbound 
>draft, that some people feel may have resulted in a change of 
>the consensus on this draft's technical direction.
>
>So I'd like to ask people who care and think they understand 
>the differences to answer the following questions, on-list and 
>on-the record.
>
>
>Proposal 1: Make CRLF the only keepalive mechanism for TCP and 
>TLS over TCP flow (retain STUN for UDP flows).
>
>Yes (accept the proposal and use CRLF for TCP) or No (reject 
>the proposal and keep STUN for TCP)
>
>
>Proposal 2: Make outbound registrations over UDP flows 
>optional for UAs.
>
>Yes (Accept the proposal) or No (Reject the proposal).
>
>
>Proposal 3: Include (in Outbound) the statement that its is 
>RECOMMENDED that the outbound-proxy-set is configured to 
>result in registrations over TCP or TLS whenever the UA can't 
>accept incoming TCP flows. Further, include text that explain 
>that for historical purposes a lot of deployments still only 
>use UDP, but that we expect/ hope those deployments to 
>eventually migrate to TCP when doing outbound.
>
>Yes (Accept the proposal and add the text) or No (Reject the 
>proposal and keep the text as-is).
>
>
>--
>Dean Willis (wearing his chair sombrero)
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use 
>sip-implementors@cs.columbia.edu for questions on current sip 
>Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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From sip-bounces@ietf.org Mon Dec 11 04:01:06 2006
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Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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From: <Markus.Isomaki@nokia.com>
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Hi,

Yes to all proposals, especially the first one.

Markus

>-----Original Message-----
>From: ext Dean Willis [mailto:dean.willis@softarmor.com]=20
>Sent: 09 December, 2006 04:12
>To: SIP IETF
>Subject: [Sip] Poll: Proposal relating to keepalive, TCP,and=20
>UDP usage in draft-ietf-sip-outbound
>
>Previously we had recorded consensus to use STUN for keepalive=20
>with both TCP and UDP. This was not a perfect consensus (there=20
>were some dissenters). There has been ongoing discussion=20
>notably instigated by the current editor of the Outbound=20
>draft, that some people feel may have resulted in a change of=20
>the consensus on this draft's technical direction.
>
>So I'd like to ask people who care and think they understand=20
>the differences to answer the following questions, on-list and=20
>on-the record.
>
>
>Proposal 1: Make CRLF the only keepalive mechanism for TCP and=20
>TLS over TCP flow (retain STUN for UDP flows).
>
>Yes (accept the proposal and use CRLF for TCP) or No (reject=20
>the proposal and keep STUN for TCP)
>
>
>Proposal 2: Make outbound registrations over UDP flows=20
>optional for UAs.
>
>Yes (Accept the proposal) or No (Reject the proposal).
>
>
>Proposal 3: Include (in Outbound) the statement that its is=20
>RECOMMENDED that the outbound-proxy-set is configured to=20
>result in registrations over TCP or TLS whenever the UA can't=20
>accept incoming TCP flows. Further, include text that explain=20
>that for historical purposes a lot of deployments still only=20
>use UDP, but that we expect/ hope those deployments to=20
>eventually migrate to TCP when doing outbound.
>
>Yes (Accept the proposal and add the text) or No (Reject the=20
>proposal and keep the text as-is).
>
>
>--
>Dean Willis (wearing his chair sombrero)
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol Use=20
>sip-implementors@cs.columbia.edu for questions on current sip=20
>Use sipping@ietf.org for new developments on the application of sip
>

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Mon Dec 11 04:25:37 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Mon, 11 Dec 2006 11:23:45 +0200
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From: <Markus.Isomaki@nokia.com>
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Hi,

I agree with Francois here. Even though there may be some performance
limitations for TCP in the current implementations, I think that almost
everyone agrees that it is the way to go in the long run. This is why it
would be good to have an optimal solution for Outbound/TCP, instead of
trying to keep consistency between UDP and TCP.

It would be great if we could do UAs supporting Outbound with TCP/TLS
only, and leave mess with message size, fragment handling, UDP-based
security, 20 second NAT keepalive timers etc. behind us. This may not be
realistic yet, if there are enough service providers who will demand UDP
(since unfortunately there are e.g. SBCs that do not even have a proper
TCP support at all), but that should be the aim.  =20

3GPP IMS networks, if such are deployed in the "puristic" form, may be a
different thing, but even there TCP is a better option.=20

Markus


>-----Original Message-----
>From: ext Francois Audet [mailto:audet@nortel.com]=20
>Sent: 09 December, 2006 02:31
>To: Cullen Jennings; Hisham Khartabil
>Cc: Rohan Mahy; Fredrik Thulin; SIP; Isomaki Markus=20
>(Nokia-SIR/Espoo); Peter Leis; Niemi Aki (Nokia-NRC/Helsinki);=20
>Dean Willis; Keith Drage
>Subject: RE: [Sip] Re: issue with oubound and the used=20
>transport protocol
>
>Cullen,
>
>My best guest on why this is happening is that people used to=20
>be thinking more about UDP than TCP and consequently expressed=20
>a preference for the STUN approach.
>
>It seems now that more people are paying attention to TCP=20
>because of a bunch of issues with UDP (including=20
>fragmentation, and TLS usage catching up since it took us so=20
>long to finalize this document). Many are reaching the=20
>conclusion that the CRLF approach makes more sense for TCP.
>
>It seems that Rohan's last proposal of using CRLF for TCP, and=20
>STUN for UDP to be reasonable and logical. And it seems to=20
>have lots of support on the list.
>
>
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Friday, December 08, 2006 09:43
>> To: Hisham Khartabil
>> Cc: Rohan Mahy; Fredrik Thulin; SIP; Isomaki Markus=20
>(Nokia-TP/Espoo);=20
>> Peter Leis; Aki Niemi; Dean Willis; Keith Drage
>> Subject: Re: [Sip] Re: issue with oubound and the used transport=20
>> protocol
>>=20
>>=20
>> On Dec 8, 2006, at 5:49 AM, Hisham Khartabil wrote:
>>=20
>> > Please, lets not wait until March to discuss this. We can
>> take a hum
>> > here on the mailing list
>>=20
>> We had most of a room hum the other direction so, which did leave me=20
>> thinking, assuming we don't get a single no on the the list,=20
>how many=20
>> yes do think the chairs should be looking for to overturn this? I=20
>> don't really want to know the answer to this as this is=20
>something that=20
>> only the chairs need to figure out but it is an interesting question=20
>> to think about when thinking about reopening this.
>>=20
>> And just so people understand my position on this, if we reopen this=20
>> issue, I will be voting for whichever path I think can get the draft=20
>> to the IESG the soonest. I've implemented both and they are both=20
>> trivial to implement. I really believe that what we need is to stop=20
>> delaying this draft.
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use=20
>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>> sipping@ietf.org for new developments on the application of sip
>>=20
>

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Mon Dec 11 05:20:25 2006
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From: "Alfred E. Heggestad" <aeh@db.org>
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Subject: Re: [Sip] Poll: Proposal relating to keepalive, TCP,	and UDP usage
	in draft-ietf-sip-outbound
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Dean Willis wrote:
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS over 
> TCP flow (retain STUN for UDP flows).
> 
> Yes (accept the proposal and use CRLF for TCP) or No (reject the 
> proposal and keep STUN for TCP)
> 

YES.


> 
> Proposal 2: Make outbound registrations over UDP flows optional for UAs.
> 
> Yes (Accept the proposal) or No (Reject the proposal).
> 

YES.


> 
> Proposal 3: Include (in Outbound) the statement that its is RECOMMENDED 
> that the outbound-proxy-set is configured to result in registrations 
> over TCP or TLS whenever the UA can't accept incoming TCP flows. 
> Further, include text that explain that for historical purposes a lot of 
> deployments still only use UDP, but that we expect/hope those 
> deployments to eventually migrate to TCP when doing outbound.
> 
> Yes (Accept the proposal and add the text) or No (Reject the proposal 
> and keep the text as-is).
> 

YES.



/alfred

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From sip-bounces@ietf.org Mon Dec 11 05:35:49 2006
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Subject: RE: [Sip] Re: issue with oubound and the used transport protocol
Date: Mon, 11 Dec 2006 11:34:25 +0100
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From: "STURA, Marco, VF-IT" <Marco.STURA@vodafone.com>
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> 3GPP IMS networks, if such are deployed in the "puristic" form, may be =
a different=20
>thing, but even there TCP is a better option.=20

I don't think such "puristic" deployments exist to date. In general I do =
agree with Markus that TCP would be a better option in 3GPP IMS networks =
too.

Br,
Marco

-----Original Message-----
From: Markus.Isomaki@nokia.com [mailto:Markus.Isomaki@nokia.com]=20
Sent: luned=EC 11 dicembre 2006 10.24
To: audet@nortel.com; fluffy@cisco.com; hisham.khartabil@telio.no
Cc: rohan@ekabal.com; ft@it.su.se; sip@ietf.org; peter.leis@siemens.com; =
aki.niemi@nokia.com; dean.willis@softarmor.com; drage@lucent.com
Subject: RE: [Sip] Re: issue with oubound and the used transport =
protocol

Hi,

I agree with Francois here. Even though there may be some performance =
limitations for TCP in the current implementations, I think that almost =
everyone agrees that it is the way to go in the long run. This is why it =
would be good to have an optimal solution for Outbound/TCP, instead of =
trying to keep consistency between UDP and TCP.

It would be great if we could do UAs supporting Outbound with TCP/TLS =
only, and leave mess with message size, fragment handling, UDP-based =
security, 20 second NAT keepalive timers etc. behind us. This may not be =
realistic yet, if there are enough service providers who will demand UDP =
(since unfortunately there are e.g. SBCs that do not even have a proper
TCP support at all), but that should be the aim.  =20

3GPP IMS networks, if such are deployed in the "puristic" form, may be a =
different thing, but even there TCP is a better option.=20

Markus


>-----Original Message-----
>From: ext Francois Audet [mailto:audet@nortel.com]
>Sent: 09 December, 2006 02:31
>To: Cullen Jennings; Hisham Khartabil
>Cc: Rohan Mahy; Fredrik Thulin; SIP; Isomaki Markus (Nokia-SIR/Espoo);=20
>Peter Leis; Niemi Aki (Nokia-NRC/Helsinki); Dean Willis; Keith Drage
>Subject: RE: [Sip] Re: issue with oubound and the used transport=20
>protocol
>
>Cullen,
>
>My best guest on why this is happening is that people used to be=20
>thinking more about UDP than TCP and consequently expressed a=20
>preference for the STUN approach.
>
>It seems now that more people are paying attention to TCP because of a=20
>bunch of issues with UDP (including fragmentation, and TLS usage=20
>catching up since it took us so long to finalize this document). Many=20
>are reaching the conclusion that the CRLF approach makes more sense for =

>TCP.
>
>It seems that Rohan's last proposal of using CRLF for TCP, and STUN for =

>UDP to be reasonable and logical. And it seems to have lots of support=20
>on the list.
>
>
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Friday, December 08, 2006 09:43
>> To: Hisham Khartabil
>> Cc: Rohan Mahy; Fredrik Thulin; SIP; Isomaki Markus
>(Nokia-TP/Espoo);
>> Peter Leis; Aki Niemi; Dean Willis; Keith Drage
>> Subject: Re: [Sip] Re: issue with oubound and the used transport=20
>> protocol
>>=20
>>=20
>> On Dec 8, 2006, at 5:49 AM, Hisham Khartabil wrote:
>>=20
>> > Please, lets not wait until March to discuss this. We can
>> take a hum
>> > here on the mailing list
>>=20
>> We had most of a room hum the other direction so, which did leave me=20
>> thinking, assuming we don't get a single no on the the list,
>how many
>> yes do think the chairs should be looking for to overturn this? I=20
>> don't really want to know the answer to this as this is
>something that
>> only the chairs need to figure out but it is an interesting question=20
>> to think about when thinking about reopening this.
>>=20
>> And just so people understand my position on this, if we reopen this=20
>> issue, I will be voting for whichever path I think can get the draft=20
>> to the IESG the soonest. I've implemented both and they are both=20
>> trivial to implement. I really believe that what we need is to stop=20
>> delaying this draft.
>>=20
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol Use=20
>> sip-implementors@cs.columbia.edu for questions on current sip Use=20
>> sipping@ietf.org for new developments on the application of sip
>>=20
>

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use =
sip-implementors@cs.columbia.edu for questions on current sip Use =
sipping@ietf.org for new developments on the application of sip


_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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<DIV><FONT face=3DArial size=3D2>Both networks it, also contains =
slew!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Hardware internet graphics offtopic =
lounge?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Replies edit, vb code? Money ebooks, =
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click.</FONT></DIV>
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From sip-bounces@ietf.org Mon Dec 11 06:22:18 2006
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Subject: Re: [Sip] Poll: Proposal relating to keepalive, TCP,	and UDP usage
	in draft-ietf-sip-outbound
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Yes to all.

Dean Willis wrote:
> Previously we had recorded consensus to use STUN for keepalive with both 
> TCP and UDP. This was not a perfect consensus (there were some 
> dissenters). There has been ongoing discussion notably instigated by the 
> current editor of the Outbound draft, that some people feel may have 
> resulted in a change of the consensus on this draft's technical direction.
> 
> So I'd like to ask people who care and think they understand the 
> differences to answer the following questions, on-list and on-the record.
> 
> 
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS over 
> TCP flow (retain STUN for UDP flows).
> 
> Yes (accept the proposal and use CRLF for TCP) or No (reject the 
> proposal and keep STUN for TCP)
> 
> 
> Proposal 2: Make outbound registrations over UDP flows optional for UAs.
> 
> Yes (Accept the proposal) or No (Reject the proposal).
> 
> 
> Proposal 3: Include (in Outbound) the statement that its is RECOMMENDED 
> that the outbound-proxy-set is configured to result in registrations 
> over TCP or TLS whenever the UA can't accept incoming TCP flows. 
> Further, include text that explain that for historical purposes a lot of 
> deployments still only use UDP, but that we expect/hope those 
> deployments to eventually migrate to TCP when doing outbound.
> 
> Yes (Accept the proposal and add the text) or No (Reject the proposal 
> and keep the text as-is).
> 
> 
> -- 
> Dean Willis (wearing his chair sombrero)
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
sip:miguel.garcia@neonsite.net
Nokia Research Center      Helsinki, Finland

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Mon Dec 11 08:56:20 2006
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
Date: Mon, 11 Dec 2006 14:53:54 +0100
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On Dec 9, 2006, at 3:12 AM, Dean Willis wrote:

> Previously we had recorded consensus to use STUN for keepalive with 
> both TCP and UDP. This was not a perfect consensus (there were some 
> dissenters). There has been ongoing discussion notably instigated by 
> the current editor of the Outbound draft, that some people feel may 
> have resulted in a change of the consensus on this draft's technical 
> direction.
>
> So I'd like to ask people who care and think they understand the 
> differences to answer the following questions, on-list and on-the 
> record.
>
>
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS 
> over TCP flow (retain STUN for UDP flows).
>
> Yes (accept the proposal and use CRLF for TCP) or No (reject the 
> proposal and keep STUN for TCP)
>

yes

>
> Proposal 2: Make outbound registrations over UDP flows optional for 
> UAs.
>
> Yes (Accept the proposal) or No (Reject the proposal).
>

yes

>
> Proposal 3: Include (in Outbound) the statement that its is 
> RECOMMENDED that the outbound-proxy-set is configured to result in 
> registrations over TCP or TLS whenever the UA can't accept incoming 
> TCP flows. Further, include text that explain that for historical 
> purposes a lot of deployments still only use UDP, but that we 
> expect/hope those deployments to eventually migrate to TCP when doing 
> outbound.
>
> Yes (Accept the proposal and add the text) or No (Reject the proposal 
> and keep the text as-is).
>

yes

Hisham

>
> --
> Dean Willis (wearing his chair sombrero)
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>


_______________________________________________
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From sip-bounces@ietf.org Mon Dec 11 08:56:20 2006
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From: Raj <ponrajit@yahoo.com>
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Hello To all,
   
                  Recently I read an article and I came across the following line, 
   
  "Software or hardware appliance containing one or more SIP user agents".
   
               My doubt is considering a phone equipment, can it contain two SIP user agents in it? if yes, can any one tell me about the operation of both the user agents?
   
  with regards
  Raj,

 
---------------------------------
Cheap Talk? Check out Yahoo! Messenger's low PC-to-Phone call rates.
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<div>Hello To all,</div>  <div>&nbsp;</div>  <div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Recently I read an article and I came across the following line, </div>  <div>&nbsp;</div>  <div>"<STRONG>Software or hardware appliance containing one or more SIP user agents".</STRONG></div>  <div>&nbsp;</div>  <div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; My doubt is considering a phone equipment, can it contain two SIP user agents in it? if yes, can any one tell me about the operation of both the user agents?</div>  <div>&nbsp;</div>  <div>with regards</div>  <div>Raj,</div><p>&#32;
<hr size=1>Cheap Talk? <a href="http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com">Check out</a> Yahoo! Messenger's low PC-to-Phone call rates.
--0-454612841-1165845343=:57947--


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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
Date: Mon, 11 Dec 2006 15:50:32 +0100
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
To: <Erkki.Koivusalo@nokia.com>, <dean.willis@softarmor.com>, <sip@ietf.org>
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Hi Erkki,

When you talk about OPTIONS, extra roundtrips etc, you forget that the
edge proxy can insert the keep-alive parameter e.g. in the Path header
(or Service-Route, depending on what happens to Jonathan's route
construct draft) of the REGISTER to indicate its support of STUN.

Regards,

Christer=20

> -----Original Message-----
> From: Erkki.Koivusalo@nokia.com [mailto:Erkki.Koivusalo@nokia.com]=20
> Sent: 11. joulukuuta 2006 10:07
> To: dean.willis@softarmor.com; sip@ietf.org
> Subject: RE: [Sip] Poll: Proposal relating to keepalive,=20
> TCP,and UDP usage in draft-ietf-sip-outbound
>=20
>=20
> Hi,
>=20
> Strong yes to all those three proposals. Please see the=20
> details of my reasons below, I believe they are important for=20
> the your consideration.
>=20
>=20
> In IETF#65 meeting I was one of those who voted for STUN as=20
> keepalive mechanism over TCP. I did that due to consistency
> reason: as STUN was selected as a keepalive mechanism instead=20
> of SIP PING for UDP, it appeared to me that using STUN over=20
> TCP as well would be the most straightforward way to go.=20
> However I have changed my opinion due to the following pieces=20
> of normative text which have been added to Outbound draft=20
> AFTER the Dallas hum:
>=20
> 1. Usage of UDP flows now appear as a bad idea in general,
>    thus my original "consistency reason" has lost its value.
>    UA could just set up TCP flows with CRLF keepalives and
>    NEVER use STUN for keepalives related to Outbound:
>=20
>    Use of outbound with UDP flows typically prevents messages larger
>    than 1300 bytes from being delivered using TCP as described in
>    Section 18.1.1 RFC 3261.  Use of outbound in conjunction with UDP
>    flows in environments where requests are expected to be larger than
>    this size is therefore NOT RECOMMENDED.
>=20
> 2. Mechanisms to for the UA to validate if the proxy supports=20
>    STUN or not appear more complex than hoped. Using OPTIONS for
>    the check also consumes an extra round-trip. No such validation
>    is needed for CRLF keepalives, which are totally RFC3261=20
> compatible:
>=20
>    Because sending and receiving binary STUN data on the same=20
> ports used
>    for SIP is a significant and non-backwards compatible change to RFC
>    3261, this section requires a number of checks before sending STUN
>    messages to a SIP node.  If a SIP node sends STUN requests (for
>    example due to misconfiguration) despite these warnings, the node
>    could be blacklisted for UDP traffic, or cause its TCP server to
>    loose framing over its connection.  For each target node (as
>    determined by IP address, address family, and port number), the
>    sender needs to determine if that destination is validated=20
> to support
>    STUN, that it does not support STUN, or that it needs to be
>    validated.
>=20
>    A SIP node MUST NOT send STUN requests over a flow unless it has an
>    explicit indication that the target next hop SIP server claims to
>    support STUN.  For example, automatic or manual configuration of an
>    outbound-proxy-set which contains the keepalive=3Dstun parameter is
>    considered sufficient explicit indication.  Note that UACs MUST NOT
>    use an ambiguous configuration option such as "Work=20
> through NATs?" or
>    "Do Keepalives?" to imply next hop STUN support.  A SIP=20
> node MAY also
>    probe the next hop using a SIP OPTIONS request to check for support
>    of the 'sip-stun' option tag in a Supported header field.
>=20
> 3. If the UA does not support TCP keepalive and the proxy appears not
>    to support STUN, then no keepalive over TCP is possible !=20
> If UA could
>    simply send CRLF keepalives this would NOT be a problem:
>=20
>    Futhermore, even with explicit indication of next hop STUN=20
> support, a
>    SIP node needs to validate support for STUN the first time it sends
>    traffic to a specific unvalidated target destination.  If=20
> an explicit
>    probe indicates support for the 'sip-stun' option-tag, that
>    destination is validated for STUN support. If an explicit=20
> probe does
>    not indicate support for the 'sip-stun' option-tag, the target
>    destination does not support STUN request, and the UAC=20
> MUST NOT send
>    further STUN requests to this destination.  A SIP node MAY send one
>    STUN request and its retransmissions to an unvalidated destination.
>    If a STUN request ever succeeds to a destination, that=20
> destination is
>    thereafter validated for STUN support.  If this initial=20
> STUN request
>    does not result in a STUN response, the SIP node MUST NOT send
>    additional STUN requests over this flow, unless and until=20
> a next-hop
>    probe later validates the destination.  In addition, the SIP node
>    SHOULD remember unvalidated destination nodes that have been used
>    within one hour and SHOULD NOT send additional STUN messages to any
>    of these destinations.  Note that until STUN support has been
>    verified, an initial STUN failure over UDP is not considered a flow
>    failure.  For UDP flows, an unvalidated flow can still be=20
> reused for
>    SIP traffic, however for unvalidated TCP or TLS over TCP flows, the
>    connection over which STUN requests were sent MUST be closed.
>=20
> 4. Interaction of SIP, STUN and SigComp over TCP clearly wastes
>    bandwidth, exactly on cases where SigComp would otherwise be
>    useful to save bandwidth. Instead of sending a couple of bytes
>    (CRLF keepalive) each keepalive would consist of more than 30 bytes
>    (13 bytes of SigComp shim header plus 20 bytes of STUN header):
>=20
>    For TCP-based flows, SigComp requires that all messages=20
> are processed
>    by the SigComp compressor to facilitate framing.  For these
>    transports, STUN messages are sent encapsulated in the=20
> SigComp "well-
>    known shim header" as described in Section 11 of [I-D.ietf-rohc-
>    sigcomp-impl-guide].
>=20
>=20
> For an UA that would support Outbound only over TCP using=20
> CRLF keepalives, all that complicated processing together=20
> with waste of round-trips and bandwidth would be easily avoided.
>=20
>=20
> >Previously we had recorded consensus to use STUN for keepalive with=20
> >both TCP and UDP.
>=20
> In my understanding that concensus was reached by hums in
> IETF#65 meeting in Dallas. My observation from the meeting=20
> was that the discussion was focusing to a selection between=20
> STUN and SIP PING as a global keepalive mechanism, which=20
> raised a hectic email rally before the meeting. CRLF was=20
> touched but the debate was really about the choice between=20
> STUN and PING.
> There were two votes in the meeting, the first one for UDP=20
> keepalive mechanism and the second for TCP. My feeling was=20
> that the meeting participants focused their thoughts very=20
> much on the STUN vs. PING issue so that STUN vs. CRLF was not=20
> considered too carefully before the votes. And as=20
> demonstrated above, the decision made in Dallas brought a=20
> number of complications to Outbound in a way which could not=20
> be anticipated yet in that meeting.
>=20
> Regards,
>=20
> Erkki=20
>=20
> >-----Original Message-----
> >From: ext Dean Willis [mailto:dean.willis@softarmor.com]
> >Sent: 09.December.2006 04:12
> >To: SIP IETF
> >Subject: [Sip] Poll: Proposal relating to keepalive, TCP,and=20
> UDP usage=20
> >in draft-ietf-sip-outbound
> >
> >Previously we had recorded consensus to use STUN for keepalive with=20
> >both TCP and UDP. This was not a perfect consensus (there were some=20
> >dissenters). There has been ongoing discussion notably instigated by=20
> >the current editor of the Outbound draft, that some people feel may=20
> >have resulted in a change of the consensus on this draft's technical=20
> >direction.
> >
> >So I'd like to ask people who care and think they understand the=20
> >differences to answer the following questions, on-list and on-the=20
> >record.
> >
> >
> >Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS =20
> >over TCP flow (retain STUN for UDP flows).
> >
> >Yes (accept the proposal and use CRLF for TCP) or No (reject the =20
> >proposal and keep STUN for TCP)
> >
> >
> >Proposal 2: Make outbound registrations over UDP flows=20
> >optional for UAs.
> >
> >Yes (Accept the proposal) or No (Reject the proposal).
> >
> >
> >Proposal 3: Include (in Outbound) the statement that its is =20
> >RECOMMENDED that the outbound-proxy-set is configured to result in =20
> >registrations over TCP or TLS whenever the UA can't accept incoming =20
> >TCP flows. Further, include text that explain that for historical =20
> >purposes a lot of deployments still only use UDP, but that=20
> we expect/=20
> >hope those deployments to eventually migrate to TCP when doing=20
> >outbound.
> >
> >Yes (Accept the proposal and add the text) or No (Reject the=20
> proposal =20
> >and keep the text as-is).
> >
> >
> >--
> >Dean Willis (wearing his chair sombrero)
> >
> >_______________________________________________
> >Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> >This list is for NEW development of the core SIP Protocol
> >Use sip-implementors@cs.columbia.edu for questions on current sip
> >Use sipping@ietf.org for new developments on the application of sip
> >
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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This is a somewhat obscure semantic issue. If a box containing a SIP 
implementation registers twice, in order to accept calls addressed to 
two different AORs, is it one UA or two UAs? You can make an argument 
either way, but this is one way to get two user agents in a piece of 
equipment.

Some might consider it one UA if it is a single application supporting 
multiple addresses and similar function. A clearer case is a PC running 
a phone application and an entirely independent sip Instant Messaging 
application.

	Paul

Raj wrote:
> Hello To all,
>  
>                 Recently I read an article and I came across the 
> following line,
>  
> "*Software or hardware appliance containing one or more SIP user agents".*
>  
>              My doubt is considering a phone equipment, can it contain 
> two SIP user agents in it? if yes, can any one tell me about the 
> operation of both the user agents?
>  
> with regards
> Raj,
> 
> ------------------------------------------------------------------------
> Cheap Talk? Check out 
> <http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.com> 
> Yahoo! Messenger's low PC-to-Phone call rates.
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

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Subject: RE: [Sip] About one or more user agents?
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Thread-Topic: [Sip] About one or more user agents?
thread-index: AccdLE0Y+9LxOQ12T2OBlxaprvRB7wACSFNA
From: "Attila Sipos" <Attila.Sipos@vegastream.com>
To: "Raj" <ponrajit@yahoo.com>,
	"SIP forum" <sip@ietf.org>
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=20
Imagine a box with multiple independent POTS ports.
=20
So to work with SIP, each POTS port would have a SIP UA.
=20
Does this help?
=20
Regards,

Attila
=20
=20

-----Original Message-----
From: Raj [mailto:ponrajit@yahoo.com]
Sent: 11 December 2006 13:56
To: SIP forum
Subject: [Sip] About one or more user agents?


Hello To all,
=20
                Recently I read an article and I came across the =
following line,=20
=20
"Software or hardware appliance containing one or more SIP user agents".
=20
             My doubt is considering a phone equipment, can it contain =
two SIP user agents in it? if yes, can any one tell me about the =
operation of both the user agents?
=20
with regards
Raj,



  _____ =20

Cheap Talk? Check  =
<http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com=
/evt=3D39663/*http://voice.yahoo.com> out Yahoo! Messenger's low =
PC-to-Phone call rates.


------_=_NextPart_001_01C71D35.AAD5E92A
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 6.00.2900.2668" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =

size=3D2>Imagine a box with&nbsp;multiple independent POTS=20
ports.</FONT></SPAN></DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =
size=3D2>So to=20
work with SIP, each POTS port would have a SIP UA.</FONT></SPAN></DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =
size=3D2>Does=20
this help?</FONT></SPAN></DIV>
<DIV><SPAN class=3D605520215-11122006></SPAN><SPAN =
class=3D605520215-11122006><FONT=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =

size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =

size=3D2><BR>Attila</FONT></SPAN></DIV>
<DIV><SPAN class=3D605520215-11122006></SPAN><SPAN =
class=3D605520215-11122006><FONT=20
face=3DArial color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D605520215-11122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<BLOCKQUOTE>
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Raj=20
  [mailto:ponrajit@yahoo.com]<BR><B>Sent:</B> 11 December 2006=20
  13:56<BR><B>To:</B> SIP forum<BR><B>Subject:</B> [Sip] About one or =
more user=20
  agents?<BR><BR></FONT></DIV>
  <DIV>Hello To all,</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  =
<DIV>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;=20
  Recently I read an article and I came across the following line, =
</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV>"<STRONG>Software or hardware appliance containing one or more =
SIP user=20
  agents".</STRONG></DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  =
<DIV>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;=20
  My doubt is considering a phone equipment, can it contain two SIP user =
agents=20
  in it? if yes, can any one tell me about the operation of both the =
user=20
  agents?</DIV>
  <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</DIV>
  <DIV>with regards</DIV>
  <DIV>Raj,</DIV>
  <P>
  <HR SIZE=3D1>
  Cheap Talk? <A=20
  =
href=3D"http://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.ya=
hoo.com/evt=3D39663/*http://voice.yahoo.com">Check=20
  out</A> Yahoo! Messenger's low PC-to-Phone call=20
rates.</BLOCKQUOTE></BODY></HTML>

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_______________________________________________
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From sip-bounces@ietf.org Mon Dec 11 11:42:47 2006
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [Sip] Re: issue with oubound and the used transport protocol
Date: Mon, 11 Dec 2006 08:41:33 -0800
To: Fredrik Thulin <ft@it.su.se>
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Hi Fredrik,

In the 3GPP environment I believe the specs would allow a IPsec 
protected TCP connection from the P-CSCF to the UA using a IPsec SA 
already established by a UDP flow.  If a NAT is present, I am told the 
IPsec traffic would be sent in UDP.  I am not convinced this TCP in UDP 
is an improvement and within the spirit of "sending over a congestion 
controlled transport", but I don't want to go down that rathole.

I don't think there is any desire by anyone to do switchover with 
outbound.  I'm not even convinced that anyone really uses it without 
outbound.  It appears that the folks who want to do TCP or TLS will 
only use that transport (no benefit to using UDP if you already have a 
TCP or TLS connection up), and I suspect that outside the 3GPP case the 
UDP folks are planning to only use UDP and use the "get out of jail 
free" statement in RFC 3261 that says if you tried to switched over and 
got an ICMP error, you go back to sending using UDP.

thanks,
-rohan


On Dec 11, 2006, at 0:29, Fredrik Thulin wrote:
> Rohan Mahy wrote:
>> Edge Proxies and Registrars would still need to support outbound with 
>> UDP.
>
> Hi
>
> Before answering Dean's poll, I would like to know more about this.
>
> Have you figured out a (reasonable) way for the edge 
> proxy/authoritative proxy (or the combination thereof) to switch from 
> UDP to TCP for large messages destined for a UAC that uses Outbound?
>
> I'm thinking of the problems in doing that that we discussed briefly 
> in these two messages :
>
>   http://www1.ietf.org/mail-archive/web/sip/current/msg17418.html
>   http://www1.ietf.org/mail-archive/web/sip/current/msg17420.html
>
> Thanks
>
> /Fredrik


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Yes to all.

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]=20
> Sent: Friday, December 08, 2006 18:12
> To: SIP IETF
> Subject: [Sip] Poll: Proposal relating to keepalive, TCP, and=20
> UDP usage in draft-ietf-sip-outbound
>=20
> Previously we had recorded consensus to use STUN for=20
> keepalive with both TCP and UDP. This was not a perfect=20
> consensus (there were some dissenters). There has been=20
> ongoing discussion notably instigated by the current editor=20
> of the Outbound draft, that some people feel may have=20
> resulted in a change of the consensus on this draft's=20
> technical direction.
>=20
> So I'd like to ask people who care and think they understand=20
> the differences to answer the following questions, on-list=20
> and on-the record.
>=20
>=20
> Proposal 1: Make CRLF the only keepalive mechanism for TCP=20
> and TLS over TCP flow (retain STUN for UDP flows).
>=20
> Yes (accept the proposal and use CRLF for TCP) or No (reject=20
> the proposal and keep STUN for TCP)
>=20
>=20
> Proposal 2: Make outbound registrations over UDP flows=20
> optional for UAs.
>=20
> Yes (Accept the proposal) or No (Reject the proposal).
>=20
>=20
> Proposal 3: Include (in Outbound) the statement that its is=20
> RECOMMENDED that the outbound-proxy-set is configured to=20
> result in registrations over TCP or TLS whenever the UA can't=20
> accept incoming TCP flows. Further, include text that explain=20
> that for historical purposes a lot of deployments still only=20
> use UDP, but that we expect/ hope those deployments to=20
> eventually migrate to TCP when doing outbound.
>=20
> Yes (Accept the proposal and add the text) or No (Reject the=20
> proposal and keep the text as-is).
>=20
>=20
> --
> Dean Willis (wearing his chair sombrero)
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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	and UDP usage in draft-ietf-sip-outbound
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	Yes on all three.

Best regards,
Byron Campen

> Previously we had recorded consensus to use STUN for keepalive with  
> both TCP and UDP. This was not a perfect consensus (there were some  
> dissenters). There has been ongoing discussion notably instigated  
> by the current editor of the Outbound draft, that some people feel  
> may have resulted in a change of the consensus on this draft's  
> technical direction.
>
> So I'd like to ask people who care and think they understand the  
> differences to answer the following questions, on-list and on-the  
> record.
>
>
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS  
> over TCP flow (retain STUN for UDP flows).
>
> Yes (accept the proposal and use CRLF for TCP) or No (reject the  
> proposal and keep STUN for TCP)
>
>
> Proposal 2: Make outbound registrations over UDP flows optional for  
> UAs.
>
> Yes (Accept the proposal) or No (Reject the proposal).
>
>
> Proposal 3: Include (in Outbound) the statement that its is  
> RECOMMENDED that the outbound-proxy-set is configured to result in  
> registrations over TCP or TLS whenever the UA can't accept incoming  
> TCP flows. Further, include text that explain that for historical  
> purposes a lot of deployments still only use UDP, but that we  
> expect/hope those deployments to eventually migrate to TCP when  
> doing outbound.
>
> Yes (Accept the proposal and add the text) or No (Reject the  
> proposal and keep the text as-is).
>
>
> --
> Dean Willis (wearing his chair sombrero)
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
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--===============1104495377==--




From sip-bounces@ietf.org Mon Dec 11 16:31:45 2006
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To: Dean Willis <dean.willis@softarmor.com>
Subject: Re: [Sip] Poll: Proposal relating to keepalive, TCP,	and UDP usage
	in draft-ietf-sip-outbound
References: <EE3CDAC2-9BD6-4537-BD02-3A03B286BC1B@softarmor.com>
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Dean Willis wrote:
> Previously we had recorded consensus to use STUN for keepalive with both
> TCP and UDP. This was not a perfect consensus (there were some
> dissenters). There has been ongoing discussion notably instigated by the
> current editor of the Outbound draft, that some people feel may have
> resulted in a change of the consensus on this draft's technical direction.
> 
> So I'd like to ask people who care and think they understand the
> differences to answer the following questions, on-list and on-the record.
> 
> 
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS over
> TCP flow (retain STUN for UDP flows).
> 
> Yes (accept the proposal and use CRLF for TCP) or No (reject the
> proposal and keep STUN for TCP)

Yes

> 
> 
> Proposal 2: Make outbound registrations over UDP flows optional for UAs.
> 
> Yes (Accept the proposal) or No (Reject the proposal).

No

> 
> 
> Proposal 3: Include (in Outbound) the statement that its is RECOMMENDED
> that the outbound-proxy-set is configured to result in registrations
> over TCP or TLS whenever the UA can't accept incoming TCP flows.
> Further, include text that explain that for historical purposes a lot of
> deployments still only use UDP, but that we expect/hope those
> deployments to eventually migrate to TCP when doing outbound.
> 
> Yes (Accept the proposal and add the text) or No (Reject the proposal
> and keep the text as-is).

No

-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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From sip-bounces@ietf.org Tue Dec 12 03:41:38 2006
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	in draft-ietf-sip-outbound
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Dean Willis wrote:
...
> So I'd like to ask people who care and think they understand the 
> differences to answer the following questions, on-list and on-the record.
> 
> 
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS over 
> TCP flow (retain STUN for UDP flows).

Yes

> Proposal 2: Make outbound registrations over UDP flows optional for UAs.

Yes, and I also want it optional for edge proxies/registrars.

> Proposal 3: Include (in Outbound) the statement that its is RECOMMENDED 
> that the outbound-proxy-set is configured to result in registrations 
> over TCP or TLS whenever the UA can't accept incoming TCP flows. 
> Further, include text that explain that for historical purposes a lot of 
> deployments still only use UDP, but that we expect/hope those 
> deployments to eventually migrate to TCP when doing outbound.

Yes

/Fredrik

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From sip-bounces@ietf.org Tue Dec 12 05:21:25 2006
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	ClientInitiated	Connections in the Session Initiation Protocol (SIP)
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>Performance is important at least for service providers, 
>because they have to buy the hardware to run the servers, pay 
>the licenses per CPU, pay the datacenters cost, pay the 
>sysadmins to run the servers and pay the managers to manage 
>the sysadmins.
>
>So yes, performance is important, but perhaps not for the 
>hardware vendors, the software vendors and the sysadmins.

This is all relative.

Your draft seems to add enormous amounts of complexity -- to both UAs and
outbound proxies -- to get around what you claim to be a performance
bottleneck. First of all, I disagree with the assumption that running TCP
only is that much more of a burden. And secondly, I believe complexity is
the biggest enemy here, much more so than the performance concerns. 

Cheers,
Aki


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From: Raj <ponrajit@yahoo.com>
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Hello To all,
   
                  We know that SIP uses TCP or UDP as the transport layer protocol. There is some size restrictions on UDP packets and TCP packets. It is been stated that one packet carries exactly one SIP request or SIP response messages.
   
                  My doubts is if the size of the SIP messages is more than the size of the UDP or TCP packets, then what happens, will the packet will be fragemented or SIP message will be discarded from being transmitted.
   
  with regards
  Raj.

 
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<div>Hello To all,</div>  <div>&nbsp;</div>  <div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We know that SIP uses TCP or UDP as the transport layer protocol. There is some size restrictions on UDP packets and TCP packets. It is been stated that one packet carries exactly one SIP request or SIP response messages.</div>  <div>&nbsp;</div>  <div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; My doubts is if the size of the SIP messages is more than the size of the UDP or TCP packets, then what happens, will the packet will be fragemented or SIP message will be discarded from being transmitted.</div>  <div>&nbsp;</div>  <div>with regards</div>  <div>Raj.</div><p>&#32;

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============2041210169==--




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Hi Raj,

=20

If the SIP packet is too large, message must be represented in "short"
format.

=20

Ex:

Call-ID: 44fer23ei4291dekfer34231232

i:  44fer23ei4291dekfer34231232

=20

From: Thomas Edison<sips:edison@electric.com>;tag=3D532

F: Thomas Edison<sips:edison@electric.com>;tag=3D532

=20

=20

Thanks & Regards

Adarsh.

=20

=20

=20

________________________________

From: Raj [mailto:ponrajit@yahoo.com]=20
Sent: Tuesday, December 12, 2006 5:21 PM
To: SIP forum
Subject: [Sip] Message size - regarding

=20

Hello To all,

=20

                We know that SIP uses TCP or UDP as the transport layer
protocol. There is some size restrictions on UDP packets and TCP
packets. It is been stated that one packet carries exactly one SIP
request or SIP response messages.

=20

                My doubts is if the size of the SIP messages is more
than the size of the UDP or TCP packets, then what happens, will the
packet will be fragemented or SIP message will be discarded from being
transmitted.

=20

with regards

Raj.

 =20

________________________________

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</head>

<body lang=3DEN-US link=3Dblue vlink=3Dblue>

<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi =
Raj,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>If the SIP packet is too large, =
message
must be represented in &#8220;short&#8221; =
format.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Ex:<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3D"#3366ff"
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'>Call-ID:
44fer23ei4291dekfer34231232<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3D"#3366ff"
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'>i: =
&nbsp;44fer23ei4291dekfer34231232<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3D"#3366ff" face=3DArial><span
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'><o:p>&nbsp;</o=
:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3D"#3366ff"
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'>From:
Thomas =
Edison&lt;sips:edison@electric.com&gt;;tag=3D532<o:p></o:p></span></font>=
</p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3D"#3366ff"
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'>F: Thomas
Edison&lt;sips:edison@electric.com&gt;;tag=3D532<o:p></o:p></span></font>=
</p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3Dblack
face=3DCourier><span =
style=3D'font-size:10.0pt;font-family:Courier;color:black'><o:p>&nbsp;</o=
:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3Dblack
face=3DCourier><span =
style=3D'font-size:10.0pt;font-family:Courier;color:black'><o:p>&nbsp;</o=
:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3D"#3366ff"
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'>Thanks
&amp; Regards<o:p></o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3D"#3366ff"
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'>Adarsh.<o:p></=
o:p></span></font></p>

<p class=3DMsoNormal style=3D'text-autospace:none'><font size=3D2 =
color=3D"#3366ff"
face=3DArial><span =
style=3D'font-size:10.0pt;font-family:Arial;color:#3366FF'><o:p>&nbsp;</o=
:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span =
style=3D'font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font =
size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Raj
[mailto:ponrajit@yahoo.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Tuesday, December =
12, 2006
5:21 PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> SIP forum<br>
<b><span style=3D'font-weight:bold'>Subject:</span></b> [Sip] Message =
size -
regarding</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Hello To all,<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We
know that SIP uses TCP or UDP as the transport layer protocol. There is =
some
size restrictions on UDP packets and TCP packets. It is been stated that =
one packet
carries exactly one SIP request or SIP response =
messages.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;
My doubts is if the size of the SIP messages is more than the size of =
the UDP
or TCP packets, then what happens, will the packet will be fragemented =
or SIP
message will be discarded from being =
transmitted.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>&nbsp;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>with regards<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Raj.<o:p></o:p></span></font></p>

</div>

<p><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:12.0pt'>&nbsp; <o:p></o:p></span></font></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font =
size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D1 width=3D"100%" align=3Dcenter>

</span></font></div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
style=3D'font-size:
12.0pt'>Check out <a
href=3D"http://us.rd.yahoo.com/evt=3D43257/*http:/advision.webevents.yaho=
o.com/mailbeta">the
all-new Yahoo! Mail beta</a> - Fire up a more powerful email and get =
things
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
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Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0652456843==--




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Hi Adarsh,
=20
         To represent in 'short', is which part of Implementation?  SIP =
or UDP? =20
thanks & regards,=20
vamsi.
  ----- Original Message -----=20
  From: Adarsh Guler=20
  To: Raj ; SIP forum=20
  Sent: Tuesday, December 12, 2006 5:32 PM
  Subject: RE: [Sip] Message size - regarding


  Hi Raj,

  =20

  If the SIP packet is too large, message must be represented in "short" =
format.

  =20

  Ex:

  Call-ID: 44fer23ei4291dekfer34231232

  i:  44fer23ei4291dekfer34231232

  =20

  From: Thomas Edison<sips:edison@electric.com>;tag=3D532

  F: Thomas Edison<sips:edison@electric.com>;tag=3D532

  =20

  =20

  Thanks & Regards

  Adarsh.

  =20

  =20

  =20


-------------------------------------------------------------------------=
-----

  From: Raj [mailto:ponrajit@yahoo.com]=20
  Sent: Tuesday, December 12, 2006 5:21 PM
  To: SIP forum
  Subject: [Sip] Message size - regarding

  =20

  Hello To all,

  =20

                  We know that SIP uses TCP or UDP as the transport =
layer protocol. There is some size restrictions on UDP packets and TCP =
packets. It is been stated that one packet carries exactly one SIP =
request or SIP response messages.

  =20

                  My doubts is if the size of the SIP messages is more =
than the size of the UDP or TCP packets, then what happens, will the =
packet will be fragemented or SIP message will be discarded from being =
transmitted.

  =20

  with regards

  Raj.

   =20


-------------------------------------------------------------------------=
-----

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<BODY lang=3DEN-US vLink=3Dblue link=3Dblue bgColor=3D#ffffff>
<DIV><FONT color=3D#0000ff>Hi Adarsh,</FONT></DIV>
<DIV><FONT color=3D#0000ff>&nbsp;</FONT></DIV>
<DIV><FONT =
color=3D#0000ff>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; To=20
represent in 'short', is which part of Implementation?&nbsp; SIP or =
UDP?&nbsp;=20
</FONT></DIV>
<DIV><FONT color=3D#0000ff>thanks &amp; regards, </FONT></DIV>
<DIV><FONT color=3D#0000ff>vamsi.</FONT></DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A title=3Dadarshg@TechMahindra.com=20
  href=3D"mailto:adarshg@TechMahindra.com">Adarsh Guler</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Dponrajit@yahoo.com=20
  href=3D"mailto:ponrajit@yahoo.com">Raj</A> ; <A title=3Dsip@ietf.org=20
  href=3D"mailto:sip@ietf.org">SIP forum</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, December 12, =
2006 5:32=20
  PM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> RE: [Sip] Message size =
-=20
  regarding</DIV>
  <DIV><BR></DIV>
  <DIV class=3DSection1>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Hi=20
  Raj,<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">If the SIP =
packet is=20
  too large, message must be represented in =93short=94=20
  format.<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Ex:<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">Call-ID: =

  44fer23ei4291dekfer34231232<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">i:=20
  &nbsp;44fer23ei4291dekfer34231232<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">From: =
Thomas=20
  =
Edison&lt;sips:edison@electric.com&gt;;tag=3D532<o:p></o:p></SPAN></FONT>=
</P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">F: =
Thomas=20
  =
Edison&lt;sips:edison@electric.com&gt;;tag=3D532<o:p></o:p></SPAN></FONT>=
</P>
  <P class=3DMsoNormal><FONT face=3DCourier color=3Dblack size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: =
Courier"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DCourier color=3Dblack size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: =
Courier"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">Thanks =
&amp;=20
  Regards<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: =
Arial">Adarsh.<o:p></o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff size=3D2><SPAN =

  style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
  style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <DIV>
  <DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" =
align=3Dcenter><FONT=20
  face=3D"Times New Roman" size=3D3><SPAN style=3D"FONT-SIZE: 12pt">
  <HR tabIndex=3D-1 align=3Dcenter width=3D"100%" SIZE=3D2>
  </SPAN></FONT></DIV>
  <P class=3DMsoNormal><B><FONT face=3DTahoma size=3D2><SPAN=20
  style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma">From:</SPAN></FONT></B><FONT=20
  face=3DTahoma size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma"> Raj=20
  [mailto:ponrajit@yahoo.com] <BR><B><SPAN=20
  style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Tuesday, December 12, =
2006 5:21=20
  PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> SIP =
forum<BR><B><SPAN=20
  style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> [Sip] Message size -=20
  regarding</SPAN></FONT><o:p></o:p></P></DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt">Hello To =
all,<o:p></o:p></SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: =
12pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We=20
  know that SIP uses TCP or UDP as the transport layer protocol. There =
is some=20
  size restrictions on UDP packets and TCP packets. It is been stated =
that one=20
  packet carries exactly one SIP request or SIP response=20
  messages.<o:p></o:p></SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: =
12pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;=20
  My doubts is if the size of the SIP messages is more than the size of =
the UDP=20
  or TCP packets, then what happens, will the packet will be fragemented =
or SIP=20
  message will be discarded from being=20
  transmitted.<o:p></o:p></SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt">with =
regards<o:p></o:p></SPAN></FONT></P></DIV>
  <DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt">Raj.<o:p></o:p></SPAN></FONT></P></DIV>
  <P><FONT face=3D"Times New Roman" size=3D3><SPAN style=3D"FONT-SIZE: =
12pt">&nbsp;=20
  <o:p></o:p></SPAN></FONT></P>
  <DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" =
align=3Dcenter><FONT=20
  face=3D"Times New Roman" size=3D3><SPAN style=3D"FONT-SIZE: 12pt">
  <HR align=3Dcenter width=3D"100%" SIZE=3D1>
  </SPAN></FONT></DIV>
  <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
  style=3D"FONT-SIZE: 12pt">Check out <A=20
  =
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  all-new Yahoo! Mail beta</A> - Fire up a more powerful email and get =
things=20
  done faster.<o:p></o:p></SPAN></FONT></P></DIV>
  <P>
  <HR>

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Hi Vamsi,
=20
Short representation is a part of SIP itself.=20
=20
Any SIP implementation compliant with RFC3261 should be able to =
recognize header in short or normal form. Even a single SIP request can =
have some header in short form and some in normal, although no =
implemenation will do that as they would designed to send messages =
either in short or the long form.=20
=20
However, with this the compression would not be much, espeically if the =
message contains few headers and a large body. In that case sigcomp can =
be used for compessing the messages.
=20
Regards,
Vineet.

-----Original Message-----
From: vamsi [mailto:annasagaram.vamsi@aftek.com]
Sent: Tuesday, December 12, 2006 7:23 PM
To: sip@ietf.org
Subject: Re: [Sip] Message size - regarding


Hi Adarsh,
=20
         To represent in 'short', is which part of Implementation?  SIP =
or UDP? =20
thanks & regards,=20
vamsi.

----- Original Message -----=20
From: Adarsh Guler <mailto:adarshg@TechMahindra.com> =20
To: Raj <mailto:ponrajit@yahoo.com>  ; SIP forum <mailto:sip@ietf.org> =20
Sent: Tuesday, December 12, 2006 5:32 PM
Subject: RE: [Sip] Message size - regarding



Hi Raj,

=20

If the SIP packet is too large, message must be represented in "short" =
format.

=20

Ex:

Call-ID: 44fer23ei4291dekfer34231232

i:  44fer23ei4291dekfer34231232

=20

From: Thomas Edison<sips:edison@electric.com>;tag=3D532

F: Thomas Edison<sips:edison@electric.com>;tag=3D532

=20

=20

Thanks & Regards

Adarsh.

=20

=20

=20


  _____ =20


From: Raj [mailto:ponrajit@yahoo.com]=20
Sent: Tuesday, December 12, 2006 5:21 PM
To: SIP forum
Subject: [Sip] Message size - regarding

=20

Hello To all,

=20

                We know that SIP uses TCP or UDP as the transport layer =
protocol. There is some size restrictions on UDP packets and TCP =
packets. It is been stated that one packet carries exactly one SIP =
request or SIP response messages.

=20

                My doubts is if the size of the SIP messages is more =
than the size of the UDP or TCP packets, then what happens, will the =
packet will be fragemented or SIP message will be discarded from being =
transmitted.

=20

with regards

Raj.

 =20


  _____ =20


Check out the  =
<http://us.rd.yahoo.com/evt=3D43257/*http:/advision.webevents.yahoo.com/m=
ailbeta> all-new Yahoo! Mail beta - Fire up a more powerful email and =
get things done faster.



  _____ =20




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip


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<BODY lang=3DEN-US vLink=3Dblue link=3Dblue bgColor=3D#ffffff>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =
size=3D2>Hi=20
Vamsi,</FONT></SPAN></DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =
size=3D2>Short=20
representation is a part of SIP itself. </FONT></SPAN></DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =
size=3D2>Any=20
SIP implementation compliant with RFC3261 should be able to recognize =
header in=20
short or normal form. Even a single SIP request can have some header in =
short=20
form and some in normal, although no implemenation will do that as they =
would=20
designed to send messages either in short or the long form. =
</FONT></SPAN></DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =

size=3D2>However, with this the compression would not be much, =
espeically if the=20
message contains few headers and a large body. In that case sigcomp can =
be used=20
for compessing the messages.</FONT></SPAN></DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =

size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D966185413-12122006><FONT face=3DArial color=3D#0000ff =

size=3D2>Vineet.</FONT></SPAN></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader dir=3Dltr align=3Dleft><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> vamsi=20
  [mailto:annasagaram.vamsi@aftek.com]<BR><B>Sent:</B> Tuesday, December =
12,=20
  2006 7:23 PM<BR><B>To:</B> sip@ietf.org<BR><B>Subject:</B> Re: [Sip] =
Message=20
  size - regarding<BR><BR></FONT></DIV>
  <DIV><FONT color=3D#0000ff>Hi Adarsh,</FONT></DIV>
  <DIV><FONT color=3D#0000ff></FONT>&nbsp;</DIV>
  <DIV><FONT =
color=3D#0000ff>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; To=20
  represent in 'short', is which part of Implementation?&nbsp; SIP or =
UDP?&nbsp;=20
  </FONT></DIV>
  <DIV><FONT color=3D#0000ff>thanks &amp; regards, </FONT></DIV>
  <DIV><FONT color=3D#0000ff>vamsi.</FONT></DIV>
  <BLOCKQUOTE=20
  style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
    <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
    <DIV=20
    style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
    <A title=3Dadarshg@TechMahindra.com=20
    href=3D"mailto:adarshg@TechMahindra.com">Adarsh Guler</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A =
title=3Dponrajit@yahoo.com=20
    href=3D"mailto:ponrajit@yahoo.com">Raj</A> ; <A title=3Dsip@ietf.org =

    href=3D"mailto:sip@ietf.org">SIP forum</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Tuesday, December 12, =
2006 5:32=20
    PM</DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> RE: [Sip] Message =
size -=20
    regarding</DIV>
    <DIV><FONT face=3DArial color=3D#0000ff size=3D2></FONT><FONT =
face=3DArial=20
    color=3D#0000ff size=3D2></FONT><BR></DIV>
    <DIV class=3DSection1>
    <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Hi=20
    Raj,<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">If the =
SIP packet=20
    is too large, message must be represented in "short"=20
    format.<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial">Ex:<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: =
Arial">Call-ID:=20
    44fer23ei4291dekfer34231232<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">i:=20
    &nbsp;44fer23ei4291dekfer34231232<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">From: =
Thomas=20
    =
Edison&lt;sips:edison@electric.com&gt;;tag=3D532<o:p></o:p></SPAN></FONT>=
</P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">F: =
Thomas=20
    =
Edison&lt;sips:edison@electric.com&gt;;tag=3D532<o:p></o:p></SPAN></FONT>=
</P>
    <P class=3DMsoNormal><FONT face=3DCourier color=3Dblack =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: =
Courier"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DCourier color=3Dblack =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: =
Courier"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: Arial">Thanks =
&amp;=20
    Regards<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: =
Arial">Adarsh.<o:p></o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3D#3366ff =
size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: #3366ff; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <P class=3DMsoNormal><FONT face=3DArial color=3Dnavy size=3D2><SPAN=20
    style=3D"FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: =
Arial"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <DIV>
    <DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" =
align=3Dcenter><FONT=20
    face=3D"Times New Roman" size=3D3><SPAN style=3D"FONT-SIZE: 12pt">
    <HR tabIndex=3D-1 align=3Dcenter width=3D"100%" SIZE=3D2>
    </SPAN></FONT></DIV>
    <P class=3DMsoNormal><B><FONT face=3DTahoma size=3D2><SPAN=20
    style=3D"FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma">From:</SPAN></FONT></B><FONT=20
    face=3DTahoma size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Tahoma"> Raj=20
    [mailto:ponrajit@yahoo.com] <BR><B><SPAN=20
    style=3D"FONT-WEIGHT: bold">Sent:</SPAN></B> Tuesday, December 12, =
2006 5:21=20
    PM<BR><B><SPAN style=3D"FONT-WEIGHT: bold">To:</SPAN></B> SIP=20
    forum<BR><B><SPAN style=3D"FONT-WEIGHT: bold">Subject:</SPAN></B> =
[Sip]=20
    Message size - regarding</SPAN></FONT><o:p></o:p></P></DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt"><o:p>&nbsp;</o:p></SPAN></FONT></P>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt">Hello To =
all,<o:p></o:p></SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: =
12pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We=20
    know that SIP uses TCP or UDP as the transport layer protocol. There =
is some=20
    size restrictions on UDP packets and TCP packets. It is been stated =
that one=20
    packet carries exactly one SIP request or SIP response=20
    messages.<o:p></o:p></SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: =
12pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;=20
    My doubts is if the size of the SIP messages is more than the size =
of the=20
    UDP or TCP packets, then what happens, will the packet will be =
fragemented=20
    or SIP message will be discarded from being=20
    transmitted.<o:p></o:p></SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt">&nbsp;<o:p></o:p></SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt">with =
regards<o:p></o:p></SPAN></FONT></P></DIV>
    <DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt">Raj.<o:p></o:p></SPAN></FONT></P></DIV>
    <P><FONT face=3D"Times New Roman" size=3D3><SPAN style=3D"FONT-SIZE: =
12pt">&nbsp;=20
    <o:p></o:p></SPAN></FONT></P>
    <DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" =
align=3Dcenter><FONT=20
    face=3D"Times New Roman" size=3D3><SPAN style=3D"FONT-SIZE: 12pt">
    <HR align=3Dcenter width=3D"100%" SIZE=3D1>
    </SPAN></FONT></DIV>
    <P class=3DMsoNormal><FONT face=3D"Times New Roman" size=3D3><SPAN=20
    style=3D"FONT-SIZE: 12pt">Check out <A=20
    =
href=3D"http://us.rd.yahoo.com/evt=3D43257/*http:/advision.webevents.yaho=
o.com/mailbeta">the=20
    all-new Yahoo! Mail beta</A> - Fire up a more powerful email and get =
things=20
    done faster.<o:p></o:p></SPAN></FONT></P></DIV>
    <P>
    <HR>

    <P></P>_______________________________________________<BR>Sip =
mailing=20
    list&nbsp; https://www1.ietf.org/mailman/listinfo/sip<BR>This list =
is for=20
    NEW development of the core SIP Protocol<BR>Use=20
    sip-implementors@cs.columbia.edu for questions on current sip<BR>Use =

    sipping@ietf.org for new developments on the application of=20
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0463435990==--




From sip-bounces@ietf.org Tue Dec 12 09:32:22 2006
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Date: Tue, 12 Dec 2006 15:32:00 +0100
From: "Hisham Khartabil" <hisham.khartabil@gmail.com>
To: Raj <ponrajit@yahoo.com>
Subject: Re: [Sip] Message size - regarding
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On 12/12/06, Raj <ponrajit@yahoo.com> wrote:
>
> Hello To all,
>
>                 We know that SIP uses TCP or UDP as the transport layer
> protocol. There is some size restrictions on UDP packets and TCP packets. It
> is been stated that one packet carries exactly one SIP request or SIP
> response messages.
>

This is only true for  UDP. For TCP, you can have the sip messages
fragmented.

Hisham


                My doubts is if the size of the SIP messages is more than
> the size of the UDP or TCP packets, then what happens, will the packet will
> be fragemented or SIP message will be discarded from being transmitted.
>
> with regards
> Raj.
>
> ------------------------------
> Check out the all-new Yahoo! Mail beta<http://us.rd.yahoo.com/evt=43257/*http://advision.webevents.yahoo.com/mailbeta>- Fire up a more powerful email and get things done faster.
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>
>

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<br><br><div><span class="gmail_quote">On 12/12/06, <b class="gmail_sendername">Raj</b> &lt;<a href="mailto:ponrajit@yahoo.com">ponrajit@yahoo.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>Hello To all,</div>  <div>&nbsp;</div>  <div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We know that SIP uses TCP or UDP as the transport layer protocol. There is some size restrictions on UDP packets and TCP packets. It is been stated that one packet carries exactly one SIP request or SIP response messages.
</div></blockquote><div><br>This is only true for&nbsp; UDP. For TCP, you can have the sip messages fragmented.<br><br>Hisham<br>&nbsp;</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; My doubts is if the size of the SIP messages is more than the size of the UDP or TCP packets, then what happens, will the packet will be fragemented or SIP message will be discarded from being transmitted.
</div>  <div>&nbsp;</div>  <div>with regards</div><span class="sg">  <div>Raj.</div></span><span class="ad"><p> 

</p><hr size="1">Check out <a href="http://us.rd.yahoo.com/evt=43257/*http://advision.webevents.yahoo.com/mailbeta" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">the all-new Yahoo! Mail beta</a> - Fire up a more powerful email and get things done faster.
<p></p></span><br>_______________________________________________<br>Sip mailing list &nbsp;<a onclick="return top.js.OpenExtLink(window,event,this)" href="https://www1.ietf.org/mailman/listinfo/sip" target="_blank">https://www1.ietf.org/mailman/listinfo/sip
</a><br>This list is for NEW development of the core SIP Protocol<br>Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sip-implementors@cs.columbia.edu">sip-implementors@cs.columbia.edu</a> for questions on current sip
<br>Use <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:sipping@ietf.org">sipping@ietf.org</a> for new developments on the application of sip<br><br></blockquote></div><br>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1056362195==--




From sip-bounces@ietf.org Tue Dec 12 09:57:48 2006
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Subject: Re: [Sip] Message size - regarding
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   From: Raj <ponrajit@yahoo.com>

   It is been stated that one packet carries exactly one SIP request
   or SIP response messages.

That is correct for UDP, but not for TCP.

Dale

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This list is for NEW development of the core SIP Protocol
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From searchThe@gikas.org Tue Dec 12 11:56:47 2006
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From:	"ti piaciuto" <searchThe@gikas.org>
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From sip-bounces@ietf.org Tue Dec 12 12:44:19 2006
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From: Marc Petit-Huguenin <petithug@acm.org>
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To: Aki Niemi <aki.niemi@nokia.com>
Subject: Re: [Sip] I-D: Preventing Fragmentation for
	ClientInitiated	Connections in the Session Initiation Protocol (SIP)
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Aki Niemi wrote:
>> Performance is important at least for service providers, 
>> because they have to buy the hardware to run the servers, pay 
>> the licenses per CPU, pay the datacenters cost, pay the 
>> sysadmins to run the servers and pay the managers to manage 
>> the sysadmins.
>>
>> So yes, performance is important, but perhaps not for the 
>> hardware vendors, the software vendors and the sysadmins.
> 
> This is all relative.
> 
> Your draft seems to add enormous amounts of complexity -- to both UAs and
> outbound proxies -- to get around what you claim to be a performance
> bottleneck. 

No, this draft is not about performance.  This draft is about be able to
use UDP or TCP with Outbound, i.e. be compliant with RFC 3261 (Section
18: "All SIP elements MUST implement UDP and TCP."

The performance issue is about implementers be able to use UDP or TCP or
whatever they want, as explicitly authorized by RFC 3261.

> First of all, I disagree with the assumption that running TCP
> only is that much more of a burden. 

This is an implementation problem. Outbound MUST support TCP and UDP
because RFC 3261 says it.  The only thing I ask is to let implementers
do their job.

> And secondly, I believe complexity is
> the biggest enemy here, much more so than the performance concerns. 

Well, complexity is relative, and again it is not about performance, but
about compliance.

I am currently implementing my two drafts about fragmentation in my
endpoints and servers, so I will have a better idea of the real
complexity soon.  I will eventually ask the chairs for a slot at the
68th IETF meeting to present this.

-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Tue Dec 12 14:29:22 2006
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From: Byron Campen <bcampen@estacado.net>
Subject: Re: [Sip] outbound-06 submitted
Date: Tue, 12 Dec 2006 13:26:49 -0600
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	I have a question about the usage of STUN as a keepalive over UDP.  
Are clients supposed to start sending STUN before they try to  
REGISTER? If not, I can see the scenario where a UA sends the  
REGISTER, sets a timer (btw 24-29 seconds) for the first STUN  
keepalive, but before that STUN request is sent, the pinhole closes.  
Both ends will think the flow is ok (the STUN keepalive will open a  
new pinhole, but the UA won't know that), and the mess won't get  
cleaned up until the UA re-registers. Sending the keepalive "soon"  
after the REGISTER would probably solve this problem also, if sending  
before is not acceptable.

Best regards,
Byron Campen

> Hi,
>
> I submitted -06 of outbound so folks can at least take a look to see
> that the many small changes we have agreed to have been incorporated.
> I also included the proposed text for FASTEST, since the draft would
> need significant surgery anyway if we go with CLEANEST.
>
> Until it appears in the archive you can find it here:
> http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip- 
> outbound-06.html
> http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip- 
> outbound-06.txt
>
> thanks,
> -rohan
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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Use sipping@ietf.org for new developments on the application of sip
--===============0192410790==--




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Hi Byron,

We could add text recommending that the UA compare the rport/received
with the XOR-MAPPED-ADDRESS.  That would appear to solve the problem.
Can you think of any problem with this?

thanks,
-rohan

On 12/12/06, Byron Campen <bcampen@estacado.net> wrote:
>         I have a question about the usage of STUN as a keepalive over UDP.
> Are clients supposed to start sending STUN before they try to
> REGISTER? If not, I can see the scenario where a UA sends the
> REGISTER, sets a timer (btw 24-29 seconds) for the first STUN
> keepalive, but before that STUN request is sent, the pinhole closes.
> Both ends will think the flow is ok (the STUN keepalive will open a
> new pinhole, but the UA won't know that), and the mess won't get
> cleaned up until the UA re-registers. Sending the keepalive "soon"
> after the REGISTER would probably solve this problem also, if sending
> before is not acceptable.
>
> Best regards,
> Byron Campen
>
> > Hi,
> >
> > I submitted -06 of outbound so folks can at least take a look to see
> > that the many small changes we have agreed to have been incorporated.
> > I also included the proposed text for FASTEST, since the draft would
> > need significant surgery anyway if we go with CLEANEST.
> >
> > Until it appears in the archive you can find it here:
> > http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-
> > outbound-06.html
> > http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-
> > outbound-06.txt
> >
> > thanks,
> > -rohan
> >
> > _______________________________________________
> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP Protocol
> > Use sip-implementors@cs.columbia.edu for questions on current sip
> > Use sipping@ietf.org for new developments on the application of sip
>
>
>
>

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============0645926029==--




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--===============0832611532==--




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Date: Wed, 13 Dec 2006 11:47:13 +0800
From: Xiao Wang <wangsmile@huawei.com>
Subject: RE: [Sip] Message size - regarding
In-reply-to: <20061212115119.43142.qmail@web52904.mail.yahoo.com>
To: 'Raj' <ponrajit@yahoo.com>
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Hi, Raj

 

         It is no problem for TCP, because TCP can prevent IP fragment by
MTU. Otherwise, UDP can occur to IP fragment.

So, in order to involve this problem, using TCP send SIP  message when the
size of SIP message more than the size of UDP.

But in a SIP transaction, it must use the same transport layer protocol.
Thus there may be a problem, if SIP request use UDP, 

corresponding SIP response use UDP, too. However, SIP response message size
may beyond the size of UDP. If the SIP response

is sent by UDP,  IP fragment will be  appearance.

 

Xiao

  _____  

From: Raj [mailto:ponrajit@yahoo.com] 
Sent: Tuesday, December 12, 2006 7:51 PM
To: SIP forum
Subject: [Sip] Message size - regarding

 

Hello To all,

 

                We know that SIP uses TCP or UDP as the transport layer
protocol. There is some size restrictions on UDP packets and TCP packets. It
is been stated that one packet carries exactly one SIP request or SIP
response messages.

 

                My doubts is if the size of the SIP messages is more than
the size of the UDP or TCP packets, then what happens, will the packet will
be fragemented or SIP message will be discarded from being transmitted.

 

with regards

Raj.

  

  _____  

Check out the
<http://us.rd.yahoo.com/evt=43257/*http:/advision.webevents.yahoo.com/mailbe
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<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>Hi, Raj</span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; It
is no problem for TCP, because TCP can prevent IP fragment by MTU. Otherwise, UDP
can occur to IP fragment.</span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>So, in order to involve
this problem, using TCP send SIP &nbsp;message when the size of SIP message
more than the size of UDP.</span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>But in a SIP transaction,
it must use the same transport layer protocol. Thus there may be a problem, if SIP
request use UDP, </span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>corresponding SIP response
use UDP, too. However, SIP response message size may beyond the size of UDP. If
the SIP response</span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>is sent by UDP,&nbsp; IP fragment
will be &nbsp;appearance.</span></font></p>

<p class=MsoNormal><font size=1 color=navy face=Arial><span lang=EN-US
style='font-size:9.0pt;font-family:Arial;color:navy'>&nbsp;</span></font></p>

<div>

<p class=MsoNormal><font size=2 color=navy face="Times New Roman"><span
lang=EN-US style='font-size:10.0pt;color:navy'>Xiao</span></font></p>

</div>

<div style='border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'>

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<div class=MsoNormal align=center style='text-align:center'><font size=3
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<hr size=2 width="100%" align=center tabindex=-1>

</span></font></div>

<p class=MsoNormal><b><font size=2 face=Tahoma><span lang=EN-US
style='font-size:10.0pt;font-family:Tahoma;font-weight:bold'>From:</span></font></b><font
size=2 face=Tahoma><span lang=EN-US style='font-size:10.0pt;font-family:Tahoma'>
Raj [mailto:ponrajit@yahoo.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Tuesday, December 12, 2006
7:51 PM<br>
<b><span style='font-weight:bold'>To:</span></b> SIP forum<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Sip] Message size -
regarding</span></font></p>

</div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;</span></font></p>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>Hello To all,</span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;We
know that SIP uses TCP or UDP as the transport layer protocol. There is some
size restrictions on UDP packets and TCP packets. It is been stated that one
packet carries exactly one SIP request or SIP response messages.</span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
My doubts is if the size of the SIP messages is more than the size of the UDP
or TCP packets, then what happens, will the packet will be fragemented or SIP
message will be discarded from being transmitted.</span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>&nbsp;</span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>with regards</span></font></p>

</div>

<div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>Raj.</span></font></p>

</div>

<p><font size=3 face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>&nbsp;
</span></font></p>

<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span lang=EN-US style='font-size:12.0pt'>

<hr size=1 width="100%" align=center>

</span></font></div>

<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>Check out <a
href="http://us.rd.yahoo.com/evt=43257/*http:/advision.webevents.yahoo.com/mailbeta">the
all-new Yahoo! Mail beta</a> - Fire up a more powerful email and get things
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--===============1334737397==--




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=20
Hi,

>This is an implementation problem. Outbound MUST support TCP and UDP
>because RFC 3261 says it.  The only thing I ask is to let implementers
>do their job.
>
>> And secondly, I believe complexity is
>> the biggest enemy here, much more so than the performance concerns.=20
>
>Well, complexity is relative, and again it is not about=20
>performance, but about compliance.

This reasoning sounds very confusing to me.

Is your suggestion really like this:

a) You do not actually have any technical reason why Outbound
   must support UDP, not even the performance would do.=20

b) The only reason for Outbound to support UDP would be its
   full compatibility to RFC 3261.

If this is the case, I would ask WG to consider this reasoning
very carefully and if it makes any sense or not. To me it sounds
like: "UDP is actually not needed but because the WG decided to
mandate it a few years ago, it just has to be there."

According to your I-D the client has to anyway support TCP (or
other congestion controlled transport) so it would not even allow
to implement UDP-only clients (which might be desirable for
some vendors that currently only support UDP). Thus what is
the point with the UDP flows ? Why could they not be optional
for an UA ?

Regards,

Erkki

>-----Original Message-----
>From: ext Marc Petit-Huguenin [mailto:petithug@acm.org]=20
>Sent: 12.December.2006 19:42
>To: Niemi Aki (Nokia-NRC/Helsinki)
>Cc: 'IETF SIP List'; 'Francois Audet'
>Subject: Re: [Sip] I-D: Preventing Fragmentation=20
>forClientInitiated Connections in the Session Initiation Protocol (SIP)
>
>Aki Niemi wrote:
>>> Performance is important at least for service providers,=20
>>> because they have to buy the hardware to run the servers, pay=20
>>> the licenses per CPU, pay the datacenters cost, pay the=20
>>> sysadmins to run the servers and pay the managers to manage=20
>>> the sysadmins.
>>>
>>> So yes, performance is important, but perhaps not for the=20
>>> hardware vendors, the software vendors and the sysadmins.
>>=20
>> This is all relative.
>>=20
>> Your draft seems to add enormous amounts of complexity -- to=20
>both UAs and
>> outbound proxies -- to get around what you claim to be a performance
>> bottleneck.=20
>
>No, this draft is not about performance.  This draft is about=20
>be able to
>use UDP or TCP with Outbound, i.e. be compliant with RFC 3261 (Section
>18: "All SIP elements MUST implement UDP and TCP."
>
>The performance issue is about implementers be able to use UDP=20
>or TCP or
>whatever they want, as explicitly authorized by RFC 3261.
>
>> First of all, I disagree with the assumption that running TCP
>> only is that much more of a burden.=20
>
>This is an implementation problem. Outbound MUST support TCP and UDP
>because RFC 3261 says it.  The only thing I ask is to let implementers
>do their job.
>
>> And secondly, I believe complexity is
>> the biggest enemy here, much more so than the performance concerns.=20
>
>Well, complexity is relative, and again it is not about=20
>performance, but
>about compliance.
>
>I am currently implementing my two drafts about fragmentation in my
>endpoints and servers, so I will have a better idea of the real
>complexity soon.  I will eventually ask the chairs for a slot at the
>68th IETF meeting to present this.
>
>--=20
>Marc Petit-Huguenin           [                                 ]
>Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
>Work: marc@8x8.com            [                                 ]
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

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From sip-bounces@ietf.org Wed Dec 13 01:26:20 2006
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Subject: RE: [Sip] Poll: Proposal relating to keepalive, TCP,
	and UDP usagein draft-ietf-sip-outbound
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From: <Erkki.Koivusalo@nokia.com>
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Hi,

It seems to me that the I-D Preventing Fragmentation for
Client Initiated Connections in the Session Initiation Protocol (SIP)
written by Marc Petit-Huguenin demonstrates very well the resons=20
why UDP support for Outbound UAs should be made optional. Trying=20
to really solve the problems with UDP together with Outbound would=20
make the total solution more and more complex.=20

>> Proposal 2: Make outbound registrations over UDP flows=20
>optional for UAs.
>>=20
>> Yes (Accept the proposal) or No (Reject the proposal).
>
>No

Based on another email from Marc I understood that the reason
to his No is the compatibility to RFC 3261 which mandates UDP.
I would like to know if there any other, more technical reasons=20
(rather than the old WG decision within RFC 3261) to mandate UDP=20
with Outbound ? This far I have not heard any convincing one.

Regards,

Erkki

>-----Original Message-----
>From: ext Marc Petit-Huguenin [mailto:petithug@acm.org]=20
>Sent: 11.December.2006 23:31
>To: Dean Willis
>Cc: SIP IETF
>Subject: Re: [Sip] Poll: Proposal relating to keepalive, TCP,=20
>and UDP usagein draft-ietf-sip-outbound
>
>Dean Willis wrote:
>> Previously we had recorded consensus to use STUN for=20
>keepalive with both
>> TCP and UDP. This was not a perfect consensus (there were some
>> dissenters). There has been ongoing discussion notably=20
>instigated by the
>> current editor of the Outbound draft, that some people feel may have
>> resulted in a change of the consensus on this draft's=20
>technical direction.
>>=20
>> So I'd like to ask people who care and think they understand the
>> differences to answer the following questions, on-list and=20
>on-the record.
>>=20
>>=20
>> Proposal 1: Make CRLF the only keepalive mechanism for TCP=20
>and TLS over
>> TCP flow (retain STUN for UDP flows).
>>=20
>> Yes (accept the proposal and use CRLF for TCP) or No (reject the
>> proposal and keep STUN for TCP)
>
>Yes
>
>>=20
>>=20
>> Proposal 2: Make outbound registrations over UDP flows=20
>optional for UAs.
>>=20
>> Yes (Accept the proposal) or No (Reject the proposal).
>
>No
>
>>=20
>>=20
>> Proposal 3: Include (in Outbound) the statement that its is=20
>RECOMMENDED
>> that the outbound-proxy-set is configured to result in registrations
>> over TCP or TLS whenever the UA can't accept incoming TCP flows.
>> Further, include text that explain that for historical=20
>purposes a lot of
>> deployments still only use UDP, but that we expect/hope those
>> deployments to eventually migrate to TCP when doing outbound.
>>=20
>> Yes (Accept the proposal and add the text) or No (Reject the proposal
>> and keep the text as-is).
>
>No
>
>--=20
>Marc Petit-Huguenin           [                                 ]
>Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
>Work: marc@8x8.com            [                                 ]
>
>_______________________________________________
>Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>This list is for NEW development of the core SIP Protocol
>Use sip-implementors@cs.columbia.edu for questions on current sip
>Use sipping@ietf.org for new developments on the application of sip
>

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From sip-bounces@ietf.org Wed Dec 13 01:33:48 2006
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From: Juha Heinanen <jh@tutpro.com>
To: <Erkki.Koivusalo@nokia.com>
Subject: RE: [Sip] I-D: Preventing Fragmentation
	forClientInitiated	Connections in the Session Initiation
	Protocol (SIP)
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just a reality check.  yesterday when i was going to make some sip tcp
tests, i tried to find a sip phone that supports tcp.  the most popular
ones from linksys/sipura (cisco) and grandstream didn't.

-- juha


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From sip-bounces@ietf.org Wed Dec 13 09:53:26 2006
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From: Byron Campen <bcampen@estacado.net>
Subject: Re: [Sip] outbound-06 submitted
Date: Wed, 13 Dec 2006 08:50:05 -0600
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	That sounds reasonable to me.

Best regards,
Byron Campen

> Hi Byron,
>
> We could add text recommending that the UA compare the rport/received
> with the XOR-MAPPED-ADDRESS.  That would appear to solve the problem.
> Can you think of any problem with this?
>
> thanks,
> -rohan
>
> On 12/12/06, Byron Campen <bcampen@estacado.net> wrote:
>>         I have a question about the usage of STUN as a keepalive  
>> over UDP.
>> Are clients supposed to start sending STUN before they try to
>> REGISTER? If not, I can see the scenario where a UA sends the
>> REGISTER, sets a timer (btw 24-29 seconds) for the first STUN
>> keepalive, but before that STUN request is sent, the pinhole closes.
>> Both ends will think the flow is ok (the STUN keepalive will open a
>> new pinhole, but the UA won't know that), and the mess won't get
>> cleaned up until the UA re-registers. Sending the keepalive "soon"
>> after the REGISTER would probably solve this problem also, if sending
>> before is not acceptable.
>>
>> Best regards,
>> Byron Campen
>>
>> > Hi,
>> >
>> > I submitted -06 of outbound so folks can at least take a look to  
>> see
>> > that the many small changes we have agreed to have been  
>> incorporated.
>> > I also included the proposed text for FASTEST, since the draft  
>> would
>> > need significant surgery anyway if we go with CLEANEST.
>> >
>> > Until it appears in the archive you can find it here:
>> > http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-
>> > outbound-06.html
>> > http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-
>> > outbound-06.txt
>> >
>> > thanks,
>> > -rohan
>> >
>> > _______________________________________________
>> > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> > This list is for NEW development of the core SIP Protocol
>> > Use sip-implementors@cs.columbia.edu for questions on current sip
>> > Use sipping@ietf.org for new developments on the application of sip
>>
>>
>>
>>


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Erkki.Koivusalo@nokia.com wrote:
>  
> Hi,
> 
>> This is an implementation problem. Outbound MUST support TCP and UDP
>> because RFC 3261 says it.  The only thing I ask is to let implementers
>> do their job.
>>
>>> And secondly, I believe complexity is
>>> the biggest enemy here, much more so than the performance concerns. 
>> Well, complexity is relative, and again it is not about 
>> performance, but about compliance.
> 
> This reasoning sounds very confusing to me.
> 
> Is your suggestion really like this:
> 
> a) You do not actually have any technical reason why Outbound
>    must support UDP, not even the performance would do. 

No, performance is still the reason why *I* will mix UDP and TCP in my
endpoints and servers.  The discussions about making UDP obsolete are
not new, it is always someone claiming that SIP over TCP has the same
performances than SIP over UDP, it is just that the SIP implementers are
morons who do not know how to use select or poll, or epoll, etc...
At first I believed this, and I was ready to move all our customers to
TCP only, until something I read made me doubt this.  So I asked our QA
department to write some tests and to run them on our servers.  The I-D
was written because of the result of this tests.

The problem with performance is that if I publish my results someone
else will come with different results to prove the opposite.  Also not
everybody is interested in performance.  I am, because we have to pay
for every server, data center, etc... to run SIP.  I guess that the
vendors selling the servers, or the software licenses does not care as much.

So I think that we should accept that different members of this WG have
different needs and that we should not have any performance discussion,
as long as the implementers have the tools they need to design servers
and endpoints that works for them.

So this is why I am switching my focus from performance to 1) do not
prevent implementers to use UDP if they need it and 2) anyway RFC 3261
mandates UDP AND TCP on all elements.

> 
> b) The only reason for Outbound to support UDP would be its
>    full compatibility to RFC 3261.

This is a good enough reason.

> 
> If this is the case, I would ask WG to consider this reasoning
> very carefully and if it makes any sense or not. To me it sounds
> like: "UDP is actually not needed but because the WG decided to
> mandate it a few years ago, it just has to be there."
> 
> According to your I-D the client has to anyway support TCP (or
> other congestion controlled transport) so it would not even allow
> to implement UDP-only clients (which might be desirable for
> some vendors that currently only support UDP). Thus what is
> the point with the UDP flows ? Why could they not be optional
> for an UA ?

UDP-only clients are not SIP compliant.  So how a UDP flow can be
optional if UDP in the endpoint is not?

> 
> Regards,
> 
> Erkki
> 
>> -----Original Message-----
>> From: ext Marc Petit-Huguenin [mailto:petithug@acm.org] 
>> Sent: 12.December.2006 19:42
>> To: Niemi Aki (Nokia-NRC/Helsinki)
>> Cc: 'IETF SIP List'; 'Francois Audet'
>> Subject: Re: [Sip] I-D: Preventing Fragmentation 
>> forClientInitiated Connections in the Session Initiation Protocol (SIP)
>>
>> Aki Niemi wrote:
>>>> Performance is important at least for service providers, 
>>>> because they have to buy the hardware to run the servers, pay 
>>>> the licenses per CPU, pay the datacenters cost, pay the 
>>>> sysadmins to run the servers and pay the managers to manage 
>>>> the sysadmins.
>>>>
>>>> So yes, performance is important, but perhaps not for the 
>>>> hardware vendors, the software vendors and the sysadmins.
>>> This is all relative.
>>>
>>> Your draft seems to add enormous amounts of complexity -- to 
>> both UAs and
>>> outbound proxies -- to get around what you claim to be a performance
>>> bottleneck. 
>> No, this draft is not about performance.  This draft is about 
>> be able to
>> use UDP or TCP with Outbound, i.e. be compliant with RFC 3261 (Section
>> 18: "All SIP elements MUST implement UDP and TCP."
>>
>> The performance issue is about implementers be able to use UDP 
>> or TCP or
>> whatever they want, as explicitly authorized by RFC 3261.
>>
>>> First of all, I disagree with the assumption that running TCP
>>> only is that much more of a burden. 
>> This is an implementation problem. Outbound MUST support TCP and UDP
>> because RFC 3261 says it.  The only thing I ask is to let implementers
>> do their job.
>>
>>> And secondly, I believe complexity is
>>> the biggest enemy here, much more so than the performance concerns. 
>> Well, complexity is relative, and again it is not about 
>> performance, but
>> about compliance.
>>
>> I am currently implementing my two drafts about fragmentation in my
>> endpoints and servers, so I will have a better idea of the real
>> complexity soon.  I will eventually ask the chairs for a slot at the
>> 68th IETF meeting to present this.
>>


-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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Jari,

In chapter 4.3.1 In my opinion the "path" parameter shall be stated as
mandatory.

In chapter 4.6.3 the same handling is defined for selector locating
non-existing element and multiple elements. I'd rather distinguish these
cases as subscription to non-existing element may have real usage (as
stated in chapter 4.1), but subscription to multiple elements (in the
sense of one selector for multiple elements) shall not be possible at
all.

Regards,

Pavel

-----Original Message-----
From: Jari Urpalainen [mailto:jari.urpalainen@nokia.com]=20
Sent: Monday, December 11, 2006 1:01 AM
To: SIP IETF
Subject: [Sip] xcap-diff-event

Hi all!

I've submitted an event package i-d=20
<http://www.ietf.org/internet-drafts/draft-urpalainen-sip-xcap-diff-even
t-00.txt>=20
about syncing xcap resources with xcap-diff format. The event package=20
part is pretty much similar what Jonathan had in xcap-package before it=20
was "moved" into the config-framework. The biggest difference is=20
probably an OMA requirement to have subscriptions to XCAP components=20
also. Any comments ?
thanks, Jari

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Marc Petit-Huguenin wrote:
...
> So this is why I am switching my focus from performance to 1) do not
> prevent implementers to use UDP if they need it and 2) anyway RFC 3261
> mandates UDP AND TCP on all elements.

Not even RFC3261 manages to get you everything if using UDP though - 
large messages for one, security (SIPS) for another.

As I see it, it is too complicated to try and make SIP over UDP work 
across NATs to even try it. For those cases where the properties of 
Outbound are desired, everyone will be better off using TCP.

For the cases where you have small messages, don't require security and 
don't have to worry about NATs or firewalls in your network paths, then 
UDP is fine (if I don't think about the re-inveted reliability). I don't 
mind anyone using it where it actually works.

I'm genuinely worried though that Outbound with UDP won't actually work 
in the field once finished, because of something we fail to think of at 
this time.

If anyone can tell me of some other UDP based network application that 
actually attempts to work through NATs by using keepalives (that means: 
not DNS), and without users configuring their NAT with port forwards or 
similar then please tell me so that I don't have to worry that we're 
spending all this energy chasing a solution that might not even exist in 
the first place. There must be some kind of precedence in this area, I'm 
sure.

/Fredrik


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From sip-bounces@ietf.org Wed Dec 13 11:52:43 2006
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Date: Wed, 13 Dec 2006 08:51:02 -0800
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To: Fredrik Thulin <ft@it.su.se>
Subject: Re: [Sip] I-D: Preventing Fragmentation forClientInitiated	Connections
	in the Session Initiation Protocol (SIP)
References: <8B1D53AEF7B03449A6D3771B3B7F850F031D3DC4@esebe103.NOE.Nokia.com>
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Fredrik Thulin wrote:
> Marc Petit-Huguenin wrote:
> ...
>> So this is why I am switching my focus from performance to 1) do not
>> prevent implementers to use UDP if they need it and 2) anyway RFC 3261
>> mandates UDP AND TCP on all elements.
> 
> Not even RFC3261 manages to get you everything if using UDP though -
> large messages for one, security (SIPS) for another.

Let's exclude SIPS from the discussion, as DTLS can be used here.

Yes, large messages must go through TCP, but the reality is that not all
messages mandates to use TCP, and this is especially true for messages
that are sent periodically, REGISTER and SUBSCRIBE for example.

RFC 3261 already can mix UDP and TCP when there is no NAT[1].  The
problem here is that it not longer works behind a NAT, even with
Outbound.  My proposal is to extend Outbound so it works the same way.

> 
> As I see it, it is too complicated to try and make SIP over UDP work
> across NATs to even try it. For those cases where the properties of
> Outbound are desired, everyone will be better off using TCP.
> 
> For the cases where you have small messages, don't require security and
> don't have to worry about NATs or firewalls in your network paths, then
> UDP is fine (if I don't think about the re-inveted reliability). I don't
> mind anyone using it where it actually works.
> 
> I'm genuinely worried though that Outbound with UDP won't actually work
> in the field once finished, because of something we fail to think of at
> this time.

Outbound with UDP does not work.  Because of this some people want to
exclude UDP from Outbound, which is inconvenient for implementers who
care about performance.   What I am proposing is to make UDP working
with Outbound instead of excluding it.

Remember that this is not the endpoint that decide to use UDP or TCP,
its the servers (i.e. NAPTR records as described by RFC 3263).  If the
server decides that the endpoint must use TCP, then everything is fine
with Outbound.  But if the server decides that the endpoint must use UDP
(and switch to TCP if the message is too big), then things breaks for
the cases described in my draft, i.e. incoming messages too big and
outgoing responses too big.

> 
> If anyone can tell me of some other UDP based network application that
> actually attempts to work through NATs by using keepalives (that means:
> not DNS), and without users configuring their NAT with port forwards or
> similar then please tell me so that I don't have to worry that we're
> spending all this energy chasing a solution that might not even exist in
> the first place. There must be some kind of precedence in this area, I'm
> sure.
> 
> /Fredrik
> 
> 

[1] Excepted for the fragmentation response problem. See:
http://www.ietf.org/internet-drafts/draft-petithuguenin-sip-fragmentation-responses-01.txt
http://www.ietf.org/internet-drafts/draft-gurbani-sip-large-udp-response-00.txt

-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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Subject: Re: [Sip] I-D: Preventing Fragmentation
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On Wed, 2006-12-13 at 07:26 -0800, Marc Petit-Huguenin wrote:
> > b) The only reason for Outbound to support UDP would be its
> >    full compatibility to RFC 3261.
> 
> This is a good enough reason.
> 
> > 
> > If this is the case, I would ask WG to consider this reasoning
> > very carefully and if it makes any sense or not. To me it sounds
> > like: "UDP is actually not needed but because the WG decided to
> > mandate it a few years ago, it just has to be there."
> > 
> > According to your I-D the client has to anyway support TCP (or
> > other congestion controlled transport) so it would not even allow
> > to implement UDP-only clients (which might be desirable for
> > some vendors that currently only support UDP). Thus what is
> > the point with the UDP flows ? Why could they not be optional
> > for an UA ?
> 
> UDP-only clients are not SIP compliant.  So how a UDP flow can be
> optional if UDP in the endpoint is not?

I think it is perfectly reasonable to say that a new enhancement to SIP
(eg Outbound) is defined to work only over certain transports, if doing
so is either required to make it work at all, to limit the complexity of
the specification, or to make the resulting implementations more robust
in the wild.

-- 
Scott Lawrence  tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com
  sipXpbx project coordinator - SIPfoundry    http://www.sipfoundry.org/sipX
  Chief Technology Officer    - Pingtel Corp. http://www.pingtel.com/


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From sip-bounces@ietf.org Wed Dec 13 18:21:41 2006
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Hi Juha,

I proposed that since edge proxies and registrars have to support both
UDP and TCP, UAs that want to implement outbound for TCP don't have to
also implement outbound for UDP.  My proposal does not prevent
UDP-only devices from working.

I would also like to point out that devices that don't support TCP are
not compliant with RFC 3261.  I suspect one of the reasons these
devices have not implemented TCP is because the devices you mentioned
are usually used in a residential setting and therefore need a
standardized mechanism to get inbound requests when they are behind a
NAT.  This was the original intent of outbound and the title of the
document reflects that: " Managing Client Initiated *Connections*".
There are clearly people who want to implement this spec for TCP, so I
don't see what the problem is with finally doing something that makes
it so we can have interoperable implementations that use TCP behind
NATs.

thanks,
-rohan


On 12/12/06, Juha Heinanen <jh@tutpro.com> wrote:
> just a reality check.  yesterday when i was going to make some sip tcp
> tests, i tried to find a sip phone that supports tcp.  the most popular
> ones from linksys/sipura (cisco) and grandstream didn't.
>
> -- juha
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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Marc,

To date, I think you are the only person who has spoken on the list
who actually wants to use TCP for some requests and UDP for others.

What I have heard from the UDP-apologist crowd is generally that they
are using UDP for either historical reasons (with which I sympathize)
or for performance reasons (of which I am skeptical).  The historical
argument basically boils down to having a large number of components
that will all would need to be replaced or drastically reconfigured at
once.  I think the language I proposed explains and makes room for
this.

On the performance side, I consistently hear several UDP-apologists
concerned with the maximum number of connections they can handle.
Your approach does nothing to address this concern since it ends up in
the same number of TCP connections as the pure-TCP approach.  From the
TCP-advocates I hear that in their implementations they can handle a
very large number of TCP connections by getting rid of poll/select,
and that they can handle about the same number of UAs using UDP as
TCP.  The performance bottleneck I usually hear about with these
implementations is access to the registration/location server
database.  (I have also heard about the registration/location server
database being a bottleneck in UDP-only implementations as well.) Your
approach would effectively double the number of entries in the
registration database and therefore seems to make things worse rather
than better.  Certainly it does not address the perceived performance
problems of either the pro-UDP or the pro-TCP crowd.

thanks,
-rohan

On 12/13/06, Marc Petit-Huguenin <petithug@acm.org> wrote:
> Fredrik Thulin wrote:
> > Marc Petit-Huguenin wrote:
> > ...
> >> So this is why I am switching my focus from performance to 1) do not
> >> prevent implementers to use UDP if they need it and 2) anyway RFC 3261
> >> mandates UDP AND TCP on all elements.
> >
> > Not even RFC3261 manages to get you everything if using UDP though -
> > large messages for one, security (SIPS) for another.
>
> Let's exclude SIPS from the discussion, as DTLS can be used here.
>
> Yes, large messages must go through TCP, but the reality is that not all
> messages mandates to use TCP, and this is especially true for messages
> that are sent periodically, REGISTER and SUBSCRIBE for example.
>
> RFC 3261 already can mix UDP and TCP when there is no NAT[1].  The
> problem here is that it not longer works behind a NAT, even with
> Outbound.  My proposal is to extend Outbound so it works the same way.
>
> >
> > As I see it, it is too complicated to try and make SIP over UDP work
> > across NATs to even try it. For those cases where the properties of
> > Outbound are desired, everyone will be better off using TCP.
> >
> > For the cases where you have small messages, don't require security and
> > don't have to worry about NATs or firewalls in your network paths, then
> > UDP is fine (if I don't think about the re-inveted reliability). I don't
> > mind anyone using it where it actually works.
> >
> > I'm genuinely worried though that Outbound with UDP won't actually work
> > in the field once finished, because of something we fail to think of at
> > this time.
>
> Outbound with UDP does not work.  Because of this some people want to
> exclude UDP from Outbound, which is inconvenient for implementers who
> care about performance.   What I am proposing is to make UDP working
> with Outbound instead of excluding it.
>
> Remember that this is not the endpoint that decide to use UDP or TCP,
> its the servers (i.e. NAPTR records as described by RFC 3263).  If the
> server decides that the endpoint must use TCP, then everything is fine
> with Outbound.  But if the server decides that the endpoint must use UDP
> (and switch to TCP if the message is too big), then things breaks for
> the cases described in my draft, i.e. incoming messages too big and
> outgoing responses too big.
>
> >
> > If anyone can tell me of some other UDP based network application that
> > actually attempts to work through NATs by using keepalives (that means:
> > not DNS), and without users configuring their NAT with port forwards or
> > similar then please tell me so that I don't have to worry that we're
> > spending all this energy chasing a solution that might not even exist in
> > the first place. There must be some kind of precedence in this area, I'm
> > sure.
> >
> > /Fredrik
> >
> >
>
> [1] Excepted for the fragmentation response problem. See:
> http://www.ietf.org/internet-drafts/draft-petithuguenin-sip-fragmentation-responses-01.txt
> http://www.ietf.org/internet-drafts/draft-gurbani-sip-large-udp-response-00.txt
>
> --
> Marc Petit-Huguenin           [                                 ]
> Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
> Work: marc@8x8.com            [                                 ]
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
>

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> On the performance side, I consistently hear several UDP-apologists
> concerned with the maximum number of connections they can handle.
> Your approach does nothing to address this concern since it ends up in
> the same number of TCP connections as the pure-TCP approach.  From the
> TCP-advocates I hear that in their implementations they can handle a
> very large number of TCP connections by getting rid of poll/select,
> and that they can handle about the same number of UAs using UDP as
> TCP.  The performance bottleneck I usually hear about with these
> implementations is access to the registration/location server
> database.  (I have also heard about the registration/location server
> database being a bottleneck in UDP-only implementations as well.) Your
> approach would effectively double the number of entries in the
> registration database and therefore seems to make things worse rather
> than better.  Certainly it does not address the perceived performance
> problems of either the pro-UDP or the pro-TCP crowd.

No, this is not how it would work.   The TCP connections are temporary,
not permanent.

I think that we can compare this to a HTTP server.  Even if I am
constantly logged in slashdot.org, there is no permanent TCP connection
between my browser and slashdot.org's HTTP server.  This is because it
is more efficient to close the connection at the end of the transaction,
and to open a new one when I click on reload.

So there is 3 possible combinations of transports for SIP; UDP alone,
permanent TCP and UDP + temporary TCP.  I excluded UDP alone, because of
the fragmentation problem. So we wrote two tests with the same kind of
transactions an endpoints would do, the first test with a permanent TCP
connection, and the second test with a mix of UDP and temporary TCP.
The TCP/UDP mixed test, even if it have to create TCP connections and
close them each time a SIP message bigger than the MTU have to be sent
or received, was using half the resources of the test with the permanent
TCP connection.  And the UDP+TCP test was using the additional mechanism
to request the opening of the TCP connection.

So the idea is that at a specific time, a specific flow can use UDP or
TCP but not both.  The transport for the flow changes depending on what
needs to be used for incoming request or outgoing response.  After the
end of the transaction that needed TCP, the TCP flows is replaced by an
UDP flow and the TCP connection is closed.

Obviously this works only if the transaction that create the flow, i.e.
the REGISTER, does not need TCP.  If it is the case, then the only
transactions that need TCP are INVITE that carries a big SDP (thanks to
ICE) and the NOTIFY that carries big XML documents.

> 
> thanks,
> -rohan
> 
> On 12/13/06, Marc Petit-Huguenin <petithug@acm.org> wrote:
>> Fredrik Thulin wrote:
>> > Marc Petit-Huguenin wrote:
>> > ...
>> >> So this is why I am switching my focus from performance to 1) do not
>> >> prevent implementers to use UDP if they need it and 2) anyway RFC 3261
>> >> mandates UDP AND TCP on all elements.
>> >
>> > Not even RFC3261 manages to get you everything if using UDP though -
>> > large messages for one, security (SIPS) for another.
>>
>> Let's exclude SIPS from the discussion, as DTLS can be used here.
>>
>> Yes, large messages must go through TCP, but the reality is that not all
>> messages mandates to use TCP, and this is especially true for messages
>> that are sent periodically, REGISTER and SUBSCRIBE for example.
>>
>> RFC 3261 already can mix UDP and TCP when there is no NAT[1].  The
>> problem here is that it not longer works behind a NAT, even with
>> Outbound.  My proposal is to extend Outbound so it works the same way.
>>
>> >
>> > As I see it, it is too complicated to try and make SIP over UDP work
>> > across NATs to even try it. For those cases where the properties of
>> > Outbound are desired, everyone will be better off using TCP.
>> >
>> > For the cases where you have small messages, don't require security and
>> > don't have to worry about NATs or firewalls in your network paths, then
>> > UDP is fine (if I don't think about the re-inveted reliability). I
>> don't
>> > mind anyone using it where it actually works.
>> >
>> > I'm genuinely worried though that Outbound with UDP won't actually work
>> > in the field once finished, because of something we fail to think of at
>> > this time.
>>
>> Outbound with UDP does not work.  Because of this some people want to
>> exclude UDP from Outbound, which is inconvenient for implementers who
>> care about performance.   What I am proposing is to make UDP working
>> with Outbound instead of excluding it.
>>
>> Remember that this is not the endpoint that decide to use UDP or TCP,
>> its the servers (i.e. NAPTR records as described by RFC 3263).  If the
>> server decides that the endpoint must use TCP, then everything is fine
>> with Outbound.  But if the server decides that the endpoint must use UDP
>> (and switch to TCP if the message is too big), then things breaks for
>> the cases described in my draft, i.e. incoming messages too big and
>> outgoing responses too big.
>>
>> >
>> > If anyone can tell me of some other UDP based network application that
>> > actually attempts to work through NATs by using keepalives (that means:
>> > not DNS), and without users configuring their NAT with port forwards or
>> > similar then please tell me so that I don't have to worry that we're
>> > spending all this energy chasing a solution that might not even
>> exist in
>> > the first place. There must be some kind of precedence in this area,
>> I'm
>> > sure.
>> >
>> > /Fredrik
>> >
>> >
>>
>> [1] Excepted for the fragmentation response problem. See:
>> http://www.ietf.org/internet-drafts/draft-petithuguenin-sip-fragmentation-responses-01.txt
>>
>> http://www.ietf.org/internet-drafts/draft-gurbani-sip-large-udp-response-00.txt

-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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From sip-bounces@ietf.org Wed Dec 13 20:03:23 2006
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Marc Petit-Huguenin wrote:
>> On the performance side, I consistently hear several UDP-apologists
>> concerned with the maximum number of connections they can handle.
>> Your approach does nothing to address this concern since it ends up in
>> the same number of TCP connections as the pure-TCP approach.  From the
>> TCP-advocates I hear that in their implementations they can handle a
>> very large number of TCP connections by getting rid of poll/select,
>> and that they can handle about the same number of UAs using UDP as
>> TCP.  The performance bottleneck I usually hear about with these
>> implementations is access to the registration/location server
>> database.  (I have also heard about the registration/location server
>> database being a bottleneck in UDP-only implementations as well.) Your
>> approach would effectively double the number of entries in the
>> registration database and therefore seems to make things worse rather
>> than better.  Certainly it does not address the perceived performance
>> problems of either the pro-UDP or the pro-TCP crowd.
>>     
>
> No, this is not how it would work.   The TCP connections are temporary,
> not permanent.
>
> I think that we can compare this to a HTTP server.  Even if I am
> constantly logged in slashdot.org, there is no permanent TCP connection
> between my browser and slashdot.org's HTTP server.  This is because it
> is more efficient to close the connection at the end of the transaction,
> and to open a new one when I click on reload.
>
> So there is 3 possible combinations of transports for SIP; UDP alone,
> permanent TCP and UDP + temporary TCP.  I excluded UDP alone, because of
> the fragmentation problem. So we wrote two tests with the same kind of
> transactions an endpoints would do, the first test with a permanent TCP
> connection, and the second test with a mix of UDP and temporary TCP.
> The TCP/UDP mixed test, even if it have to create TCP connections and
> close them each time a SIP message bigger than the MTU have to be sent
> or received, was using half the resources of the test with the permanent
> TCP connection.  And the UDP+TCP test was using the additional mechanism
> to request the opening of the TCP connection.
>   

Doesn't this all go down the drain when you add (D)TLS to the mix? The
relative costs of establishment are orders of magnitude different right?
(I wish I had that number off the top of my head, but I don't) That's
really the thing that's swayed me with these debates since we shouldn't
be counting on non-use of crypto going forward.

       Mike

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From sip-bounces@ietf.org Wed Dec 13 22:17:34 2006
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Michael Thomas wrote:
> Marc Petit-Huguenin wrote:
>>
>> So there is 3 possible combinations of transports for SIP; UDP alone,
>> permanent TCP and UDP + temporary TCP.  I excluded UDP alone, because of
>> the fragmentation problem. So we wrote two tests with the same kind of
>> transactions an endpoints would do, the first test with a permanent TCP
>> connection, and the second test with a mix of UDP and temporary TCP.
>> The TCP/UDP mixed test, even if it have to create TCP connections and
>> close them each time a SIP message bigger than the MTU have to be sent
>> or received, was using half the resources of the test with the permanent
>> TCP connection.  And the UDP+TCP test was using the additional mechanism
>> to request the opening of the TCP connection.
>>   
> 
> Doesn't this all go down the drain when you add (D)TLS to the mix? The
> relative costs of establishment are orders of magnitude different right?
> (I wish I had that number off the top of my head, but I don't) That's
> really the thing that's swayed me with these debates since we shouldn't
> be counting on non-use of crypto going forward.
> 

This is a good point.  I'll try to find an answer to this.

Thanks.


-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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From sip-bounces@ietf.org Thu Dec 14 00:32:23 2006
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Rohan Mahy writes:

 > I proposed that since edge proxies and registrars have to support both
 > UDP and TCP, UAs that want to implement outbound for TCP don't have to
 > also implement outbound for UDP.  My proposal does not prevent
 > UDP-only devices from working.

yes, i didn't actually comment on your proposal, but just how far from
rfc3261 requirements the real world is.

 > There are clearly people who want to implement this spec for TCP, so I
 > don't see what the problem is with finally doing something that makes
 > it so we can have interoperable implementations that use TCP behind
 > NATs.

we already have that.  after i finally found a sip tcp implementation
(in nokia phone), i made the phone to register from behind symmetric NAT
with my OpenSER based proxy.  the proxy kept tcp connection persistently
open and i didn't have any trouble of making both in and outbound calls
from that phone.

no non-rfc3261 compliant features were in use in the phone.  thus i
still fail to see what value add would outbound give now when its REAL
value add has been removed, i.e., mandating the phone to register with
more than one proxy.

-- juha

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Thanks Pavel,

ext Dostal, Pavel wrote:
> Jari,
>
> In chapter 4.3.1 In my opinion the "path" parameter shall be stated as
> mandatory.
>
>   
right, it should be clarified.
> In chapter 4.6.3 the same handling is defined for selector locating
> non-existing element and multiple elements. I'd rather distinguish these
> cases as subscription to non-existing element may have real usage (as
> stated in chapter 4.1), but subscription to multiple elements (in the
> sense of one selector for multiple elements) shall not be possible at
> all.
>
> Regards,
>
> Pavel
>
>   
right, the intention is to allow subscription to an non-existing element 
(as it is stated). However,  do you propose  something along the lines: 
exists = "single", "none" or "multiple" ? So none and multiple would be 
an "error" in XCAP sense where a unique node is always mandatory for a 
successful operation. Having these two separate states might be better, 
i.e. the client would probably know to change the request. Also it would 
be better in-line with XCAP where 404 and 409 error responses separate 
these cases. However, the initial subscription must then signal this 
multiple case, I suppose. So it would make the handling a bit more 
complex, but more deterministic.

Comments ?
--Jari


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Subject: RE: [Sip] outbound-06 submitted
Date: Thu, 14 Dec 2006 09:28:17 +0100
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From: "Christer Holmberg \(JO/LMF\)" <christer.holmberg@ericsson.com>
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Hi,

I see no reason why the UA should need to send STUN BEFORE the REGISTER,
but I agree it should do it very soon after it has received the REGISTER
response - which may include information whether the edge proxy supports
STUN in the first place.

Regards,

Christer =20

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]=20
> Sent: 13. joulukuuta 2006 1:51
> To: Byron Campen
> Cc: SIP WG; Cullen Jennings; Rohan Mahy
> Subject: Re: [Sip] outbound-06 submitted
>=20
> Hi Byron,
>=20
> We could add text recommending that the UA compare the=20
> rport/received with the XOR-MAPPED-ADDRESS.  That would=20
> appear to solve the problem.
> Can you think of any problem with this?
>=20
> thanks,
> -rohan
>=20
> On 12/12/06, Byron Campen <bcampen@estacado.net> wrote:
> >         I have a question about the usage of STUN as a=20
> keepalive over UDP.
> > Are clients supposed to start sending STUN before they try to=20
> > REGISTER? If not, I can see the scenario where a UA sends the=20
> > REGISTER, sets a timer (btw 24-29 seconds) for the first STUN=20
> > keepalive, but before that STUN request is sent, the pinhole closes.
> > Both ends will think the flow is ok (the STUN keepalive will open a=20
> > new pinhole, but the UA won't know that), and the mess won't get=20
> > cleaned up until the UA re-registers. Sending the keepalive "soon"
> > after the REGISTER would probably solve this problem also,=20
> if sending=20
> > before is not acceptable.
> >
> > Best regards,
> > Byron Campen
> >
> > > Hi,
> > >
> > > I submitted -06 of outbound so folks can at least take a=20
> look to see=20
> > > that the many small changes we have agreed to have been=20
> incorporated.
> > > I also included the proposed text for FASTEST, since the=20
> draft would=20
> > > need significant surgery anyway if we go with CLEANEST.
> > >
> > > Until it appears in the archive you can find it here:
> > > http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-
> > > outbound-06.html
> > > http://svn.resiprocate.org/rep/ietf-drafts/fluffy/draft-ietf-sip-
> > > outbound-06.txt
> > >
> > > thanks,
> > > -rohan
> > >
> > > _______________________________________________
> > > Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> > > This list is for NEW development of the core SIP Protocol Use=20
> > > sip-implementors@cs.columbia.edu for questions on current sip Use=20
> > > sipping@ietf.org for new developments on the application of sip
> >
> >
> >
> >
>=20
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol Use=20
> sip-implementors@cs.columbia.edu for questions on current sip=20
> Use sipping@ietf.org for new developments on the application of sip
>=20

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Only, be, found an ofwhereall these.
Your use aptget from malicious or exploited. This is production release. =
Document language report problem with, email.
Exploited mirrors soforth further planned back project page, also.
Exploited, mirrors, soforth, further.
Digitally signed ensure all files belong. It was on october th testing, =
packages that. Nihongo romacircn russkij, suomi svenska default =
document.
Into yet but they. Advantage more recent versions software see. Been =
accepted, into yet but they.
And how, becomes, etch unstable where. Other contact pagelast, modified =
thu nov utc copyright. States skip quicknav about news, getting support, =
developers corner.
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<DIV align=3Dcenter><FONT face=3DArial size=3D2><IMG alt=3D"recommend" =
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<DIV><FONT face=3DArial size=3D2>Only, be, found an ofwhereall =
these.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Your use aptget from malicious or =
exploited. This=20
is production release. Document language report problem with, =
email.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Exploited mirrors soforth further =
planned back=20
project page, also.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Exploited, mirrors, soforth, =
further.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Digitally signed ensure all files =
belong. It was on=20
october th testing, packages that. Nihongo romacircn russkij, suomi =
svenska=20
default document.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Into yet but they. Advantage more =
recent versions=20
software see. Been accepted, into yet but they.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>And how, becomes, etch unstable where. =
Other=20
contact pagelast, modified thu nov utc copyright. States skip quicknav =
about=20
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From sip-bounces@ietf.org Fri Dec 15 04:39:55 2006
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Thread-Topic: I-D ACTION:draft-fries-sip-identity-usage-bcp-01.txt 
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From: "Fries, Steffen" <steffen.fries@siemens.com>
To: "IETF SIP List" <sip@ietf.org>
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Subject: [Sip] FW: I-D ACTION:draft-fries-sip-identity-usage-bcp-01.txt 
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 Hi,

please find below the announcement for the updated draft on the SIP
Identity BCP describing the use of SIP Identity for the protection of
credential exchange between clients.
We updated the draft and discussed the use cases more elaborately and
also included the handling of the reverse direction using the connected
identity approach.

We would be interested if this work is seen as useful in the scope of
the SIP WG at is is right now or if additions are needed.

Regards
	Steffen



-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20
Sent: Monday, December 11, 2006 9:50 PM
To: i-d-announce@ietf.org
Subject: I-D ACTION:draft-fries-sip-identity-usage-bcp-01.txt=20

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


	Title		: SIP Identity Usage BCP
	Author(s)	: S. Fries, et al.
	Filename	: draft-fries-sip-identity-usage-bcp-01.txt
	Pages		: 11
	Date		: 2006-12-11
=09
This document describes a use case for RFC4474 (SIP Identity) for
   verifying a certificate that might not be publicly verifiable by
   traditional means.  It provides a best current practice document for
   binding an identity to a certificate for the duration of a session.
   The certificate may then be used to bootstrap further security
   parameters, e.g., for securing media data.  A discussion of possible
   enhancements is included in the appendix.  Editors Note: The first
   version of this draft was discussed in the SIPPING WG.  As the target
   of this draft is a BCP for current issues, the draft was updated and
   submitted to the SIP WG.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-fries-sip-identity-usage-bcp-0
1.txt

To remove yourself from the I-D Announcement list, send a message to=20
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logging in, type "cd internet-drafts" and then=20
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A list of Internet-Drafts directories can be found in
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=09
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	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.

Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

------_=_NextPart_001_01C7202C.CAD39695
Content-Type: application/octet-stream;
	name="draft-fries-sip-identity-usage-bcp-01.URL"
Content-Transfer-Encoding: base64
Content-Description: draft-fries-sip-identity-usage-bcp-01.URL
Content-Disposition: attachment;
	filename="draft-fries-sip-identity-usage-bcp-01.URL"

W0ludGVybmV0U2hvcnRjdXRdDQpVUkw9ZnRwOi8vZnRwLmlldGYub3JnL2ludGVybmV0LWRyYWZ0
cy9kcmFmdC1mcmllcy1zaXAtaWRlbnRpdHktdXNhZ2UtYmNwLTAxLnR4dA0K

------_=_NextPart_001_01C7202C.CAD39695
Content-Type: text/plain; charset="us-ascii"
MIME-Version: 1.0
Content-Transfer-Encoding: 7bit
Content-Disposition: inline

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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------_=_NextPart_001_01C7202C.CAD39695--




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Subject: Re: [Sip] Help: "About multiple early-session sdp body in a request"
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zhang.lu2@zte.com.cn wrote:
> 
> Dear Sip Group,
> 
>      There encounters some questions when I read RFC 3959 and RFC 3960 
> which refer to the Early-Session , that is

Note that you may be the first to consider implementing this. I've asked
around and haven't heard of an implementation. So answers are somewhat
speculative.

>        *1. "Is it available to include multiple early-session sdp body 
> in a request?"*

I think this is not entirely clear, but its not clear what it would mean
if you were to attempt it. Also, each offer must be matched to an
answer. If there were multiple early-session offers, then there would
need be multiple early session answers. In that case, there would need
to be some convention to decide which early answer goes with which early
offer.

Getting early-session to work at all is enough of a challenge. Getting
multiple early sessions to work seems like folly.

>     for example:
>         Content-Type: multipart/mixed; boundary="boundary1"
>         Content-Length: 401
> 
>         --boundary1
>            Content-Type: application/sdp
>            Content-Disposition: early-session
> 
>            v=0
>            o=Bob 2890844725 2890844725 IN IP4 host1.example.org
>            s=
>            c=IN IP4 192.0.2.2
>            t=0 0
>            m=audio 30000 RTP/AVP 0
> 
>        --boundary1
>           Content-Type: application/sdp
>           Content-Disposition: early-session
> 
>           v=0
>           o=Bob 2890844714 2890844714 IN IP4 host2.example.org
>           s=
>           c=IN IP4 192.0.2.3
>           t=0 0
>           m=audio 30002 RTP/AVP 0
> 
>        --boundary1--
> 
> 
>      *2. "How can a UAS decide to handle a early-session sdp ,and not to 
> handle other early-session sdps when     UAS receive a request with 
> multiple early-session sdps?"*

Well, that is an issue isn't it. It assumes that you don't want to
handle them all. This problem exists with early media in forking
situations without use of early-session, and in fact was one of the
reasons early-session was proposed. If you introduce multiple early
sessions, then in some sense the problem has been reintroduced.

>     This question arises from the following scene:
>     Before A and B establish communication£¬
>     (1) AS-1 wants to send media to B
>     (2) AS-2 wants to send media to A

Is this in the context of IMS?

To be able to do this, the AS's must be UAs. For the call flow you show,
both AS-1 and AS-2 must act as B2BUAs. To make sense of all this, and
achieve the sessions you show, I think they must have a lot of
understanding of what is going on in this particular case. In the
abstract I don't see how they could know that, but in a particular case
perhaps that is possible.

In the case you show it appears to me that you might never have to
include two early-session descriptions in the same message. Instead,
wouldn't one be removed, because it is being handled, and another added
to be handled on a different leg? But its hard to say for sure without a
concrete example of exactly what you are trying to do.

IMO you are on a dangerous course in uncharted waters. It would be
better if you found a more mainstream way to do what you want. Since
apparently you have a B2BUA at each step, I think you could probably
achieve what you want in a more straightforward way by carefully
managing the contents of standard offers and answers along the way.

	Paul

>     To reach the goal , some principles should obey:
>     (1)precondtion defined in RFC 3312
>     (2)Early-Session defined in RFC 3959, RFC 3960
> 
>     The red circles mark special processing.
> 
>    
>    
> 
> 
> 
> 
> 
> --------------------------------------------------------
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> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
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> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip

_______________________________________________
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This list is for NEW development of the core SIP Protocol
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From sip-bounces@ietf.org Fri Dec 15 13:53:30 2006
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Subject: [Sip] Keeping session data in SIP messages?
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Hello!

I am writing a load balancing SIP proxy and would like to keep it as
simple as possible, ie. I would like to avoid using any databases.

Currently I have come up with the following solution:

When first INVITE arrives, calculate branch parameter, determine the
server to forward the message to, store the relation between branch
and server name in database. When another message in this session
arrives (the same branch parameter of Via header) the server name is
looked up from database and the message is forwarded to that server.

Can I put some header or parameter for some header in the message
that client would be required to include in the next request message
it sends out during the session?

Thanks!

--
Ahti Legonkov






_______________________________________________
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This list is for NEW development of the core SIP Protocol
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Ahti Legonkov wrote:
> Hello!
> 
> I am writing a load balancing SIP proxy and would like to keep it as
> simple as possible, ie. I would like to avoid using any databases.
> 
> Currently I have come up with the following solution:
> 
> When first INVITE arrives, calculate branch parameter, determine the
> server to forward the message to, store the relation between branch
> and server name in database. When another message in this session
> arrives (the same branch parameter of Via header) the server name is
> looked up from database and the message is forwarded to that server.
> 
> Can I put some header or parameter for some header in the message
> that client would be required to include in the next request message
> it sends out during the session?

Its not clear to me why you need to do anything.

If the server is the UAS, then it will return a Contact header with the 
uri to be used for additional requests in the session. If your proxy is 
even involved in the subsequent messages, then it will simply forward to 
that address, which will be in the R-URI.

If the server you selected is a proxy, and if it wants to be involved in 
subsequent in-dialog requests, then it can Record-Route. And if *your* 
proxy wants to be in subsequent requests in the dialog, then it also 
should Record-Route.

What more is required here?

	Paul

> Thanks!
> 
> -- 
> Ahti Legonkov
> 
> 
> 
> 
> 
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip
> 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Subject: Next Big Winner
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Paul Kyzivat wrote:

> If the server you selected is a proxy, and if it wants to be involved in 
> subsequent in-dialog requests, then it can Record-Route. And if *your* 
> proxy wants to be in subsequent requests in the dialog, then it also 
> should Record-Route.

There had to be a simple and elegant solutiond and this is it. Thanks!

--
Ahti Legonkov


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From: "Rohan Mahy" <rohan.mahy@gmail.com>
To: "Marc Petit-Huguenin" <petithug@acm.org>
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Hi Marc,

What is the edge proxy supposed to do when it wants to deliver a large
INVITE request to the UA and the UA does not have a TCP-connection
established at the moment?

thanks,
-rohan

On 12/13/06, Marc Petit-Huguenin <petithug@acm.org> wrote:
> > On the performance side, I consistently hear several UDP-apologists
> > concerned with the maximum number of connections they can handle.
> > Your approach does nothing to address this concern since it ends up in
> > the same number of TCP connections as the pure-TCP approach.  From the
> > TCP-advocates I hear that in their implementations they can handle a
> > very large number of TCP connections by getting rid of poll/select,
> > and that they can handle about the same number of UAs using UDP as
> > TCP.  The performance bottleneck I usually hear about with these
> > implementations is access to the registration/location server
> > database.  (I have also heard about the registration/location server
> > database being a bottleneck in UDP-only implementations as well.) Your
> > approach would effectively double the number of entries in the
> > registration database and therefore seems to make things worse rather
> > than better.  Certainly it does not address the perceived performance
> > problems of either the pro-UDP or the pro-TCP crowd.
>
> No, this is not how it would work.   The TCP connections are temporary,
> not permanent.
>
> I think that we can compare this to a HTTP server.  Even if I am
> constantly logged in slashdot.org, there is no permanent TCP connection
> between my browser and slashdot.org's HTTP server.  This is because it
> is more efficient to close the connection at the end of the transaction,
> and to open a new one when I click on reload.
>
> So there is 3 possible combinations of transports for SIP; UDP alone,
> permanent TCP and UDP + temporary TCP.  I excluded UDP alone, because of
> the fragmentation problem. So we wrote two tests with the same kind of
> transactions an endpoints would do, the first test with a permanent TCP
> connection, and the second test with a mix of UDP and temporary TCP.
> The TCP/UDP mixed test, even if it have to create TCP connections and
> close them each time a SIP message bigger than the MTU have to be sent
> or received, was using half the resources of the test with the permanent
> TCP connection.  And the UDP+TCP test was using the additional mechanism
> to request the opening of the TCP connection.
>
> So the idea is that at a specific time, a specific flow can use UDP or
> TCP but not both.  The transport for the flow changes depending on what
> needs to be used for incoming request or outgoing response.  After the
> end of the transaction that needed TCP, the TCP flows is replaced by an
> UDP flow and the TCP connection is closed.
>
> Obviously this works only if the transaction that create the flow, i.e.
> the REGISTER, does not need TCP.  If it is the case, then the only
> transactions that need TCP are INVITE that carries a big SDP (thanks to
> ICE) and the NOTIFY that carries big XML documents.
>
> >
> > thanks,
> > -rohan
> >
> > On 12/13/06, Marc Petit-Huguenin <petithug@acm.org> wrote:
> >> Fredrik Thulin wrote:
> >> > Marc Petit-Huguenin wrote:
> >> > ...
> >> >> So this is why I am switching my focus from performance to 1) do not
> >> >> prevent implementers to use UDP if they need it and 2) anyway RFC 3261
> >> >> mandates UDP AND TCP on all elements.
> >> >
> >> > Not even RFC3261 manages to get you everything if using UDP though -
> >> > large messages for one, security (SIPS) for another.
> >>
> >> Let's exclude SIPS from the discussion, as DTLS can be used here.
> >>
> >> Yes, large messages must go through TCP, but the reality is that not all
> >> messages mandates to use TCP, and this is especially true for messages
> >> that are sent periodically, REGISTER and SUBSCRIBE for example.
> >>
> >> RFC 3261 already can mix UDP and TCP when there is no NAT[1].  The
> >> problem here is that it not longer works behind a NAT, even with
> >> Outbound.  My proposal is to extend Outbound so it works the same way.
> >>
> >> >
> >> > As I see it, it is too complicated to try and make SIP over UDP work
> >> > across NATs to even try it. For those cases where the properties of
> >> > Outbound are desired, everyone will be better off using TCP.
> >> >
> >> > For the cases where you have small messages, don't require security and
> >> > don't have to worry about NATs or firewalls in your network paths, then
> >> > UDP is fine (if I don't think about the re-inveted reliability). I
> >> don't
> >> > mind anyone using it where it actually works.
> >> >
> >> > I'm genuinely worried though that Outbound with UDP won't actually work
> >> > in the field once finished, because of something we fail to think of at
> >> > this time.
> >>
> >> Outbound with UDP does not work.  Because of this some people want to
> >> exclude UDP from Outbound, which is inconvenient for implementers who
> >> care about performance.   What I am proposing is to make UDP working
> >> with Outbound instead of excluding it.
> >>
> >> Remember that this is not the endpoint that decide to use UDP or TCP,
> >> its the servers (i.e. NAPTR records as described by RFC 3263).  If the
> >> server decides that the endpoint must use TCP, then everything is fine
> >> with Outbound.  But if the server decides that the endpoint must use UDP
> >> (and switch to TCP if the message is too big), then things breaks for
> >> the cases described in my draft, i.e. incoming messages too big and
> >> outgoing responses too big.
> >>
> >> >
> >> > If anyone can tell me of some other UDP based network application that
> >> > actually attempts to work through NATs by using keepalives (that means:
> >> > not DNS), and without users configuring their NAT with port forwards or
> >> > similar then please tell me so that I don't have to worry that we're
> >> > spending all this energy chasing a solution that might not even
> >> exist in
> >> > the first place. There must be some kind of precedence in this area,
> >> I'm
> >> > sure.
> >> >
> >> > /Fredrik
> >> >
> >> >
> >>
> >> [1] Excepted for the fragmentation response problem. See:
> >> http://www.ietf.org/internet-drafts/draft-petithuguenin-sip-fragmentation-responses-01.txt
> >>
> >> http://www.ietf.org/internet-drafts/draft-gurbani-sip-large-udp-response-00.txt
>
> --
> Marc Petit-Huguenin           [                                 ]
> Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
> Work: marc@8x8.com            [                                 ]
>

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From sip-bounces@ietf.org Fri Dec 15 21:04:37 2006
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	in the Session Initiation Protocol (SIP)
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Rohan Mahy wrote:
> Hi Marc,
> 
> What is the edge proxy supposed to do when it wants to deliver a large
> INVITE request to the UA and the UA does not have a TCP-connection
> established at the moment?

As I proposed in my draft[1], the edge proxy sends a STUN ForceCCP[2]
request over UDP to the UA to replace the flow by a TCP flow.  The edge
proxy can then use the TCP connection to send the INVITE.  The TCP flow
is replaced by an UDP flow at the end of the SIP transaction, or after 5
seconds if there is no SIP transaction.  The TCP connection is then closed.

Note that the draft is missing lot of stuff, e.g. how an edge proxy
knows that the endpoint supports this, etc...


[1]
http://tools.ietf.org/html/draft-petithuguenin-sip-outbound-fragmentation-00#section-6
[2] Force Congestion Controlled Transport Protocol

> 
> thanks,
> -rohan
> 
> On 12/13/06, Marc Petit-Huguenin <petithug@acm.org> wrote:
>> > On the performance side, I consistently hear several UDP-apologists
>> > concerned with the maximum number of connections they can handle.
>> > Your approach does nothing to address this concern since it ends up in
>> > the same number of TCP connections as the pure-TCP approach.  From the
>> > TCP-advocates I hear that in their implementations they can handle a
>> > very large number of TCP connections by getting rid of poll/select,
>> > and that they can handle about the same number of UAs using UDP as
>> > TCP.  The performance bottleneck I usually hear about with these
>> > implementations is access to the registration/location server
>> > database.  (I have also heard about the registration/location server
>> > database being a bottleneck in UDP-only implementations as well.) Your
>> > approach would effectively double the number of entries in the
>> > registration database and therefore seems to make things worse rather
>> > than better.  Certainly it does not address the perceived performance
>> > problems of either the pro-UDP or the pro-TCP crowd.
>>
>> No, this is not how it would work.   The TCP connections are temporary,
>> not permanent.
>>
>> I think that we can compare this to a HTTP server.  Even if I am
>> constantly logged in slashdot.org, there is no permanent TCP connection
>> between my browser and slashdot.org's HTTP server.  This is because it
>> is more efficient to close the connection at the end of the transaction,
>> and to open a new one when I click on reload.
>>
>> So there is 3 possible combinations of transports for SIP; UDP alone,
>> permanent TCP and UDP + temporary TCP.  I excluded UDP alone, because of
>> the fragmentation problem. So we wrote two tests with the same kind of
>> transactions an endpoints would do, the first test with a permanent TCP
>> connection, and the second test with a mix of UDP and temporary TCP.
>> The TCP/UDP mixed test, even if it have to create TCP connections and
>> close them each time a SIP message bigger than the MTU have to be sent
>> or received, was using half the resources of the test with the permanent
>> TCP connection.  And the UDP+TCP test was using the additional mechanism
>> to request the opening of the TCP connection.
>>
>> So the idea is that at a specific time, a specific flow can use UDP or
>> TCP but not both.  The transport for the flow changes depending on what
>> needs to be used for incoming request or outgoing response.  After the
>> end of the transaction that needed TCP, the TCP flows is replaced by an
>> UDP flow and the TCP connection is closed.
>>
>> Obviously this works only if the transaction that create the flow, i.e.
>> the REGISTER, does not need TCP.  If it is the case, then the only
>> transactions that need TCP are INVITE that carries a big SDP (thanks to
>> ICE) and the NOTIFY that carries big XML documents.
>>
>> >
>> > thanks,
>> > -rohan
>> >
>> > On 12/13/06, Marc Petit-Huguenin <petithug@acm.org> wrote:
>> >> Fredrik Thulin wrote:
>> >> > Marc Petit-Huguenin wrote:
>> >> > ...
>> >> >> So this is why I am switching my focus from performance to 1) do
>> not
>> >> >> prevent implementers to use UDP if they need it and 2) anyway
>> RFC 3261
>> >> >> mandates UDP AND TCP on all elements.
>> >> >
>> >> > Not even RFC3261 manages to get you everything if using UDP though -
>> >> > large messages for one, security (SIPS) for another.
>> >>
>> >> Let's exclude SIPS from the discussion, as DTLS can be used here.
>> >>
>> >> Yes, large messages must go through TCP, but the reality is that
>> not all
>> >> messages mandates to use TCP, and this is especially true for messages
>> >> that are sent periodically, REGISTER and SUBSCRIBE for example.
>> >>
>> >> RFC 3261 already can mix UDP and TCP when there is no NAT[1].  The
>> >> problem here is that it not longer works behind a NAT, even with
>> >> Outbound.  My proposal is to extend Outbound so it works the same way.
>> >>
>> >> >
>> >> > As I see it, it is too complicated to try and make SIP over UDP work
>> >> > across NATs to even try it. For those cases where the properties of
>> >> > Outbound are desired, everyone will be better off using TCP.
>> >> >
>> >> > For the cases where you have small messages, don't require
>> security and
>> >> > don't have to worry about NATs or firewalls in your network
>> paths, then
>> >> > UDP is fine (if I don't think about the re-inveted reliability). I
>> >> don't
>> >> > mind anyone using it where it actually works.
>> >> >
>> >> > I'm genuinely worried though that Outbound with UDP won't
>> actually work
>> >> > in the field once finished, because of something we fail to think
>> of at
>> >> > this time.
>> >>
>> >> Outbound with UDP does not work.  Because of this some people want to
>> >> exclude UDP from Outbound, which is inconvenient for implementers who
>> >> care about performance.   What I am proposing is to make UDP working
>> >> with Outbound instead of excluding it.
>> >>
>> >> Remember that this is not the endpoint that decide to use UDP or TCP,
>> >> its the servers (i.e. NAPTR records as described by RFC 3263).  If the
>> >> server decides that the endpoint must use TCP, then everything is fine
>> >> with Outbound.  But if the server decides that the endpoint must
>> use UDP
>> >> (and switch to TCP if the message is too big), then things breaks for
>> >> the cases described in my draft, i.e. incoming messages too big and
>> >> outgoing responses too big.
>> >>
>> >> >
>> >> > If anyone can tell me of some other UDP based network application
>> that
>> >> > actually attempts to work through NATs by using keepalives (that
>> means:
>> >> > not DNS), and without users configuring their NAT with port
>> forwards or
>> >> > similar then please tell me so that I don't have to worry that we're
>> >> > spending all this energy chasing a solution that might not even
>> >> exist in
>> >> > the first place. There must be some kind of precedence in this area,
>> >> I'm
>> >> > sure.
>> >> >
>> >> > /Fredrik
>> >> >
>> >> >
>> >>
>> >> [1] Excepted for the fragmentation response problem. See:
>> >>
>> http://www.ietf.org/internet-drafts/draft-petithuguenin-sip-fragmentation-responses-01.txt
>>
>> >>
>> >>
>> http://www.ietf.org/internet-drafts/draft-gurbani-sip-large-udp-response-00.txt
>>


-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [ RFC1855-compliant space for rent]
Work: marc@8x8.com            [                                 ]

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From sip-bounces@ietf.org Sat Dec 16 15:35:10 2006
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Practically speaking given where we are today, I would be a fan of a  
solution that meant the following goals:

1) allowed deployments that wanted to use TCP to use TCP

2) allowed deployments that wanted to use UDP to use only UDP with  
the known limitations that the UA would not be able to send large  
requests

3) allowed deployments that wanted to use both UDP and TCP to do so  
and not have the size limitations by using switchover

(note it would be easy to  meet the above goals using any type of  
keep alive so they do no impact if we use CRLF or something else)

Now I know there are people on the list that only care about one of  
these three goals but I know there are some people that want each  
one. I think that if we had some agreement that we were trying to  
meet these goals, then it would be easier for Rohan and I to help  
make progress on a proposal that everyone found agreeable. Perhaps  
Rohan's proposal meets these. I understand Rohan is making the  
argument that 2 is not needed because 3261 does not allow 2 - some  
people disagree with this regardless it is is clear that the bulk of  
deployments today are in category 2 for a variety of good reasons.

I just don't want to hold up outbound with figuring out the problems  
of TCP vs. UDP. Or figuring out how to do high reliability in SIP  
proxy farms. Or figuring out how to exactly build very scaleable  
proxy farms. Or figuring out how to deal with some security problems  
with changing IPs on UAs. All of these things are separable from  
outbound and though work is needed on them, I think we can finish  
this work without solving all o f them. I'd rather not spend endless  
time discussing if outbound should only work for IPv6 to help  
encourage people to move to v6 but instead try and find something  
that we could all live with and meant the needs of at least a bunch  
of deployments.

So if we can agree on above sort of goals, I think it will help me  
understand how to come up with some proposal that take into account   
what people are saying on this poll.

more inline...

On Dec 8, 2006, at 6:12 PM, Dean Willis wrote:

> Previously we had recorded consensus to use STUN for keepalive with  
> both TCP and UDP. This was not a perfect consensus (there were some  
> dissenters). There has been ongoing discussion notably instigated  
> by the current editor of the Outbound draft, that some people feel  
> may have resulted in a change of the consensus on this draft's  
> technical direction.
>
> So I'd like to ask people who care and think they understand the  
> differences to answer the following questions, on-list and on-the  
> record.
>
>
> Proposal 1: Make CRLF the only keepalive mechanism for TCP and TLS  
> over TCP flow (retain STUN for UDP flows).
>
> Yes (accept the proposal and use CRLF for TCP) or No (reject the  
> proposal and keep STUN for TCP)
>

I think we have heard from the people that think STUN is a good idea  
why they think STUN is a good idea. I think we previously came to the  
idea that the double CRLF was better than the CRLF so I would rather  
compare to double CRLF in the poll.

>
> Proposal 2: Make outbound registrations over UDP flows optional for  
> UAs.
>
> Yes (Accept the proposal) or No (Reject the proposal).

I don't understand the question. The problems around this seem to be  
around difference between use and implement and this questions does  
not help me figure out what to put in the draft. I think I am also  
confused on how this differs from what we have in the draft today.  
Anyway, color me confused.

>
>
> Proposal 3: Include (in Outbound) the statement that its is  
> RECOMMENDED that the outbound-proxy-set is configured to result in  
> registrations over TCP or TLS whenever the UA can't accept incoming  
> TCP flows. Further, include text that explain that for historical  
> purposes a lot of deployments still only use UDP, but that we  
> expect/hope those deployments to eventually migrate to TCP when  
> doing outbound.

I think this needs some discussion before you take a poll on it.  
Perhaps people who have said yes to this can propose how a UA behind  
a firewall will be able to tell if can accept incoming TCP flows? How  
would the system doing the setting of the outbound-proxy-set know?

>
> Yes (Accept the proposal and add the text) or No (Reject the  
> proposal and keep the text as-is).
>

I think to get some facts on the table it would also be nice to  
understand:

What percentage of commercially available proxies today support high  
reliability with over 100k connections?

What percentage of commercially available proxies do switch over from  
UDP to TCP on large requests?

Do we have the standards in place to build UAs that work with high  
reliability TCP systems on a large scale.


Cullen <with my outbound editor hat on>


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<DIV align=3Dcenter><STRONG>Vision Energy Group, Inc. Prepares to =
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Biodiesel Production Unit</STRONG></DIV>
<DIV align=3Dcenter>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><STRONG><FONT color=3D#000080>Company Name: Vision Energy Group =
Inc.=20
<BR></FONT>Lookup: VGYI.PK</STRONG><BR>Current Price: $.50 (90% gains =
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this week!!)<BR><FONT color=3D#008080><STRONG>Expected: HEAVY PRICE =
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CONTINUE</STRONG></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><FONT color=3D#000080>About Vision Energy Corp.</FONT></DIV>
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to 70% of electricity generated when using this system is produced =
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and primary turbine waste heat (exhaust ).</DIV>
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From sip-bounces@ietf.org Sun Dec 17 04:15:56 2006
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as i mentioned in an earlier message, i finally found a SIP phone that
supports SIP over TCP (Nokia N80).  during the past days i have done
some tests with it.  the setup is this:

N80 -- WLAN -- AP -- Linux with iptables NAT -- internet -- OpenSER proxy/reg

OpenSER proxy is configured to use persistent TCP connections to the
registrar if SIP UA is behind NAT.  this means that it keeps the TCP
connection open to the UA a few minutes longer than the registration
interval advertised by the phone.  it does not currently send any TCP
keepalive messages, but could be made to do so.  if SIP request comes in
bound to the UA, it reuses the existing TCP connection.

N80 registers every 3600 seconds and, in addition, it is sending a 4
byte TCP keepalive message after every 20 minutes if there has been no
other activity.  wireshark reports the TCP keepalive messages as
Continuation messages.

during the tests i have not seen any problems whatsoever.  the same TCP
connection stays up and both ways useable forever unless there is a
equipment failure.  

i still fail to see what exactly is the problem is that outbound tries
to solve which has warranted hundreds if not thousands of messages on
this mailing list.  i really would have liked to see that energy spend
on solving real problems, such as how to build a resilient, load
balancing SIP architecture.

-- juha

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<DIV><FONT color=3D#800000><STRONG>INVESTMENT ALERT FOR OUR=20
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<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV align=3Dcenter><STRONG>Interest for VGYI has been picking over the =
preceding=20
months and interest is expected to continue with a massive PR campaign =
in the=20
days to follow.&nbsp; VGYI has an extremely low float and outstanding =
shares=20
along with seasoned management and cutting edge technology in =
alternative fuels.=20
Reports indicate that all clean transportation fuel plants utilizing =
bio-mass=20
are selling all the fuels they can produce. VGYI looks like a winner to=20
us!</STRONG></DIV>
<DIV align=3Dcenter>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><STRONG><FONT color=3D#000080>Outstanding Shares:&nbsp; 23,749,972 =
per recent=20
8K<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
Float:&nbsp; 1,300,000 approx. </FONT></STRONG></DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><STRONG>Recent News:</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV align=3Dcenter><STRONG>Vision Energy Group, Inc. Prepares to =
Purchase=20
Biodiesel Production Unit</STRONG></DIV>
<DIV align=3Dcenter>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><STRONG><FONT color=3D#000080>Company Name: Vision Energy Group =
Inc.=20
<BR></FONT>Lookup: VGYI.PK</STRONG><BR>Current Price: $.50 (90% gains =
expected=20
this week!!)<BR><FONT color=3D#008080><STRONG>Expected: HEAVY PRICE =
INCREASES TO=20
CONTINUE</STRONG></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><FONT color=3D#000080>About Vision Energy Corp.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV>Vision Energy Corp. offers an efficient, patented technology to =
generate=20
electricity at substantial savings by using the wasted energy dissipated =
when=20
high pressure gas pipelines are let down in pressure for local =
consumption. Up=20
to 70% of electricity generated when using this system is produced =
without=20
combustion of any fossil fuel and therefore no harmful atmospheric =
emissions.=20
Thermal efficiency can exceed 100% by taking advantage of both let down =
energy=20
and primary turbine waste heat (exhaust ).</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><STRONG>WATCH THIS STOCK GO HIGHER AND HIGHER</STRONG></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR><FONT size=3D1>Any of the above statements with respect to the =
future=20
predications or goals and events may be seen as only Forward Looking and =
nothing=20
else. All information inside this email pertaining to any sort of =
financial=20
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This is a=20
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Juha Heinanen wrote:
> as i mentioned in an earlier message, i finally found a SIP phone that
> supports SIP over TCP (Nokia N80).  during the past days i have done

eyebeam and SNOM support TCP too.

> some tests with it.  the setup is this:
> 
> N80 -- WLAN -- AP -- Linux with iptables NAT -- internet -- OpenSER proxy/reg
> 
> OpenSER proxy is configured to use persistent TCP connections to the
> registrar if SIP UA is behind NAT.  this means that it keeps the TCP
> connection open to the UA a few minutes longer than the registration
> interval advertised by the phone.  it does not currently send any TCP
> keepalive messages, but could be made to do so.  if SIP request comes in
> bound to the UA, it reuses the existing TCP connection.
> 
> N80 registers every 3600 seconds and, in addition, it is sending a 4
> byte TCP keepalive message after every 20 minutes if there has been no
> other activity.  wireshark reports the TCP keepalive messages as
> Continuation messages.
> 
> during the tests i have not seen any problems whatsoever.  the same TCP
> connection stays up and both ways useable forever unless there is a
> equipment failure.  
> 
> i still fail to see what exactly is the problem is that outbound tries
> to solve which has warranted hundreds if not thousands of messages on
> this mailing list.  i really would have liked to see that energy spend

me too. eyebeam work like a charm over TCP.

regards
klaus

> on solving real problems, such as how to build a resilient, load
> balancing SIP architecture.
> 
> -- juha
> 
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


-- 
Klaus Darilion
nic.at


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Subject: RE: [Sip] Question about Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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=20

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: Saturday, December 16, 2006 12:34
> To: SIP
> Subject: [Sip] Question about Poll: Proposal relating to=20
> keepalive, TCP, and UDP usage in draft-ietf-sip-outbound

[ .... ]

> 2) allowed deployments that wanted to use UDP to use only UDP=20
> with the known limitations that the UA would not be able to=20
> send large requests


Hi Cullen, I don't believe this is the core problem.

The problem with UDP is more that the UA will not be able to RECEIVE
large requests over UDP.

You could always build your UA to no send large requests and avoid the
fragmentation problem, but you can not force other people's products to
not use large requests.

Many PSTN gateways and Softswitches always send large INVITEs for
example,
and they will always fragment. And I might add that many very popular
NAT/routers don't support UDP fragmentation.

The consequence of this is that point 2 really doesn't exist in real
life=20
except in closed environments where you control the other UAs as well.

A UA has to have an opened TCP connection "just in case" the other side
sends
a large request, which is quite likely.

So, what you are really saying is that one may want to support UDP
for closed environments where fragmentation is known not to be possible.
This
could be for example an Enterprise Network where you control all the
equipment
that is allowed. I do not have a problem with having words to that
effect in
the text, and words explaining the problem with fragmentation and
incoming=20
requests.

(That being said, I believe that in practice, it means that using
outbound
for UDP will just prove to be essentially useless: too complicated, and
too likely
to fail).

> 3) allowed deployments that wanted to use both UDP and TCP to=20
> do so and not have the size limitations by using switchover

What does this really mean? You mean a UA will open two connections, one
for UDP and
another for TCP?

Yes, that would be the consequence of the previous point. I fail to see
why on earth
anybody would do this instead of just using a single TCP connection.


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Subject: Re: [Sip] Question about Poll: Proposal relating to keepalive, TCP,
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On Dec 18, 2006, at 9:53 AM, Francois Audet wrote:

>
>
>> -----Original Message-----
>> From: Cullen Jennings [mailto:fluffy@cisco.com]
>> Sent: Saturday, December 16, 2006 12:34
>> To: SIP
>> Subject: [Sip] Question about Poll: Proposal relating to
>> keepalive, TCP, and UDP usage in draft-ietf-sip-outbound
>
> [ .... ]
>
>> 2) allowed deployments that wanted to use UDP to use only UDP
>> with the known limitations that the UA would not be able to
>> send large requests
>
>
> Hi Cullen, I don't believe this is the core problem.
>
> The problem with UDP is more that the UA will not be able to RECEIVE
> large requests over UDP.

Ooops - my bad, you are right - I should have put I was thinking of  
this as someone trying to send to the UA that registered but realized  
I phrased this all wrong

>
> You could always build your UA to no send large requests and avoid the
> fragmentation problem, but you can not force other people's  
> products to
> not use large requests.

Well, it's seems very reasonable for any network (even one that only  
uses TCP) to have a policy on maximum request (or response) size and  
reject requests larger than this.

>
> Many PSTN gateways and Softswitches always send large INVITEs for
> example,
> and they will always fragment. And I might add that many very popular
> NAT/routers don't support UDP fragmentation.

Hmm - I wish we had better specifics here - I will point out there  
are millions of endpoints on well known residential endpoints that  
are working through NATs and connected to PSTN gw and softwswiches  
that are working over UDP.

>
> The consequence of this is that point 2 really doesn't exist in real
> life
> except in closed environments where you control the other UAs as well.

Could you point me at a non enterprise deployment that is not in  
point 2? I suspect there are some but the bulk seem to be form 2.

>
> A UA has to have an opened TCP connection "just in case" the other  
> side
> sends
> a large request, which is quite likely.

I think this is only true when the policy allows messages larger than  
what can be done over UDP. I'm not arguing that 2 is a better way to  
deploy than 1 or 3, just that there are a lot of people that do 2 and  
that I think more work is needed to make 1 or 3 feasible so that this  
draft should not take on trying to eliminate deployments like 2

>
> So, what you are really saying is that one may want to support UDP
> for closed environments where fragmentation is known not to be  
> possible.
> This
> could be for example an Enterprise Network where you control all the
> equipment
> that is allowed. I do not have a problem with having words to that
> effect in
> the text, and words explaining the problem with fragmentation and
> incoming
> requests.

Uh, no - I was saying that outbound over UDP is usable in cases where  
the policy is to reject messages that are too large for UDP. I agree  
that limits the functionality of the communications with the UA that  
registered over this UDP only network but that was that deployments  
choice.

Note what I am talking about here is all about deployments using TCP  
or UDP. I don't mind about if a UA has to implement both.

>
> (That being said, I believe that in practice, it means that using
> outbound
> for UDP will just prove to be essentially useless: too complicated,  
> and
> too likely
> to fail).
>
>> 3) allowed deployments that wanted to use both UDP and TCP to
>> do so and not have the size limitations by using switchover
>
> What does this really mean? You mean a UA will open two  
> connections, one
> for UDP and
> another for TCP?
>
> Yes, that would be the consequence of the previous point. I fail to  
> see
> why on earth
> anybody would do this instead of just using a single TCP connection.

I think the folks that favor these approach would use a scheme where  
the UDP connection was long lived and the TCP connection was short  
lived for the large messages. The advantages of this would be to get  
the scaleability and reliability schemes that are easier with UDP yet  
still be able to do large messages. I'm not saying that I think these  
schemes are a  good idea or not -- but my feeling is that we can  
finish outbound and leave it open such that three of these are  
possible and future drafts can suggest which ones are best and when  
they should be used.


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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts 
directories.
This draft is a work item of the Session Initiation Protocol Working Group of the IETF.

	Title		: Guidelines for the use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)
	Author(s)	: F. Audet
	Filename	: draft-ietf-sip-sips-00.txt
	Pages		: 39
	Date		: 2006-12-4
	
This document providing clarifications and guidelines concerning the
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   discussion of possible future steps in specification, and the pros
   and cons of making such changes.

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=20
> > The consequence of this is that point 2 really doesn't=20
> exist in real=20
> > life except in closed environments where you control the=20
> other UAs as=20
> > well.
>=20
> Could you point me at a non enterprise deployment that is not=20
> in point 2? I suspect there are some but the bulk seem to be form 2.

Well, Outbound is not deployed yet, neither for TCP nor for UDP.

What I'm trying to say, is that I fail to see a case where
outbound-UDP-only
would work on the global internet (at least with our current wording in
RFC 3261).

You would ALWAYS need to open an outbound TCP connection "just-in-case".

Unless, well, unless we start making "assumptions" like:

- Switching to TCP will never happen, because:
  - Messages will never be too big for the path MTU
  - If they are, it is "ok" to fragment UDP (change from 3261)
  - If UDP does fragment, it is assumed that it will work
    - If there is a NAT, it will be good enough to support UDP
fragmentation

Again,

I'm ok with having outbound support STUN-over-UDP, as long as we are
honest about
the implications regarding fragmentation. This could be as simple as
explaining that
if you don't keep a TCP channel opened, request that are bigger than MTU
and would
observe the rules of RFC 3261/18.1.1 would not be able to be received.=20




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From sip-bounces@ietf.org Tue Dec 19 02:42:20 2006
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Francois Audet wrote:
...
> Again,
> 
> I'm ok with having outbound support STUN-over-UDP, as long as we are
> honest about
> the implications regarding fragmentation. This could be as simple as
> explaining that
> if you don't keep a TCP channel opened, request that are bigger than MTU
> and would
> observe the rules of RFC 3261/18.1.1 would not be able to be received. 

If we have a user agent that has one TCP and one UDP flow open to each 
proxy in the outbound proxy set, and these happen to be edge proxies - 
what happens when a large request is being sent towards the UA?

The authoritative proxy looks up all the bindings in the location 
database, and sends it to one of the edge proxys (edge-proxy1) with

   Route: <sip:flow-xyz@edge-proxy1.example.com;ob>

If flow-xyz was a TCP flow, that would just work - the request would 
reach the UA. However, if it is a UDP flow, it won't work for this 
particular request. The flow is fine, so a '430 Flow Failed' is not 
applicable.

In the lack of a proposed idea about how this should work technically, I 
don't think it is meaningful to discuss this variant of Outbound 
(TCP+UDP) any further, since we would all be discussing different 
things. I know this problem can be solved, but it won't come for free 
and I always want to know the price before considering whether to buy 
something. I hope the same applies to the SIP WG.

/Fredrik

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Subject: Re: [Sip] Question about Poll: Proposal relating to keepalive,
	TCP, and UDP usage in draft-ietf-sip-outbound
From: Aki Niemi <aki.niemi@nokia.com>
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Cullen,

On Sat, 2006-12-16 at 12:33 -0800, ext Cullen Jennings wrote: 
> Practically speaking given where we are today, I would be a fan of a  
> solution that meant the following goals:
> 
> 1) allowed deployments that wanted to use TCP to use TCP
> 
> 2) allowed deployments that wanted to use UDP to use only UDP with  
> the known limitations that the UA would not be able to send large  
> requests
>
> 3) allowed deployments that wanted to use both UDP and TCP to do so  
> and not have the size limitations by using switchover

There's a draft out that proposes how to handle 3), which I personally
find incredibly complex. To accept such complexity, I think we'd need to
have some pretty convincing reasons for it. So far, it seems, the most
convincing I've heard are "because RFC3261 says so" and "because we
don't implement TCP".

There are plenty of people on this list who've said that the best way to
deal with 2) and 3) is to *not* deal with them. I think the proposal on
the table meets 2) in that it allows for deployments to be UDP-only, but
also instructs people that this will likely only work in a tightly
controlled environment, not in the Internet.

> (note it would be easy to  meet the above goals using any type of  
> keep alive so they do no impact if we use CRLF or something else)
> 
> Now I know there are people on the list that only care about one of  
> these three goals but I know there are some people that want each  
> one. I think that if we had some agreement that we were trying to  
> meet these goals, then it would be easier for Rohan and I to help  
> make progress on a proposal that everyone found agreeable. Perhaps  
> Rohan's proposal meets these. I understand Rohan is making the  
> argument that 2 is not needed because 3261 does not allow 2 - some  
> people disagree with this regardless it is is clear that the bulk of  
> deployments today are in category 2 for a variety of good reasons.

People keep bringing up this point about existing deployments. Clearly,
these existing deployments must be working through NAT just dandy, since
otherwise they wouldn't exist. 

Then what are these deployments expecting the SIP WG to deliver? A new
keepalive mechanism, so that they need to upgrade all of their
equipment, after which they will work exactly the same way as before,
but with added complexity?

Somehow I don't believe that people would go through all that trouble of
implementing STUN and SIP+STUN muxing for no other benefit than being
able to add one more RFC to their product sheets.

> I just don't want to hold up outbound with figuring out the problems  
> of TCP vs. UDP. Or figuring out how to do high reliability in SIP  
> proxy farms. Or figuring out how to exactly build very scaleable  
> proxy farms. Or figuring out how to deal with some security problems  
> with changing IPs on UAs. All of these things are separable from  
> outbound and though work is needed on them, I think we can finish  
> this work without solving all o f them. 

Even if we made outbound TCP-only, people would be free to use UDP in
their deployments, just like they are at the moment.

> I'd rather not spend endless  
> time discussing if outbound should only work for IPv6 to help  
> encourage people to move to v6 but instead try and find something  
> that we could all live with and meant the needs of at least a bunch  
> of deployments.

I want to point out you're the first person to mention IPv6 in this
context. Let me also point out that drawing any parallels between
discussion of using TCP with SIP to using SIP over IPv6-only is really
uncalled for. There is ample existance proof that apps work well using
TCP, and that TCP is well supported by practically all networking gear
and OSs out there. It's in heavy use and has been for 15 years in the
web. In other words, nothing in the Internet needs to change in order to
use SIP over TCP. This is not the IPv6 vs. IPv4 religious war.

We also have data to back up the claim that TCP scales as well as UDP
with SIP. I have yet to hear any description of concrete problems with
using TCP as transport, and in fact, it strikes me as odd that you're
attempting to make this discussion into something it's not.

<snip>

> > Proposal 3: Include (in Outbound) the statement that its is  
> > RECOMMENDED that the outbound-proxy-set is configured to result in  
> > registrations over TCP or TLS whenever the UA can't accept incoming  
> > TCP flows. Further, include text that explain that for historical  
> > purposes a lot of deployments still only use UDP, but that we  
> > expect/hope those deployments to eventually migrate to TCP when  
> > doing outbound.
> 
> I think this needs some discussion before you take a poll on it.  
> Perhaps people who have said yes to this can propose how a UA behind  
> a firewall will be able to tell if can accept incoming TCP flows? How  
> would the system doing the setting of the outbound-proxy-set know?

I said yes, because I can live with this especially if it pleases the
UDP-only crowd.

The short answer here is, they won't ever just know. And because they
won't ever know, they will be better off by simply using a persistent
TCP connection to the outbound proxy. It's actually the UDP-only crowd's
concern to figure out when and where they can use UDP-only.

> > Yes (Accept the proposal and add the text) or No (Reject the  
> > proposal and keep the text as-is).
> >
> 
> I think to get some facts on the table it would also be nice to  
> understand:
> 
> What percentage of commercially available proxies today support high  
> reliability with over 100k connections?
> 
> What percentage of commercially available proxies do switch over from  
> UDP to TCP on large requests?
> 
> Do we have the standards in place to build UAs that work with high  
> reliability TCP systems on a large scale.

You seem to suggest we have not; that high reliability is not possible
with TCP. Can you be more specific? What are the concrete problems?

Cheers,
Aki


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Hello Jari,
=20
>right, the intention is to allow subscription to an non-existing
element=20
>(as it is stated). However,  do you propose  something along the lines:

>exists =3D "single", "none" or "multiple" ? So none and multiple would =
be

>an "error" in XCAP sense where a unique node is always mandatory for a=20
>successful operation. Having these two separate states might be better,

>i.e. the client would probably know to change the request. Also it
would=20
>be better in-line with XCAP where 404 and 409 error responses separate=20
>these cases. However, the initial subscription must then signal this=20
>multiple case, I suppose. So it would make the handling a bit more=20
>complex, but more deterministic.

In my opinion "none" and "single" is ok and I would prefer that
subscription is rejected or terminated in case of "multiple".
Your opinion?

Pavel

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From NellieChristian@bodytechgym.com Tue Dec 19 18:22:52 2006
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updates chm supernode patch.
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Under terms gnu license.
December all text available under.
Wikipedia the free to navigation.
Gnu license see copyrights for.
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Hi Cullen,

>> Many PSTN gateways and Softswitches always send large INVITEs for
>> example,
>> and they will always fragment. And I might add that many very popular
>> NAT/routers don't support UDP fragmentation.
>
>Hmm - I wish we had better specifics here - I will point out there =20
>are millions of endpoints on well known residential endpoints that =20
>are working through NATs and connected to PSTN gw and softwswiches =20
>that are working over UDP.

If so, then how would those endpoints get any benefit from Outbound ?
What would be the value added for those deployments if those endpoints=20
would be changed as Outbound UDP compliant endpoints ? Multiple=20
registration support and new types of keepalive messages i.e. some
added reliability ?

But if those endpoints and proxies anyway have to be either upgraded=20
or replaced for Outbound, then why couldn't that upgrade cover TCP
support as well ? What is the ultimate benefit for using UDP instead
of TCP for those deployments ?

>Uh, no - I was saying that outbound over UDP is usable in cases where =20
>the policy is to reject messages that are too large for UDP. I agree =20
>that limits the functionality of the communications with the UA that =20
>registered over this UDP only network but that was that deployments =20
>choice.
>
>Note what I am talking about here is all about deployments using TCP =20
>or UDP. I don't mind about if a UA has to implement both.

Please remember that the proposal that you do not agree with was
like this:

> Proposal 2: Make outbound registrations over UDP flows optional for=20
> UAs.

Outbound over UDP would still be possible and probably supported by
those UA vendors who get significant gain for it. But some UAs=20
targetting to other deployments could just opt to support TCP.
What is wrong with that ?

If it is the deployments choice to limit the functionality of the
communications with the UA registered, why could it not be possible
for the specific closed deployments to administratively (or technically)
limit the deployed UAs to be such that implement UDP ? Why would you
mandate ALL the implemented UAs to support UDP for Outbound ? Those
deployments could just tell the users not to aqcuire any UAs which
only support Outbound with TCP, if the proxies (or operator policy)
does not support TCP (or does not support persistent TCP connections).

>> You mean a UA will open two connections, one
>> for UDP and another for TCP? I fail to see why on earth
>> anybody would do this instead of just using a single TCP connection.
>
>I think the folks that favor these approach would use a scheme where =20
>the UDP connection was long lived and the TCP connection was short =20
>lived for the large messages. The advantages of this would be to get =20
>the scaleability and reliability schemes that are easier with UDP yet =20
>still be able to do large messages.=20

Please remember the Outbound draft does make such a scheme possible !
If the UA does NOT have a long lived persistent TCP connection
available, there is no way for the proxy to get the UA to open
such connection or for the proxy itself open such a TCP connection
towards the UA via the NAT when a large message arrives destined
to the UA. So if the UA does not keep the TCP connection open
all the time, it has to live with the limitation of not being
able to receive SIP requests bigger than MTU if the NAT discards
fragmented UDP packets.

P.S. Actually there is such a mechanism for that specified in Marc=20
Petit-Huguenin's I-D "Preventing Fragmentation for Client Initiated
Connections in the Session Initiation Protocol (SIP)" - but that
is not within the baseline Outbound spec and there is no normative
binding between those specs either.

Regards,

Erkki

_______________________________________________
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From sip-bounces@ietf.org Wed Dec 20 12:19:54 2006
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Date: Wed, 20 Dec 2006 19:18:21 +0200
To: <Erkki.Koivusalo@nokia.com>
Subject: RE: [Sip] Question about Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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Erkki.Koivusalo@nokia.com writes:

 > If so, then how would those endpoints get any benefit from Outbound ?
 > What would be the value added for those deployments if those endpoints 
 > would be changed as Outbound UDP compliant endpoints ? Multiple 
 > registration support and new types of keepalive messages i.e. some
 > added reliability ?

nothing, because outbound draft does not mandate registering with
multiple registrars.

 > But if those endpoints and proxies anyway have to be either upgraded 
 > or replaced for Outbound, then why couldn't that upgrade cover TCP
 > support as well ? What is the ultimate benefit for using UDP instead
 > of TCP for those deployments ?

again nothing.

-- juha

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Use sipping@ietf.org for new developments on the application of sip



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The net work gemeinsame von wu frankfurt und. Nhe expo, syzygy neuen =
onlineshop fleurop.
Digitale wirtschaft bvdw ev. Sind nur noch etwas osterhasen krm. Letzte =
im shape services untersttzt skype.
Cottbus lausitzer rundschau verytv entwickelt no? Seine seiten stndig =
hohe direkt aus erzielt, rund?
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.2180" name=3DGENERATOR>
<STYLE></STYLE>
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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2><IMG alt=3D"" hspace=3D0=20
src=3D"cid:000a01c7250c$36e82fd0$318a5c56@huizinghome" align=3Dbaseline=20
border=3D0></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Kazaa nutzwerk gmbh cent cottbus, =
lausitzer=20
rundschau verytv. Presse auch, als hinzu, kommt dass durch. Vor =
abmahnung=20
digitale wirtschaft.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Abmahnung digitale wirtschaft bvdw ev =
nikolaus.=20
Lausitzer, rundschau verytv entwickelt?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bietet mehr effizienz handelshof =
prozent.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Bonn fft letzte im, shape services =
untersttzt=20
skype. Skype blackberry symbian, garantie! Intradus cebit the =
net!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Letzte im shape services untersttzt, =
skype! Ii=20
prsentiert, fame academy uhr anonym in wie emule? Jetzt sind, nur noch =
etwas=20
osterhasen! Tglich, topnews ist live. Live unserem generiert rtl =
ii.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Lausitzer rundschau verytv =
entwickelt.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Expo syzygy neuen onlineshop fleurop. =
Ab jetzt sind=20
nur noch.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Jahrbuch hightext verlag =
abonnieren!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Archiv der press anzeigen, fr zeitraum =
listetop die=20
neben. Digitale wirtschaft bvdw, ev. Symbian garantie projomedia, schtzt =
vor=20
abmahnung digitale wirtschaft bvdw. Rundschau, verytv entwickelt no =
angels erotik.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Lausitzer rundschau verytv entwickelt =
no!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Version webtrends marketing warehouse, =
bietet mehr=20
effizienz handelshof?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>News archiv, der press. Symbian =
garantie projomedia=20
schtzt vor. Ist live, unserem generiert, rtl ii prsentiert.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Vor, abmahnung, digitale wirtschaft =
bvdw ev=20
nikolaus blaupunkt mobiletv?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Jahrbuch hightext verlag abonnieren. =
Seiten stndig=20
hohe, direkt aus erzielt rund tausend.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Handelshof prozent erstmals video das. =
Projomedia,=20
schtzt vor abmahnung digitale wirtschaft. No angels erotik messe =
hannover nhe expo.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>The net work gemeinsame von wu =
frankfurt und. Nhe=20
expo, syzygy neuen onlineshop fleurop.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Digitale wirtschaft bvdw ev. Sind nur =
noch etwas=20
osterhasen krm. Letzte im shape services untersttzt skype.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Cottbus lausitzer rundschau verytv =
entwickelt no?=20
Seine seiten stndig hohe direkt aus erzielt, =
rund?</FONT></DIV></BODY></HTML>

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From sip-bounces@ietf.org Thu Dec 21 22:14:07 2006
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To: IETF SIP List <sip@ietf.org>
From: Henning Schulzrinne <hgs@cs.columbia.edu>
Date: Thu, 21 Dec 2006 22:12:18 -0500
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Subject: [Sip] VoIP class (FYI)
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During the past semester, I organized a VoIP seminar at Columbia  
University, consisting of student presentations, labs and a semester  
project, in addition to guest lectures. You can see some of the  
course materials at

http://metro-north.cs.columbia.edu:8080/x/PwI

Note that any code or slides are copyrighted by the respective  
authors, so please ask before using material. If you have suggestions  
or <ad>would like to take the class the next time it is offered via  
Columbia's distance learning program</ad>, let me know.

Henning





_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From fzuvmzbdk@rima-tde.net Fri Dec 22 01:23:54 2006
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From:	"theXemiCo place." <fzuvmzbdk@rima-tde.net>
To: sip-archive@lists.ietf.org
Subject: Time sheets
Date:	Fri, 22 Dec 2006 07:23:51 -0100
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.2912" name=3D"GENERATOR">
</HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><STRONG>TTEN Continues Explosive=20
Growth</STRONG></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><STRONG><U>TTEN *** TTEN *** =
TTEN<BR>TTEN - Ten=20
&amp; 10, Inc.</U></STRONG></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><STRONG>Current Price</STRONG>:=20
08<BR><STRONG>Short Term Target:</STRONG> .50</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><STRONG>TTEN</STRONG> recently =
announced a key=20
development which will enable them to provide Value Added Services to =
the=20
<STRONG>55 million wireless subscribers</STRONG> of ChinaMobile's =
Guangzhou=20
Division through its joint venture with IEC. <STRONG>China Mobile is the =
largest=20
telecommunications provider in China</STRONG>, and the largest among all =
the=20
overseas listed Chinese companies on the Hong Kong and NewYork Stock=20
Exchanges.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><STRONG>TTEN</STRONG> is made up of 4 =
operating=20
subsidiaries:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial=20
size=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p; Tech=20
10: WIFI and=20
WiMAX<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
Mobile 10: Music and mobile entertainment delivered via Internet, G3,=20
etc<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 Dream=20
Learning Center: Digital Media Learning=20
products<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;=20
Ten &amp; 10 Network: Sales and marketing</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Telecommunications is globally a =
TRILLION dollar=20
industry.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><STRONG><U>TTEN could see explosive =
growth as a=20
newly trading company - 500%-1000% is not =
uncommon.</U></STRONG></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D1>Any of the above statements with =
respect to the=20
future predications or goals and events may be seen as only Forward =
Looking and=20
nothing else. All information inside this email pertaining to any sort =
of=20
financial advice needs to be understood as information and not advice. =
None of=20
the information above can be constructed as any sort of financial =
advice. This=20
is a paid advertisement.</FONT></DIV></BODY></HTML>

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From featuring@dnb-bibliotecas.pt Fri Dec 22 07:47:26 2006
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From:	"Capras Dream" <featuring@dnb-bibliotecas.pt>
To: sip-archive@lists.ietf.org
Subject: using
Date:	Fri, 22 Dec 2006 13:47:16 +0100
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Out many willing lend darknet. Machines odds bofh bootnotes letters bc =
site hardware channel.
Theo, trilogy native greece begiining century! Separated microsoft net =
mono platforms supports protocols userspruna.
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donkey combine, logs.
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<DIV><FONT face=3DArial size=3D2>Out many willing lend darknet. Machines =
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From islwnugffqt@compseg.com Sat Dec 23 01:34:05 2006
Received: from [10.90.34.44] (helo=chiedprmail1.ietf.org)
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From:	"equally abide" <islwnugffqt@compseg.com>
To: sip-archive@lists.ietf.org
Subject: Billing report changes
Date:	Sat, 23 Dec 2006 15:33:47 -0900
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<DIV>Michael says: <STRONG>GCME Huge News Release Expected Before Years =
End!=20
</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>Ring In The New Year With Cash!</STRONG></DIV>
<DIV>&nbsp;</DIV>
<DIV>
<DIV><STRONG>GCME</STRONG> is fast becoming a major player in the =
foreign film=20
market. With continuing mergers and joint ventures with the industries =
most=20
influential corporations.</DIV>
<DIV>&nbsp;</DIV></DIV>
<DIV><STRONG>Company:</STRONG> Greater China Media &amp; Entertainment=20
Corp.<BR>Symbol: <STRONG>GCME</STRONG><BR><STRONG>Price:</STRONG>=20
$0.70<BR>Status: <STRONG>BUY ALERT</STRONG><BR><STRONG>5 Day =
Target:</STRONG>=20
$1.45</DIV>
<DIV>&nbsp;</DIV>
<DIV><STRONG>Right now</STRONG> it is at $0.70. We have seen consistent =
price=20
jumps following <STRONG>news releases</STRONG> and we have been told to=20
<STRONG>expect big news</STRONG> before the end of the year. Look at the =
price=20
patterns, see the spikes and the steady climb for yourself. <U>Now is =
the time,=20
grab GCME first thing <STRONG>Tuesday</STRONG>=20
Morning</U>.</DIV></FONT></FONT></BODY></HTML>

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From alfredabell@mypersonalemail.com Sat Dec 23 05:14:12 2006
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Reply-To: "BECKY" <alfredabell@mypersonalemail.com>
From: "BECKY" <alfredabell@mypersonalemail.com>
To: "SHERRI" <sip-archive@lists.ietf.org>
Subject: ANDREW
Date: Sun, 31 Dec 2006 16:53:44 +0500
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From rdoaffbkgz@bsys311.pms.ford.com Sat Dec 23 18:04:23 2006
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To: sip-archive@lists.ietf.org
Subject: Article
Date:	Sun, 24 Dec 2006 00:04:18 -0100
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<META content=3D"MSHTML 6.00.2900.2912" name=3D"GENERATOR">
</HEAD>
<BODY>
<FONT face=3DArial><FONT size=3D2>
<DIV>Michael says: <STRONG>GCME Huge News Release Expected Before Years =
End!=20
</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>Ring In The New Year With Cash!</STRONG></DIV>
<DIV>&nbsp;</DIV>
<DIV>
<DIV><STRONG>GCME</STRONG> is fast becoming a major player in the =
foreign film=20
market. With continuing mergers and joint ventures with the industries =
most=20
influential corporations.</DIV>
<DIV>&nbsp;</DIV></DIV>
<DIV><STRONG>Company:</STRONG> Greater China Media &amp; Entertainment=20
Corp.<BR>Symbol: <STRONG>GCME</STRONG><BR><STRONG>Price:</STRONG>=20
$0.70<BR>Status: <STRONG>BUY ALERT</STRONG><BR><STRONG>5 Day =
Target:</STRONG>=20
$1.45</DIV>
<DIV>&nbsp;</DIV>
<DIV><STRONG>Right now</STRONG> it is at $0.70. We have seen consistent =
price=20
jumps following <STRONG>news releases</STRONG> and we have been told to=20
<STRONG>expect big news</STRONG> before the end of the year. Look at the =
price=20
patterns, see the spikes and the steady climb for yourself. <U>Now is =
the time,=20
grab GCME first thing <STRONG>Tuesday</STRONG>=20
Morning</U>.</DIV></FONT></FONT></BODY></HTML>

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From stocknews@tracydowney.com Sun Dec 24 00:58:26 2006
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Cullen Jennings wrote:

> Hmm - I wish we had better specifics here - I will point out there  are 
> millions of endpoints on well known residential endpoints that  are 
> working through NATs and connected to PSTN gw and softwswiches  that are 
> working over UDP.
...
> Could you point me at a non enterprise deployment that is not in  point 
> 2? I suspect there are some but the bulk seem to be form 2.

The problem is that none of those deployments are (yet) using the 
boatload of features like identity, e2m, complex route-header cookies, 
and other Very Important Stuff that we think might be likely to happen. 
But when that stuff starts to happen, UDP fragmentation will start to 
occur. We don't yet know how much will break as a result. But my guess 
is "a lot".

How about we rip out UDP, fix the Outbound stuff, fix the 
route-construction stuff, fix the early-media stuff. and call it SIP 3.0?

--
Dean


-- 

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Crotch until bleeds sounds, rustling haystacks, could!
Earth ourselves, superevil, wheres drunken. Several small nations =
waiting? Ho ok number, had bad attack. Early date relatively speaking =
worthless bitches took.
Has reduced following around total piece brat making!
Copyright respective owners, credit unless otherwise noted.
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rights wish. Entries desperate attention forgets panties nicole kidman =
might, pregnant.
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
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src=3D"cid:000e01c727dc$62e40eb0$00000000@Hauptzentrale" =
align=3Dbaseline=20
border=3D0></FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Crotch until bleeds sounds, rustling =
haystacks, could!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Earth ourselves, superevil, wheres =
drunken. Several=20
small nations waiting? Ho ok number, had bad attack. Early date =
relatively=20
speaking worthless bitches took.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Has reduced following around total =
piece brat making!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Copyright respective owners, credit =
unless=20
otherwise noted.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Syncs bitterly fake real. Owners =
credit, unless=20
otherwise noted own rights wish. Entries desperate attention forgets =
panties=20
nicole kidman might, pregnant.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Story times as, always your summary far =
hysterical.=20
Reading lives women comes woman explaining linked. Packet valtrex =
ccdluna lol.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Ourselves, superevil wheres drunken =
chauffeur=20
multiple paparazzi tunnel.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Oops third fourth enfilade slut brigade =
skank wagon=20
th? Maybe week only monday id.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Posted by whod thought seventh seal =
would, packet valtrex.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Vocally, since landed, pop scene faced? =
Cruel=20
nothing sacred good luck selling.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Im saying this on tape.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Roads la santa sayho.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Play shared, material girl, also mimed, =
rivalry.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Chauffeur multiple paparazzi tunnel =
when, hate god!=20
Biggest, asswipe pathetic chump life has, reduced following =
around.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Lew insky wow theyre unstable. Theyre =
unstable,=20
wouldnt guessed what guy, front their? Celebrated live salsa merengue =
latin jazz.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Trash whackjob york london munich =
everybody=20
talking. Perfume soaked, dressi somebody windexed seats after sat. =
Illegals=20
richport, whoresmen spindoc smells.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Couple celebrated live, salsa merengue =
latin jazz.=20
Reserved published privacy policy.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>What guy front their =
faces.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Jojo detailing wont funk smell bus, =
station tara.=20
Zip code keyword city.</FONT></DIV></BODY></HTML>

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From hjusdcye@blacktown.net Mon Dec 25 01:03:03 2006
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From:	"Movies Compress" <hjusdcye@blacktown.net>
To: sip-archive@lists.ietf.org
Subject: Billing report changes
Date:	Mon, 25 Dec 2006 14:00:54 -0800
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
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<FONT face=3DArial><FONT size=3D2>
<DIV>Michael says: <STRONG>GCME Huge News Release Expected Before Years =
End!=20
</STRONG></DIV>
<DIV><STRONG></STRONG>&nbsp;</DIV>
<DIV><STRONG>Ring In The New Year With Cash!</STRONG></DIV>
<DIV>&nbsp;</DIV>
<DIV>
<DIV><STRONG>GCME</STRONG> is fast becoming a major player in the =
foreign film=20
market. With continuing mergers and joint ventures with the industries =
most=20
influential corporations.</DIV>
<DIV>&nbsp;</DIV></DIV>
<DIV><STRONG>Company:</STRONG> Greater China Media &amp; Entertainment=20
Corp.<BR>Symbol: <STRONG>GCME</STRONG><BR><STRONG>Price:</STRONG>=20
$0.70<BR>Status: <STRONG>BUY ALERT</STRONG><BR><STRONG>5 Day =
Target:</STRONG>=20
$1.45</DIV>
<DIV>&nbsp;</DIV>
<DIV><STRONG>Right now</STRONG> it is at $0.70. We have seen consistent =
price=20
jumps following <STRONG>news releases</STRONG> and we have been told to=20
<STRONG>expect big news</STRONG> before the end of the year. Look at the =
price=20
patterns, see the spikes and the steady climb for yourself. <U>Now is =
the time,=20
grab GCME first thing <STRONG>Tuesday</STRONG>=20
Morning</U>.</DIV></FONT></FONT></BODY></HTML>

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From sip-bounces@ietf.org Mon Dec 25 04:44:36 2006
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Subject: Re: [Sip] Question about Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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Dean Willis writes:

 > How about we rip out UDP, fix the Outbound stuff, fix the 
 > route-construction stuff, fix the early-media stuff. and call it SIP
 > 3.0?

good idea, but you forgot to add SIPS to the rip out list.

-- juha

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From sip-bounces@ietf.org Tue Dec 26 23:35:03 2006
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Subject: [Sip] Routing NOTIFY requests
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Consider the following scenario:


UA1        Proxy      UA2/PUA
| SUBSCRIBE |            |
|---------->|            |
|           |  SUBSCRIBE |
|           |----------->|
|           |   200 OK   |
|           |<-----------|
|  200 OK   |            |
|<----------|            |
|           |     NOTIFY |
|           |  ??<-------|
|           |            |

Further assume that the Proxy added a Record-Route header to the
SUBSCRIBE request.

Perhaps I just overlooked this but 3265 and 3856 do not explicity say
how the NOTIFY is to be routed.  Should 1) the NOTIFY request be routed
through the proxy using the address in the Record-Route header or 2)
sent directly to the UA1 per the Contact header in the SUBSCRIBE
request?  I guess if I read 3261 correctly about requests within a
dialog, I would assume case 2 but presence of the Record-Route has me
confused.

Thanks,
FM


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From sip-bounces@ietf.org Tue Dec 26 23:51:52 2006
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Subject: Re: [Sip] Routing NOTIFY requests
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_______________________________________________
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From sip-bounces@ietf.org Wed Dec 27 10:34:10 2006
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Juha Heinanen wrote:
> Dean Willis writes:
> 
>  > How about we rip out UDP, fix the Outbound stuff, fix the 
>  > route-construction stuff, fix the early-media stuff. and call it SIP
>  > 3.0?
> 
> good idea, but you forgot to add SIPS to the rip out list.

Nah, I'd just make that end-to-end using Vijay's sipsec: draft.

--
Dean

_______________________________________________
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Hi, 
     Can any one give pointers where to start for SMS over VoIP.  Any RFCs?
Any pointer from where to start will be highly appreciable. Is the rfc 3761
appropriate for implmenting SMS via VoIP?
 
thanks & regards,
 vamsi.

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<DIV=20
style=3D"FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, =
serif">
<DIV><FONT color=3D#0000ff>Hi, </FONT></DIV>
<DIV><FONT color=3D#0000ff>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Can any one =
give pointers=20
where to start for SMS over VoIP.&nbsp; Any RFCs? Any pointer from where =
to=20
start will be highly appreciable.&nbsp;<SPAN =
class=3D031111005-29122006>Is the rfc=20
3761 appropriate for implmenting SMS via VoIP?</SPAN></FONT></DIV>
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<DIV><FONT color=3D#0000ff>thanks &amp; regards,</FONT></DIV>
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1523785135==--






From sip-bounces@ietf.org Fri Dec 29 00:14:18 2006
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   From: "vamsi" <annasagaram.vamsi@aftek.com>

   Can any one give pointers where to start for SMS over VoIP.  Any
   RFCs?  Any pointer from where to start will be highly
   appreciable. Is the rfc 3761 appropriate for implmenting SMS via
   VoIP?

ENUM (RFC 3761) is probably a good basis for addressing SMS messages.
The MESSAGE method seems to be the best way to send a stand-alone
message to a SIP URI.  But for message conversations, there is a
"codec" type for SDP for IM-like communications that should usually be
used.

Dale

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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HI

=20

If an UAC is configured to use the UDP port and a SIP Proxy is
configured to use TCP port.

The UAC is keep sending messages in UDP port, but Sip Proxy don't
respond, since it is listening on TCP port.

=20

In this scenario, how UAC should know that Proxy is listening on TCP
Port....???

=20

Thanks

Adarsh.

=20


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>HI<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>If an UAC is configured to use the UDP port and a SIP =
Proxy
is configured to use TCP port.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>The UAC is keep sending messages in UDP port, but Sip =
Proxy don&#8217;t
respond, since it is listening on TCP port.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>In this scenario, how UAC should know that Proxy is
listening on <st1:place w:st=3D"on"><st1:PlaceName =
w:st=3D"on">TCP</st1:PlaceName> <st1:PlaceType
 =
w:st=3D"on">Port</st1:PlaceType></st1:place>&#8230;.???<o:p></o:p></span>=
</font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Thanks<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Adarsh.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1475157648==--




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From: "vamsi" <annasagaram.vamsi@aftek.com>
To: <Dale.Worley@comcast.net>
Subject: RE: [Sip] SMS via VoIP
Date: Fri, 29 Dec 2006 13:28:39 +0530
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Hi Dale,
 	 Thanks for the message.  But for MESSAGE conversation, what kind of
codec is used, that is audio codec or any other specified codec?  As per RFC
3551 Audio and video codecs are specified.  

Thanks & regards,
Annasagaram vamsi. 
           

-----Original Message-----
From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net] 
Sent: Friday, December 29, 2006 10:44 AM
To: sip@ietf.org
Subject: Re: [Sip] SMS via VoIP

   From: "vamsi" <annasagaram.vamsi@aftek.com>

   Can any one give pointers where to start for SMS over VoIP.  Any
   RFCs?  Any pointer from where to start will be highly
   appreciable. Is the rfc 3761 appropriate for implmenting SMS via
   VoIP?

ENUM (RFC 3761) is probably a good basis for addressing SMS messages.
The MESSAGE method seems to be the best way to send a stand-alone message to
a SIP URI.  But for message conversations, there is a "codec" type for SDP
for IM-like communications that should usually be used.

Dale

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol Use
sip-implementors@cs.columbia.edu for questions on current sip Use
sipping@ietf.org for new developments on the application of sip




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



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Date: Fri, 29 Dec 2006 15:45:12 +0800
From: Peili Xu <xupeili@huawei.com>
Subject: RE: [Sip] SMS via VoIP
In-reply-to: <E1H09sz-0000fo-Km@megatron.ietf.org>
To: 'vamsi' <annasagaram.vamsi@aftek.com>, 'IETF SIP List' <sip@ietf.org>
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Hi Vamsi,
 
If you are considering to transport traditional SMS service over SIP, 3GPP
23.204/24.341 is dealing with this topic.
It is based on SIP MESSAGE request. You can get the specs from
http://www.3gpp.org/specs/specs.htm.
 
If you are considering implementing SMS-like service by SIP, the MESSAGE
request is ready for you.
 
Cheers.
Peili
 
  _____  

From: vamsi [mailto:annasagaram.vamsi@aftek.com] 
Sent: Friday, December 29, 2006 1:15 PM
To: 'IETF SIP List'
Cc: 'Balaji Murlitharan'
Subject: [Sip] SMS via VoIP


Hi, 
     Can any one give pointers where to start for SMS over VoIP.  Any RFCs?
Any pointer from where to start will be highly appreciable. Is the rfc 3761
appropriate for implmenting SMS via VoIP?
 
thanks & regards,
 vamsi.

--Boundary_(ID_xum2wmW5CQ4lTVq5l7IfSQ)
Content-type: text/html; charset=US-ASCII
Content-transfer-encoding: 7BIT

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=us-ascii">
<META content="MSHTML 6.00.2800.1555" name=GENERATOR></HEAD>
<BODY>
<DIV><SPAN class=153533507-29122006><FONT face=Arial size=2>Hi 
Vamsi,</FONT></SPAN></DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial size=2>If you are 
considering to transport traditional SMS service over SIP, 3GPP 23.204/24.341 is 
dealing with this topic.</FONT></SPAN></DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial size=2>It is based on SIP 
MESSAGE request. You can get the specs from <A 
href="http://www.3gpp.org/specs/specs.htm">http://www.3gpp.org/specs/specs.htm</A>.</FONT></SPAN></DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial size=2>If you are 
considering implementing SMS-like service by SIP, the MESSAGE request is ready 
for you.</FONT></SPAN></DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial 
size=2>Cheers.</FONT></SPAN></DIV>
<DIV><SPAN class=153533507-29122006><FONT face=Arial 
size=2>Peili</FONT></SPAN></DIV>
<DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV>
<DIV class=OutlookMessageHeader lang=zh-cn dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> vamsi [mailto:annasagaram.vamsi@aftek.com] 
<BR><B>Sent:</B> Friday, December 29, 2006 1:15 PM<BR><B>To:</B> 'IETF SIP 
List'<BR><B>Cc:</B> 'Balaji Murlitharan'<BR><B>Subject:</B> [Sip] SMS via 
VoIP<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=ltr align=left><FONT face=Arial size=2>
<DIV 
style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">
<DIV><FONT color=#0000ff>Hi, </FONT></DIV>
<DIV><FONT color=#0000ff>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Can any one give pointers 
where to start for SMS over VoIP.&nbsp; Any RFCs? Any pointer from where to 
start will be highly appreciable.&nbsp;<SPAN class=031111005-29122006>Is the rfc 
3761 appropriate for implmenting SMS via VoIP?</SPAN></FONT></DIV>
<DIV><FONT color=#0000ff></FONT>&nbsp;</DIV>
<DIV><FONT color=#0000ff>thanks &amp; regards,</FONT></DIV>
<DIV><FONT color=#0000ff>&nbsp;<SPAN 
class=031111005-29122006>vamsi.</SPAN></FONT></DIV></DIV></FONT></DIV></BODY></HTML>

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1643778392==--




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From: =?iso-8859-1?Q?Fatih_Ey=FCp_NAR?= <fenar@yahoo.com>
Subject: Re: [Sip] SMS via VoIP
To: vamsi <annasagaram.vamsi@aftek.com>, Dale.Worley@comcast.net
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The Message Session Relay Protocol (MSRP)=0A=0Ahttp://www.ietf.org/internet=
-drafts/draft-ietf-simple-message-sessions-18.txt=0A=0A Message Session Rel=
ay Protocol; a  protocol for transmitting a series of related instant messa=
ges in the context of a session.  Message sessions are treated like any oth=
er media stream when set up via a rendezvous or session creation protocol s=
uch as the Session Initiation Protocol.=0A=0A m=3Dmessage 7654 TCP/MSRP *=
=0A=0A=0A----- Original Message ----=0AFrom: vamsi <annasagaram.vamsi@aftek=
.com>=0ATo: Dale.Worley@comcast.net=0ACc: IETF SIP List <sip@ietf.org>=0ASe=
nt: Friday, December 29, 2006 9:58:39 AM=0ASubject: RE: [Sip] SMS via VoIP=
=0A=0A=0AHi Dale,=0A     Thanks for the message.  But for MESSAGE conversat=
ion, what kind of=0Acodec is used, that is audio codec or any other specifi=
ed codec?  As per RFC=0A3551 Audio and video codecs are specified.  =0A=0AT=
hanks & regards,=0AAnnasagaram vamsi. =0A           =0A=0A-----Original Mes=
sage-----=0AFrom: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net] =
=0ASent: Friday, December 29, 2006 10:44 AM=0ATo: sip@ietf.org=0ASubject: R=
e: [Sip] SMS via VoIP=0A=0A   From: "vamsi" <annasagaram.vamsi@aftek.com>=
=0A=0A   Can any one give pointers where to start for SMS over VoIP.  Any=
=0A   RFCs?  Any pointer from where to start will be highly=0A   appreciabl=
e. Is the rfc 3761 appropriate for implmenting SMS via=0A   VoIP?=0A=0AENUM=
 (RFC 3761) is probably a good basis for addressing SMS messages.=0AThe MES=
SAGE method seems to be the best way to send a stand-alone message to=0Aa S=
IP URI.  But for message conversations, there is a "codec" type for SDP=0Af=
or IM-like communications that should usually be used.=0A=0ADale=0A=0A_____=
__________________________________________=0ASip mailing list  https://www1=
.ietf.org/mailman/listinfo/sip=0AThis list is for NEW development of the co=
re SIP Protocol Use=0Asip-implementors@cs.columbia.edu for questions on cur=
rent sip Use=0Asipping@ietf.org for new developments on the application of =
sip=0A=0A=0A=0A=0A_______________________________________________=0ASip mai=
ling list  https://www1.ietf.org/mailman/listinfo/sip=0AThis list is for NE=
W development of the core SIP Protocol=0AUse sip-implementors@cs.columbia.e=
du for questions on current sip=0AUse sipping@ietf.org for new developments=
 on the application of sip

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
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From: =?iso-8859-1?Q?Fatih_Ey=FCp_NAR?= <fenar@yahoo.com>
Subject: Re: [Sip] Transport used by SIP UA
To: Adarsh Guler <adarshg@TechMahindra.com>, sip@ietf.org
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--0-1617520466-1167383187=:60079
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RFC3261  18.2.1:=0A-------------------------------=0A"A server SHOULD be pr=
epared to receive requests on any IP address, port and transport combinatio=
n that can be the result of a DNS lookup on a SIP or SIPS URI [4] that is h=
anded out for the purposes of communicating with that server. "=0A=0A=0A---=
-- Original Message ----=0AFrom: Adarsh Guler <adarshg@TechMahindra.com>=0A=
To: sip@ietf.org=0ASent: Friday, December 29, 2006 8:35:17 AM=0ASubject: [S=
ip] Transport used by SIP UA=0A=0A=0AHI=0A =0AIf an UAC is configured to us=
e the UDP port and a SIP Proxy is configured to use TCP port.=0AThe UAC is =
keep sending messages in UDP port, but Sip Proxy don=92t respond, since it =
is listening on TCP port.=0A =0AIn this scenario, how UAC should know that =
Proxy is listening on TCP Port =85.???=0A =0AThanks=0AAdarsh.=0A =0A_______=
________________________________________=0ASip mailing list  https://www1.i=
etf.org/mailman/listinfo/sip=0AThis list is for NEW development of the core=
 SIP Protocol=0AUse sip-implementors@cs.columbia.edu for questions on curre=
nt sip=0AUse sipping@ietf.org for new developments on the application of si=
p
--0-1617520466-1167383187=:60079
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<html><head><style type=3D"text/css"><!-- DIV {margin:0px;} --></style></he=
ad><body><div style=3D"font-family:times new roman, new york, times, serif;=
font-size:12pt"><DIV style=3D"FONT-SIZE: 12pt; FONT-FAMILY: times new roman=
, new york, times, serif">RFC3261&nbsp; 18.2.1:</DIV>=0A<DIV style=3D"FONT-=
SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">---------=
----------------------</DIV>=0A<DIV style=3D"FONT-SIZE: 12pt; FONT-FAMILY: =
times new roman, new york, times, serif">"A server SHOULD be prepared to re=
ceive requests on any IP address, port and transport combination that can b=
e the result of a DNS lookup on a SIP or SIPS URI [4] that is handed out fo=
r the purposes of communicating with that server.&nbsp;"</DIV>=0A<DIV style=
=3D"FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">=
<BR>&nbsp;</DIV>=0A<DIV style=3D"FONT-SIZE: 12pt; FONT-FAMILY: times new ro=
man, new york, times, serif">----- Original Message ----<BR>From: Adarsh Gu=
ler &lt;adarshg@TechMahindra.com&gt;<BR>To: sip@ietf.org<BR>Sent: Friday, D=
ecember 29, 2006 8:35:17 AM<BR>Subject: [Sip] Transport used by SIP UA<BR><=
BR>=0A<STYLE>=0A<!--=0A=0A /* Style Definitions */=0A p.MsoNormal, li.MsoNo=
rmal, div.MsoNormal=0A=09{margin:0in;=0Amargin-bottom:.0001pt;=0Afont-size:=
12.0pt;=0Afont-family:"Times New Roman";}=0Aa:link, span.MsoHyperlink=0A=09=
{color:blue;=0Atext-decoration:underline;}=0Aa:visited, span.MsoHyperlinkFo=
llowed=0A=09{color:purple;=0Atext-decoration:underline;}=0Aspan.EmailStyle1=
7=0A=09{=0Afont-family:Arial;=0Acolor:windowtext;}=0A _filtered {=0Amargin:=
1.0in 1.25in 1.0in 1.25in;}=0Adiv.Section1=0A=09{}=0A-->=0A</STYLE>=0A=0A<D=
IV class=3DSection1>=0A<P class=3DMsoNormal><FONT face=3DArial size=3D2><SP=
AN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">HI</SPAN></FONT></P>=0A<P =
class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE: 10p=
t; FONT-FAMILY: Arial">&nbsp;</SPAN></FONT></P>=0A<P class=3DMsoNormal><FON=
T face=3DArial size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial"=
>If an UAC is configured to use the UDP port and a SIP Proxy is configured =
to use TCP port.</SPAN></FONT></P>=0A<P class=3DMsoNormal><FONT face=3DAria=
l size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">The UAC is k=
eep sending messages in UDP port, but Sip Proxy don=92t respond, since it i=
s listening on TCP port.</SPAN></FONT></P>=0A<P class=3DMsoNormal><FONT fac=
e=3DArial size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbs=
p;</SPAN></FONT></P>=0A<P class=3DMsoNormal><FONT face=3DArial size=3D2><SP=
AN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">In this scenario, how UAC =
should know that Proxy is listening on TCP Port =85.???</SPAN></FONT></P>=
=0A<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZ=
E: 10pt; FONT-FAMILY: Arial">&nbsp;</SPAN></FONT></P>=0A<P class=3DMsoNorma=
l><FONT face=3DArial size=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
Arial">Thanks</SPAN></FONT></P>=0A<P class=3DMsoNormal><FONT face=3DArial s=
ize=3D2><SPAN style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">Adarsh.</SPAN><=
/FONT></P>=0A<P class=3DMsoNormal><FONT face=3DArial size=3D2><SPAN style=
=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial">&nbsp;</SPAN></FONT></P></DIV>=0A<=
DIV>_______________________________________________<BR>Sip mailing list&nbs=
p;&nbsp;<A href=3D"https://www1.ietf.org/mailman/listinfo/sip" target=3D_bl=
ank>https://www1.ietf.org/mailman/listinfo/sip</A><BR>This list is for NEW =
development of the core SIP Protocol<BR>Use sip-implementors@cs.columbia.ed=
u for questions on current sip<BR>Use sipping@ietf.org for new developments=
 on the application of sip</DIV></DIV>=0A<DIV style=3D"FONT-SIZE: 12pt; FON=
T-FAMILY: times new roman, new york, times, serif"><BR></DIV></div></body><=
/html>
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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
--===============1975864528==--




From sip-bounces@ietf.org Fri Dec 29 23:06:17 2006
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	(adarshg@TechMahindra.com)
Subject: Re: [Sip] Transport used by SIP UA
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   From: "Adarsh Guler" <adarshg@TechMahindra.com>

   If an UAC is configured to use the UDP port and a SIP Proxy is
   configured to use TCP port.

   The UAC is keep sending messages in UDP port, but Sip Proxy don't
   respond, since it is listening on TCP port.

   In this scenario, how UAC should know that Proxy is listening on
   TCP Port....???

If the UAC decides to send a message to a SIP proxy, it must have
learned, by some method, the SIP proxy's host address, port number,
and protocol on which it is listening.  This information is derived
from the SIP URI which the UAC attempts to send to.  The process is
described in RFC 3263.

Dale

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This list is for NEW development of the core SIP Protocol
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Use sipping@ietf.org for new developments on the application of sip



From sip-bounces@ietf.org Fri Dec 29 23:10:38 2006
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	(annasagaram.vamsi@aftek.com)
Subject: Re: [Sip] SMS via VoIP
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   From: "vamsi" <annasagaram.vamsi@aftek.com>

   Thanks for the message.  But for MESSAGE conversation, what kind of
   codec is used, that is audio codec or any other specified codec?
   As per RFC 3551 Audio and video codecs are specified.

A MESSAGE request contains the text content directly.  See RFC 3438
for details.  (I don't think anyone has specified how to carry
recorded audio segments as messages.)

For extended interchange of messages, as Fenar says, you set up a
dialog (using INVITE) and specify the use of MSRP in the SDP.

As others have noted, if you want to interoperage with 3GPP, its
specificaitons are available.

Dale

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip



From Simpson@1st-bad-credit-home-loans.com Sat Dec 30 03:59:50 2006
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From sip-bounces@ietf.org Sat Dec 30 16:24:46 2006
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From: Charles Kristofek <ckristofek@airnetcom.com>
To: "'sip@ietf.org'" <sip@ietf.org>
Subject: RE: [Sip] SMS via VoIP
Date: Sat, 30 Dec 2006 16:22:19 -0500
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There are 2 models in SIP for sending IMs (paging and session).

Paging mode has the UA client sending the MESSAGE method to pass the
information. Please refer to RFC 3428.

Session mode has the UA client sending the INVITE method to establish a
session first.

For SMS via SIP, the MESSAGE method carrying the text
within the body of the method to the SMSC (or application server)
mimics 2G.

The establishment of a session via INVITE mimics IM where
peer information can be relayed, such as "typing".

Chuck K.

------------------------------
Date: Fri, 29 Dec 2006 23:10:27 -0500
From: Dale.Worley@comcast.net
Subject: Re: [Sip] SMS via VoIP
To: sip@ietf.org
Message-ID: <200612300410.kBU4ARqG023818@dragon.ariadne.com>

   From: "vamsi" <annasagaram.vamsi@aftek.com>

   Thanks for the message.  But for MESSAGE conversation, what kind of
   codec is used, that is audio codec or any other specified codec?
   As per RFC 3551 Audio and video codecs are specified.

A MESSAGE request contains the text content directly.  See RFC 3438
for details.  (I don't think anyone has specified how to carry
recorded audio segments as messages.)

For extended interchange of messages, as Fenar says, you set up a
dialog (using INVITE) and specify the use of MSRP in the SDP.

As others have noted, if you want to interoperage with 3GPP, its
specificaitons are available.

Dale



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From sip-bounces@ietf.org Sat Dec 30 19:48:14 2006
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Subject: Re: [Sip] Question about Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
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It took me some time to parse through this but sounds like you and I  
are in 100% agreement on what an acceptable solution would look like.


On Dec 20, 2006, at 12:18 AM, <Erkki.Koivusalo@nokia.com>  
<Erkki.Koivusalo@nokia.com> wrote:

>
> Hi Cullen,
>
>>> Many PSTN gateways and Softswitches always send large INVITEs for
>>> example,
>>> and they will always fragment. And I might add that many very  
>>> popular
>>> NAT/routers don't support UDP fragmentation.
>>
>> Hmm - I wish we had better specifics here - I will point out there
>> are millions of endpoints on well known residential endpoints that
>> are working through NATs and connected to PSTN gw and softwswiches
>> that are working over UDP.
>
> If so, then how would those endpoints get any benefit from Outbound ?
> What would be the value added for those deployments if those endpoints
> would be changed as Outbound UDP compliant endpoints ? Multiple
> registration support and new types of keepalive messages i.e. some
> added reliability ?
>
> But if those endpoints and proxies anyway have to be either upgraded
> or replaced for Outbound, then why couldn't that upgrade cover TCP
> support as well ? What is the ultimate benefit for using UDP instead
> of TCP for those deployments ?
>
>> Uh, no - I was saying that outbound over UDP is usable in cases where
>> the policy is to reject messages that are too large for UDP. I agree
>> that limits the functionality of the communications with the UA that
>> registered over this UDP only network but that was that deployments
>> choice.
>>
>> Note what I am talking about here is all about deployments using TCP
>> or UDP. I don't mind about if a UA has to implement both.
>
> Please remember that the proposal that you do not agree with was
> like this:
>
>> Proposal 2: Make outbound registrations over UDP flows optional for
>> UAs.
>
> Outbound over UDP would still be possible and probably supported by
> those UA vendors who get significant gain for it. But some UAs
> targetting to other deployments could just opt to support TCP.
> What is wrong with that ?
>
> If it is the deployments choice to limit the functionality of the
> communications with the UA registered, why could it not be possible
> for the specific closed deployments to administratively (or  
> technically)
> limit the deployed UAs to be such that implement UDP ? Why would you
> mandate ALL the implemented UAs to support UDP for Outbound ? Those
> deployments could just tell the users not to aqcuire any UAs which
> only support Outbound with TCP, if the proxies (or operator policy)
> does not support TCP (or does not support persistent TCP connections).
>
>>> You mean a UA will open two connections, one
>>> for UDP and another for TCP? I fail to see why on earth
>>> anybody would do this instead of just using a single TCP connection.
>>
>> I think the folks that favor these approach would use a scheme where
>> the UDP connection was long lived and the TCP connection was short
>> lived for the large messages. The advantages of this would be to get
>> the scaleability and reliability schemes that are easier with UDP yet
>> still be able to do large messages.
>
> Please remember the Outbound draft does make such a scheme possible !
> If the UA does NOT have a long lived persistent TCP connection
> available, there is no way for the proxy to get the UA to open
> such connection or for the proxy itself open such a TCP connection
> towards the UA via the NAT when a large message arrives destined
> to the UA. So if the UA does not keep the TCP connection open
> all the time, it has to live with the limitation of not being
> able to receive SIP requests bigger than MTU if the NAT discards
> fragmented UDP packets.
>
> P.S. Actually there is such a mechanism for that specified in Marc
> Petit-Huguenin's I-D "Preventing Fragmentation for Client Initiated
> Connections in the Session Initiation Protocol (SIP)" - but that
> is not within the baseline Outbound spec and there is no normative
> binding between those specs either.
>
> Regards,
>
> Erkki


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On Dec 24, 2006, at 3:53 PM, Dean Willis wrote:

>
> How about we rip out UDP, fix the Outbound stuff, fix the route- 
> construction stuff, fix the early-media stuff. and call it SIP 3.0?

While you are working on this, I think we should publish outbound? It  
solves a useful goal now and probably works towards the right  
direction of what you would propose as a replace for SIP 2.0.


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On Dec 18, 2006, at 2:04 PM, Francois Audet wrote:

>
> I'm ok with having outbound support STUN-over-UDP, as long as we are
> honest about
> the implications regarding fragmentation. This could be as simple as
> explaining that
> if you don't keep a TCP channel opened, request that are bigger  
> than MTU
> and would
> observe the rules of RFC 3261/18.1.1 would not be able to be received.
>

Yes 100% agree - we should explain the bad implications of not having  
a TCP channel open.


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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [Sip] Question about Poll: Proposal relating to keepalive, TCP,
	and UDP usage in draft-ietf-sip-outbound
Date: Sat, 30 Dec 2006 17:00:52 -0800
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If the people who have been expressing strong opinions on the details  
of this proposal could comment on if they agree with these high level  
goals or not, it would be really helpful for trying to get this draft  
to move forward.

On Dec 16, 2006, at 12:33 PM, Cullen Jennings wrote:

> Practically speaking given where we are today, I would be a fan of  
> a solution that meant the following goals:
>
> 1) allowed deployments that wanted to use TCP to use TCP
>
> 2) allowed deployments that wanted to use UDP to use only UDP with  
> the known limitations that the UA would not be able to send large  
> requests
>
> 3) allowed deployments that wanted to use both UDP and TCP to do so  
> and not have the size limitations by using switchover

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I agree 100% with #1 and #2. I find #3 to be of dubious value, although 
I see no harm in a solution that has that property as well (and you seem 
to get it for free if you satisfy #1 and #2).

/a



Cullen Jennings wrote:
>
> If the people who have been expressing strong opinions on the details 
> of this proposal could comment on if they agree with these high level 
> goals or not, it would be really helpful for trying to get this draft 
> to move forward.
>
> On Dec 16, 2006, at 12:33 PM, Cullen Jennings wrote:
>
>> Practically speaking given where we are today, I would be a fan of a 
>> solution that meant the following goals:
>>
>> 1) allowed deployments that wanted to use TCP to use TCP
>>
>> 2) allowed deployments that wanted to use UDP to use only UDP with 
>> the known limitations that the UA would not be able to send large 
>> requests
>>
>> 3) allowed deployments that wanted to use both UDP and TCP to do so 
>> and not have the size limitations by using switchover
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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I concur with Adam, with the caveat that any unforseen complexities  
that might be introduced by the phrase "not have the size limitations  
by using switchover" not block the work. That is, if 3 is free, or  
cheap, then great--but I would not want to significantly delay 1 and  
2 in order to get 3.

On Dec 30, 2006, at 7:06 PM, Adam Roach wrote:

> I agree 100% with #1 and #2. I find #3 to be of dubious value,  
> although I see no harm in a solution that has that property as well  
> (and you seem to get it for free if you satisfy #1 and #2).
>
> /a
>
>
>
> Cullen Jennings wrote:
>>
>> If the people who have been expressing strong opinions on the  
>> details of this proposal could comment on if they agree with these  
>> high level goals or not, it would be really helpful for trying to  
>> get this draft to move forward.
>>
>> On Dec 16, 2006, at 12:33 PM, Cullen Jennings wrote:
>>
>>> Practically speaking given where we are today, I would be a fan  
>>> of a solution that meant the following goals:
>>>
>>> 1) allowed deployments that wanted to use TCP to use TCP
>>>
>>> 2) allowed deployments that wanted to use UDP to use only UDP  
>>> with the known limitations that the UA would not be able to send  
>>> large requests
>>>
>>> 3) allowed deployments that wanted to use both UDP and TCP to do  
>>> so and not have the size limitations by using switchover
>>
>> _______________________________________________
>> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
>> This list is for NEW development of the core SIP Protocol
>> Use sip-implementors@cs.columbia.edu for questions on current sip
>> Use sipping@ietf.org for new developments on the application of sip
>
>
> _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core SIP Protocol
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sipping@ietf.org for new developments on the application of sip


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In RFC 3428, section 3, " MESSAGE requests do not establish dialogs".

 Does "Paging mode" mean that with out establishing a actual/voice session,
UA client can send IM to UA server?  


-----Original Message-----
From: Charles Kristofek [mailto:ckristofek@airnetcom.com] 
Sent: Sunday, December 31, 2006 2:52 AM
To: 'sip@ietf.org'
Subject: RE: [Sip] SMS via VoIP



There are 2 models in SIP for sending IMs (paging and session).

Paging mode has the UA client sending the MESSAGE method to pass the
information. Please refer to RFC 3428.

Session mode has the UA client sending the INVITE method to establish a
session first.

For SMS via SIP, the MESSAGE method carrying the text within the body of the
method to the SMSC (or application server) mimics 2G.

The establishment of a session via INVITE mimics IM where peer information
can be relayed, such as "typing".

Chuck K.

------------------------------
Date: Fri, 29 Dec 2006 23:10:27 -0500
From: Dale.Worley@comcast.net
Subject: Re: [Sip] SMS via VoIP
To: sip@ietf.org
Message-ID: <200612300410.kBU4ARqG023818@dragon.ariadne.com>

   From: "vamsi" <annasagaram.vamsi@aftek.com>

   Thanks for the message.  But for MESSAGE conversation, what kind of
   codec is used, that is audio codec or any other specified codec?
   As per RFC 3551 Audio and video codecs are specified.

A MESSAGE request contains the text content directly.  See RFC 3438 for
details.  (I don't think anyone has specified how to carry recorded audio
segments as messages.)

For extended interchange of messages, as Fenar says, you set up a dialog
(using INVITE) and specify the use of MSRP in the SDP.

As others have noted, if you want to interoperage with 3GPP, its
specificaitons are available.

Dale



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


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Yes, the UAC can send/convey information one way to the UAS.
This information flow feels like sending a message to a pager or 
sending an sms to a mobile.

Previous discussion on instant message, SMS, MMS interworking:

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

Message: 1
Date: Thu, 25 May 2006 15:37:15 +0200
From: "Jesus Javier Arauz \(E2/EEM\)"
      <jesus.javier.arauz@ericsson.com>
Subject: [Sipping] Re: Multi-media messaging and SIP
To: <sipping@ietf.org>
Message-ID:
     <33D71F0FC0684C4C8A3BB76989EDED9C01C2138D@eesmdmw020.eemea.ericsson.se>
Content-Type: text/plain;     charset="us-ascii"

 OK, forget it. It's already done:
draft-ietf-sipping-uri-list-conferencing-07

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

Message: 2
Date: Mon, 22 May 2006 12:04:03 +0200
From: "Jesus Javier Arauz \(E2/EEM\)"
      <jesus.javier.arauz@ericsson.com>
Subject: [Sipping] Multi-media messaging and SIP
To: <sipping@ietf.org>
Message-ID:
<33D71F0FC0684C4C8A3BB76989EDED9C01B82B3B@eesmdmw020.eemea.ericsson.se>
Content-Type: text/plain; charset="us-ascii"

Hi,
In the SIP messaging specifications, the usage of MESSAGE is limited to
small, short messages while for longer messages like e.g. multi-media
messages including video, pictures etc. MSRP must be used instead.

MESSAGE has been retro-fitted with a capability that enables a user to
specify multiple destinations for a short message (see
draft-ietf-sipping-uri-list-message-07). This capabilty needs a messaging
server that explodes the list of destinations and sends the short message to
each destination. However, SIP-based MSRP does not define that capability.
The INVITE that initiates the MSRP session cannot be targeted to mutiple
invitees by the inviting user. The UA must initiate the sessions
individually to each destination and then send the message to all the
destinations. This may lead to either more complex messaging client
implementations in UA or cumbersome usage in comparison to the short
messaging client.

One case where this lack of capability is notorious is in SMS/MMS/IMPS
interworking with IMS. When a GSM user sends an SMS to multiple destinations
this can be directly mapped by the interworking function to a MESSAGE
request including all the destinations in the request.

However, when a GPRS user sends an MMS to multiple destinations, the
interworking function must take care of initiating the SIP and MSRP
sessions, send the message over all sessions and then releasing those
sessions. The same goes for IMPS.

I think it would be advantageous to retro-fit session-based messaging with
the same capability as short messaging. One possible alternative would be to
allow the same MIME body defined in
draft-ietf-sipping-uri-list-message-07 in INVITE requests for MSRP sessions.
However this may not seem appropriate since it impacts a rather basic
message of the SIP specification. Any other alternative would require
defining a new mechanism specific for long, multi-media messages, like e.g.
content indirection.

I wonder if there've been some discussions on this issue already? What's the
oppinion of the sipping community on it?

Best regards,
/Javier Arauz
------------------------------
Message: 3
Date: Mon, 22 May 2006 09:34:56 -0400
From: "Dale R. Worley" <dworley@pingtel.com>
Subject: Re: [Sipping] Multi-media messaging and SIP
To: Sipping <sipping@ietf.org>
Message-ID: <1148304897.11977.10.camel@niagra.pingtel.com>
Content-Type: text/plain

On Mon, 2006-05-22 at 12:04 +0200, Jesus Javier Arauz (E2/EEM) wrote:
> MESSAGE has been retro-fitted with a capability that enables a user to
> specify multiple destinations for a short message (see draft-ietf- 
> sipping-uri-list-message-07). This capabilty needs a messaging server 
> that explodes the list of destinations and sends the short message to 
> each destination. However, SIP-based MSRP does not define that 
> capability. The INVITE that initiates the MSRP session cannot be 
> targeted to mutiple invitees by the inviting user. The UA must 
> initiate the sessions individually to each destination and then send 
> the message to all the destinations. This may lead to either more 
> complex messaging client implementations in UA or cumbersome usage in 
> comparison to the short messaging client.

I'm not familiar with the details of MSRP, but I do not see why a forked
INVITE cannot carry an MSRP set-up offer to more than one destination, each
of which will reply with an MSRP set-up answer, thus setting up a set of
MSRP sessions.  (In exactly the same way that a single INVITE can establish
multiple SDP sessions.)

Oops, well, yes, you have to make sure the proxies that do the forking do
not attempt to cancel all-but-one successful branch of the fork.  But that's
a known problem that applies to a lot more situations than setting up MSRP
sessions.  See, e.g., draft-worley-sipping-forking-00 for a discussion,
especially section 3.6.

Dale
------------------------------
Message: 5
Date: Mon, 22 May 2006 11:23:47 -0400
From: Richard Barnes <rbarnes@bbn.com>
Subject: Re: [Sipping] Multi-media messaging and SIP
To: "Jesus Javier Arauz (E2/EEM)" <jesus.javier.arauz@ericsson.com>
Cc: sipping@ietf.org
Message-ID: <4471D783.8080407@bbn.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

> However, when a GPRS user sends an MMS to multiple destinations, the 
> interworking function must take care of initiating the SIP and MSRP 
> sessions, send the message over all sessions and then releasing those 
> sessions. The same goes for IMPS.

Beacuse MSRP is inherently session-oriented, I don't think you're going to
get around having to initiate and release SIP/MSRP sessions.

To simplify the internetworking function, perhaps what needs to be done is
to define a way for MSRP relays to act as URI-list services.  That way, the
internetworking function could:
1. Send a forked INVITE to all recipients 2. Gather responses and construct
URI-list 3. Send message and URI-list to URI-list Service The URI-list
Service could then forward the message to all the recipients and relay MSRP
REPORTs back to the internetworking function, who could then send BYE
requests to close the SIP sessions.  Given the constraints that MSRP is
session-oriented (hence requires initiation/release), I think that's about
as simple as it can get.
(Actually, adding such a URI-list capability to MSRP relays seems like a
good idea in general, since it would simplify the implementation of
multi-party chat.)

Hope this helps,
--Richard
> 
> I think it would be advantageous to retro-fit session-based messaging 
> with the same capability as short messaging. One possible alternative 
> would be to allow the same MIME body defined in
> draft-ietf-sipping-uri-list-message-07 in INVITE requests for MSRP 
> sessions. However this may not seem appropriate since it impacts a 
> rather basic message of the SIP specification. Any other alternative 
> would require defining a new mechanism specific for long, multi-media 
> messages, like e.g. content indirection.
> 
> I wonder if there've been some discussions on this issue already? 
> What's the oppinion of the sipping community on it?
> 
> Best regards,
> /Javier Arauz
> 
> 
> 
> 
> ----------------------------------------------------------------------
End of Sipping Digest, Vol 25, Issue 28

***************************************

-----Original Message-----
From: vamsi [mailto:annasagaram.vamsi@aftek.com] 
Sent: Sunday, December 31, 2006 7:13 AM
To: 'Charles Kristofek'; sip@ietf.org
Subject: RE: [Sip] SMS via VoIP

 
In RFC 3428, section 3, " MESSAGE requests do not establish dialogs".

 Does "Paging mode" mean that with out establishing a actual/voice session,
UA client can send IM to UA server?  


-----Original Message-----
From: Charles Kristofek [mailto:ckristofek@airnetcom.com] 
Sent: Sunday, December 31, 2006 2:52 AM
To: 'sip@ietf.org'
Subject: RE: [Sip] SMS via VoIP



There are 2 models in SIP for sending IMs (paging and session).

Paging mode has the UA client sending the MESSAGE method to pass the
information. Please refer to RFC 3428.

Session mode has the UA client sending the INVITE method to establish a
session first.

For SMS via SIP, the MESSAGE method carrying the text within the body of the
method to the SMSC (or application server) mimics 2G.

The establishment of a session via INVITE mimics IM where peer information
can be relayed, such as "typing".

Chuck K.

------------------------------
Date: Fri, 29 Dec 2006 23:10:27 -0500
From: Dale.Worley@comcast.net
Subject: Re: [Sip] SMS via VoIP
To: sip@ietf.org
Message-ID: <200612300410.kBU4ARqG023818@dragon.ariadne.com>

   From: "vamsi" <annasagaram.vamsi@aftek.com>

   Thanks for the message.  But for MESSAGE conversation, what kind of
   codec is used, that is audio codec or any other specified codec?
   As per RFC 3551 Audio and video codecs are specified.

A MESSAGE request contains the text content directly.  See RFC 3438 for
details.  (I don't think anyone has specified how to carry recorded audio
segments as messages.)

For extended interchange of messages, as Fenar says, you set up a dialog
(using INVITE) and specify the use of MSRP in the SDP.

As others have noted, if you want to interoperage with 3GPP, its
specificaitons are available.

Dale



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


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