[p2p-sip] Some comments on Use-cases document

huang-ming.pan at comcast.net (Peter Pan) Tue, 21 March 2006 21:20 UTC

From: "huang-ming.pan at comcast.net"
Date: Tue, 21 Mar 2006 13:20:18 -0800
Subject: [p2p-sip] Some comments on Use-cases document
References: <200603210319.k2L3JIrj001729@cs.columbia.edu><B225AB16-9E9C-4A5F-A3C1-97309B78A920@magma.ca><e9132f820603211202k3024d6b1x548400900a75a742@mail.gmail.com> <e9132f820603211241m7a43eab4q3f4f3cf3530052c3@mail.gmail.com>
Message-ID: <010b01c64d2d$41c340a0$6400a8c0@comcast.net>

> The exact DHT algorithm we implement, the exact way the we use SIP,
> the methods, etc I think are all going to need some careful thought.
> But for P2P SIP, I think SIP is the obvious choice for DHT operations,
> and I would consider it the default unless a convincing argument is
> made against it.

Not an argument but a question: how to efficiently implement Kademlia
STORE_VALUE primitive in SIP INVITE/REGISTER when the stored values can be
of any length and type?

peter

*** dont bash me, we are of the same country ***

----- Original Message -----
From: "Bruce Lowekamp" <lowekamp at cs.wm.edu>
To: "Philip Matthews" <philip_matthews at magma.ca>
Cc: "P2P-SIP" <p2p-sip at cs.columbia.edu>
Sent: Tuesday, March 21, 2006 12:41 PM
Subject: Re: [p2p-sip] Some comments on Use-cases document


> resending since this hasn't been acked by the mailing list or appeared
> after over 30 minutes.  (I'm sure it will now immediately appear)
>
> On 3/21/06, Bruce Lowekamp <lowekamp at cs.wm.edu> wrote:
> > I don't thnk it has anything to do with use-cases, either
> >
> > but
> >
> > Building DHT maintenance on top of SIP gives us all of the advantages
> > of the routing, addressing, naming, and security issues already built
> > into SIP.  Plus the NAT traversal capabilities of STUN, TURN, and ICE.
> >  While not perfect for our use, I think with very minimal
> > modifications (whatever the final protocol is) they will provide a
> > very good solution.
> >
> > A proposal to use something else needs to:
> > - explain what the shortcomings of SIP are for this purpose
> > - explain how a new or different solution will provide equivalent
functionality
> > - explain how/why the new protocol will be better after resolving the
> > complexities that SIP has become so complex to address
> > - make a convincing enough case to justify deploying devices (and
> > we're frequently talking about very small devices) to implement two
> > separate protocol stacks.
> >
> >
> > The exact DHT algorithm we implement, the exact way the we use SIP,
> > the methods, etc I think are all going to need some careful thought.
> > But for P2P SIP, I think SIP is the obvious choice for DHT operations,
> > and I would consider it the default unless a convincing argument is
> > made against it.
> >
> > A number of comments have been made stating that NATs must be taken
> > into account from the beginning.  Again, one of the issues that using
> > SIP helps address already is NAT traversal.  It's not perfect for
> > signalling, but the framework is there.
> >
> > Bruce
> >
> > On 3/21/06, Philip Matthews <philip_matthews at magma.ca> wrote:
> > > On 20-Mar-06, at 21:19 , David Barrett wrote:
> > >
> > > > I'd add to that this key question:
> > > >
> > > >       "Will we extend SIP, or create a new protocol?"
> > > >
> > > > I'm finding it hard to make any headway without understanding the
> > > > above.
> > > > Basically, I see two major directions we could go:
> > > >
> > > > 1) Extend SIP with overlay-maintenance and resource-location
messages.
> > > >
> > > > 2) Create a new overlay protocol and develop bindings for SIP and
> > > > ICE (eg,
> > > > distributed proxy service and STUN/TURN resource-location service).
> > > >
> > > > It seems this high-level decision keeps coming up again and again in
> > > > discussions of the smaller issues.
> > > >
> > >
> > > For what it is worth, I can say that my own views on this subject
> > > have changed.
> > >
> > > For all of last year, I strongly believed that there should be two
> > > distinct layers:
> > > a SIP layer and a P2P layer (i.e., option 2). See the arguments I
> > > wrote in
> > >    http://www.p2psip.org/drafts/draft-matthews-sipping-p2p-industrial-
> > > strength-00.txt
> > > as well as those on Alan Johnston in
> > >
http://www.p2psip.org/drafts/draft-johnston-sipping-p2p-ipcom-01.txt
> > >
> > > However, in the last few months, I have come to see that there are
> > > some good reasons
> > > to put the two layers together into one (i.e., option 1):
> > >         a) NAT Traversal becomes easier because there is just one
port,
> > > rather than two
> > >            (this assumes that the P2P layer would run on a different
port
> > > than SIP)
> > >         b) Possible performance improvements. With two layers, you
have to
> > > first ask
> > >           "where is user U?" and then send the Invite message. With
one
> > > layer, there is
> > >           the possibility of sending the Invite and having it routed
to the
> > > user.
> > >           (David et al removed this from their latest draft, but this
was an
> > > option in the
> > >           earlier version, and also in the work done by Henning's
group).
> > >
> > > As others have pointed out, there are also drawbacks, so I haven't
> > > concluded anything
> > > yet, but I am a lot more open to option 1 than I was a few months ago.
> > >
> > >
> > > - Philip
> > >
> > > PS. What this has to do with the use-cases document, however, I am
> > > not clear on  ;-)
> > > _______________________________________________
> > > p2p-sip mailing list
> > > p2p-sip at cs.columbia.edu
> > > https://lists.cs.columbia.edu/cucslists/listinfo/p2p-sip
> > >
> > >
> >
>
>
>