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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 2824

        Title:	    Call Processing Language Framework and
                    Requirements 
        Author(s):  J. Lennox, H. Schulzrinne 
        Status:     Informational
	Date:       May 2000
        Mailbox:    lennox@cs.columbia.edu,
                    schulzrinne@cs.columbia.edu 
        Pages:      25
        Characters: 58711
        Updates/Obsoletes/SeeAlso:  None

        I-D Tag:    draft-ietf-iptel-cpl-framework-02.txt

        URL:        ftp://ftp.isi.edu/in-notes/rfc2824.txt


A large number of the services we wish to make possible for Internet
telephony require fairly elaborate combinations of signalling
operations, often in network devices, to complete. We want a simple
and standardized way to create such services to make them easier to
implement and deploy.  This document describes an architectural
framework for such a mechanism, which we call a call processing
language. It also outlines requirements for such a language.

This document is a product of the IP Telephony Working Group of the
IETF.

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not specify an Internet standard of any kind.  Distribution of this
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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the IP Telephony Working Group of the IETF.

	Title		: Telephony Routing over IP (TRIP)
	Author(s)	: J. Rosenberg,  H. Salama, M. Squire	
        Filename	: draft-ietf-iptel-trip-02.txt
	Pages		: 71
	Date		: 31-May-00
	
This document presents the Telephony Routing over IP (TRIP). TRIP is
a policy driven inter-administrative domain protocol for advertising
the reachability of telephony destinations between location servers,
and for advertising attributes of the routes to those destinations.
TRIPÆs operation is independent of any signaling protocol,hence TRIP
can serve as the telephony routing protocol for any signaling
protocol.
The Border Gateway Protocol (BGP-4) is used to distribute routing
information between administrative domains. TRIP is used to
distribute telephony routing information between telephony
administrative domains. The similarity between the two protocols is
obvious, and hence TRIP is modeled after BGP-4.

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From iptel-admin@lists.bell-labs.com  Fri Jun  2 09:36:36 2000
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From: Dimitris Terzis <D.Terzis@cs.ucl.ac.uk>
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Subject: [IPTEL] TRIP questions
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Hi there...

I am trying to figure out how the establishment of the end-to-end paths
for signalling and media will happen, in an IP Telephony network using
TRIP for call routing.

Given the generalised definition of a VoIP gateway, as provided by
MEGACO (RFC 2805 etc.), the Signalling Gateway, the Media Gateway
and the Media Gateway Controller can be completely independent
components, even substantially spaced apart.

Suppose that I run TRIP. Then, at each hop, the relative LS can give me the
NextHopServer, which (presumably) is the signalling server of the protocol
defining this route (destination), apart from the first hop, where the
enquiry may come from the user terminal (e.g., a SIP UAC). This way,
signalling messages can reach the other end of the path, which can be
several hops away from the caller.

I am not clear as to what the implied behaviour, AFTER the
acceptance of the call at the signalling level, will be. How is the media
path
determined? Not by TRIP, of course, but is TRIP's position that "we just
assist in finding the signalling path, then let the signalling servers deal
with
the establishment of the media path"?

Also, what is the anticipated number of gateways participating in a
GSTN-to-IP call, for the TRIP routing model? Is it always one per different
network cloud traversed ("The selection of an egress gateway for a telephony
call, traversing an IP network towards an ultimate destination in the PSTN"
(and vice versa), as mentioned in Section 3 (Introduction))?

Finally, at the start of Section 3, the text reads "TRIP can be used to
exchange attributes necessary to enforce policies and to select call routes
based on path or gateway characteristics." Is it at all envisaged that the
advertised (signalling) routes will/could also contain media-realted
attributes (e.g., codecs) of a gateway (for example, using the
Capability Information parameter)?

Thanks for the attention.

Regards,

Dimitris

---------------------------------------------------------------
    Dimitris Terzis
    PhD Candidate - Department of Computer Science, UCL
    E-Mail: D.Terzis@cs.ucl.ac.uk
    URL : http://www.cs.ucl.ac.uk/staff/D.Terzis/
---------------------------------------------------------------





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From iptel-admin@lists.bell-labs.com  Fri Jun  2 12:11:33 2000
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Dimitris Terzis wrote:
> 
> Hi there...
> 
> I am trying to figure out how the establishment of the end-to-end paths
> for signalling and media will happen, in an IP Telephony network using
> TRIP for call routing.
> 
> Given the generalised definition of a VoIP gateway, as provided by
> MEGACO (RFC 2805 etc.), the Signalling Gateway, the Media Gateway
> and the Media Gateway Controller can be completely independent
> components, even substantially spaced apart.
> 
> Suppose that I run TRIP. Then, at each hop, the relative LS can give me the
> NextHopServer, which (presumably) is the signalling server of the protocol
> defining this route (destination), apart from the first hop, where the
> enquiry may come from the user terminal (e.g., a SIP UAC). This way,
> signalling messages can reach the other end of the path, which can be
> several hops away from the caller.
> 
> I am not clear as to what the implied behaviour, AFTER the
> acceptance of the call at the signalling level, will be. How is the media
> path
> determined?

The purpose of the signaling messages is to exchange, between end
systems (PCs, softphones, softswitches, or whatever), the IP addresses
to use for media. Thus, "routing" of media itself is handled directly by
the IP network. 


> Not by TRIP, of course, but is TRIP's position that "we just
> assist in finding the signalling path, then let the signalling servers deal
> with
> the establishment of the media path"?

Signaling servers have no role in the establishment of the media path,
nor does TRIP.

> 
> Also, what is the anticipated number of gateways participating in a
> GSTN-to-IP call, for the TRIP routing model? Is it always one per different
> network cloud traversed ("The selection of an egress gateway for a telephony
> call, traversing an IP network towards an ultimate destination in the PSTN"
> (and vice versa), as mentioned in Section 3 (Introduction))?

TRIP would not help in a GSTN to IP call, rather in an IP to GSTN call.
How many gateways in such a scenario? Not sure what that means. Clearly
only one gateway terminates the call. The number of gateways that can be
propagated through TRIP itself is huge, and thats the point for having
TRIP in the first place.


> 
> Finally, at the start of Section 3, the text reads "TRIP can be used to
> exchange attributes necessary to enforce policies and to select call routes
> based on path or gateway characteristics." Is it at all envisaged that the
> advertised (signalling) routes will/could also contain media-realted
> attributes (e.g., codecs) of a gateway (for example, using the
> Capability Information parameter)?

We discussed this. It would not be part of the capability information in
TRIP itself, as that refers to TRIP capabilities negotiation between
peers. The problem with such attributes is that they are not
aggregatable, and so we decided to leave them out of TRIP for now. 

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From iptel-admin@lists.bell-labs.com  Fri Jun  2 17:51:08 2000
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Jonathan,

By glancing through the document quickly, I did not find any text describing
the nature/type of the phone numbers or number ranges mentioned in the TRIP
document.  For calls terminated to those countries that support number
portability (NP), the server (in Internet) may have both the subscriber
phone number and the network routing number (or routing number) after
retrieving the NP information.  How to retrieve the NP information is
outside the scope of the TRIP discussion.   For calls terminated to
countries that do not support NP, the subscriber phone numbers are used as
the network routing number for call routing.   IMHO, the network routing
number instead of the subscriber phone number should be used for call
routing (e.g., TRIP table lookup) because the "network routing number"
applies to both NP and non-NP environments. 

In the North America, the network routing numbers are called Location
Routing Numbers (LRNs).  They "look" like subscriber phone numbers (e.g.,
the same length) and can overlap with the subscriber phone numbers.  In some
European countries, special digit (e.g., A and B) are used to differentiate
the network routing numbers from the subscriber phone numbers.  Those
network routing numbers may have shorter length than their national
subscriber phone numbers and their values do not overlap with the subscriber
phone numbers.   The TRIP document should be specific about the nature/type
of number, the network routing number, that is to be advertised.  This is
particularly important for the North American markets.  So if a GW can reach
a particular GSTN switch or network that is allocated a network routing
number (every switch or network is allocated a network routing number in the
NP environment).   It can use the TRIP to broadcast its reachability to that
switch or network.

The network routing numbers for NP are usually national numbers.  I assume
that the number ranges broadcasted using TRIP are "international" numbers
(e.g., with country code).

If TRIP broadcasts the network routing numbers or the prefixes of the
network routing numbers (e.g., the first six digits of the LRN is unique),
the information in the TRIP tables will remain "static."  There won't  be
many changes due to the changes to the network routing numbers used by the
GSTN switches or networks.  The likely changes would be from the GWs
themselves because they may add or delete the GWs they can reach.

If TRIP is used to broadcast reachability of subscriber phone numbers in the
NP environment, the TRIP broadcast will propagate through various domains
everytime there is a number porting event.  For countries that have
implmented NP and the network routing numbers overlap with the subscriber
phone numbers, it is very important for TRIP to specify the nature/type of
the numbers to be broadcasted.  Otherwise, some may broadcast the network
routing numbers and others may broadcast the subscriber phone numbers.

James



James Yu
Principal Technical Industry Liaison
NeuStar Inc.
1120 Vermont Avenue, NW,
Suite 550
Washington, D.C. 20005
USA
+1-202-533-2814 (B)
+1-202-533-2976 (Fax)
+1-202-256-1200 (Mobile)
james.yu@neustar.com
http://www.neustar.com




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From iptel-admin@lists.bell-labs.com  Fri Jun  2 18:25:40 2000
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Jonathan,

Thanks for your comments. I have a rather different model of operation in
mind, which (on a second reading), I have to admit made what I was asking
much less clear (mea culpa). So, I 'll try to give it another run...


> The purpose of the signaling messages is to exchange, between end
> systems (PCs, softphones, softswitches, or whatever), the IP addresses
> to use for media. Thus, "routing" of media itself is handled directly by
> the IP network.

Of course, but what happens _above_ the network layer? Suppose that, for
some reason, I want to have a _specific_ set of gateways in my media path
during a call. Is there any way to build such a path with TRIP information?
This rather gets closer to source routing...


> > Not by TRIP, of course, but is TRIP's position that "we just
> > assist in finding the signalling path, then let the signalling servers
> > deal with the establishment of the media path"?
>
> Signaling servers have no role in the establishment of the media path,
> nor does TRIP.

I just mean that they will distribute the IP addresses for the media, as you
point out above. As for the role of TRIP, please refer to my previous
question.


> TRIP would not help in a GSTN to IP call, rather in an IP to GSTN call.
> How many gateways in such a scenario? Not sure what that means. Clearly
> only one gateway terminates the call. The number of gateways that can be
> propagated through TRIP itself is huge, and thats the point for having
> TRIP in the first place.

Of course only one gateway terminates the call, but would you completely
rule out the possibility of needing more than one gateway on an end-to-end
call path? Or a "pure IP" call, where the gateway would be needed for, say,
media conversion only? (You can also see this as a special case of
GSTN-to-IP, or GSTN-toIP-toIP if you prefer, although I did not mean any
direction in the original message).


> > Finally, at the start of Section 3, the text reads "TRIP can be used to
> > exchange attributes necessary to enforce policies and to select call
> > routes based on path or gateway characteristics." Is it at all envisaged
> > that the advertised (signalling) routes will/could also contain
media-realted
> > attributes (e.g., codecs) of a gateway (for example, using the
> > Capability Information parameter)?
>
> We discussed this. It would not be part of the capability information in
> TRIP itself, as that refers to TRIP capabilities negotiation between
> peers. The problem with such attributes is that they are not
> aggregatable, and so we decided to leave them out of TRIP for now.

Sure, and this philosophy is clearly mentioned throughout the TRIP spec.
I was just trying to figure out whether we could find a way of deciding
to include some gateway(s) in the media path, based on advertised
capabilities (a source route maybe?) I am referring to the generalised
generalised concept of a gateway here, where you would expect even the
signalling servers to be part of a gateway (its MGC, to be exact)...

Cheers,

Dimitris

---------------------------------------------------------------
    Dimitris Terzis
    PhD Candidate - Department of Computer Science, UCL
    E-Mail: D.Terzis@cs.ucl.ac.uk
    URL : http://www.cs.ucl.ac.uk/staff/D.Terzis/
---------------------------------------------------------------




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Dimitris Terzis wrote:
> 
> Jonathan,
> 
> Thanks for your comments. I have a rather different model of operation in
> mind, which (on a second reading), I have to admit made what I was asking
> much less clear (mea culpa). So, I 'll try to give it another run...
> 
> > The purpose of the signaling messages is to exchange, between end
> > systems (PCs, softphones, softswitches, or whatever), the IP addresses
> > to use for media. Thus, "routing" of media itself is handled directly by
> > the IP network.
> 
> Of course, but what happens _above_ the network layer? Suppose that, for
> some reason, I want to have a _specific_ set of gateways in my media path
> during a call. Is there any way to build such a path with TRIP information?
> This rather gets closer to source routing...

Routing the media is outside the scope of TRIP, but the signaling
servers involved in call setup can do source routing by manipulating the
transport addresses of the media streams.

> 
> > > Not by TRIP, of course, but is TRIP's position that "we just
> > > assist in finding the signalling path, then let the signalling servers
> > > deal with the establishment of the media path"?
> >
> > Signaling servers have no role in the establishment of the media path,
> > nor does TRIP.

We've discussed this before. In some cases, it is desirable to control
the media path, and to insert media proxies along that path. For
example, QoS and Accounting at intermediate domains. However, this is
definitely outside the scope of TRIP.

> 
> I just mean that they will distribute the IP addresses for the media, as you
> point out above. As for the role of TRIP, please refer to my previous
> question.
> 
> > TRIP would not help in a GSTN to IP call, rather in an IP to GSTN call.
> > How many gateways in such a scenario? Not sure what that means. Clearly
> > only one gateway terminates the call. The number of gateways that can be
> > propagated through TRIP itself is huge, and thats the point for having
> > TRIP in the first place.
> 
> Of course only one gateway terminates the call, but would you completely
> rule out the possibility of needing more than one gateway on an end-to-end
> call path? Or a "pure IP" call, where the gateway would be needed for, say,
> media conversion only? (You can also see this as a special case of
> GSTN-to-IP, or GSTN-toIP-toIP if you prefer, although I did not mean any
> direction in the original message).

A call with multiple intermediate signaling servers (signaling proxies)
is possible, that's what TRIP is about. Having multiple media proxies
along the media path is also possible, but outside the scope of TRIP. 

A pure IP call is also possible, if the IP destination has an E.164
number assigned to it, because E.164 is the only destination address
family TRIP currently supports.

Hussein


> 
> > > Finally, at the start of Section 3, the text reads "TRIP can be used to
> > > exchange attributes necessary to enforce policies and to select call
> > > routes based on path or gateway characteristics." Is it at all envisaged
> > > that the advertised (signalling) routes will/could also contain
> media-realted
> > > attributes (e.g., codecs) of a gateway (for example, using the
> > > Capability Information parameter)?
> >
> > We discussed this. It would not be part of the capability information in
> > TRIP itself, as that refers to TRIP capabilities negotiation between
> > peers. The problem with such attributes is that they are not
> > aggregatable, and so we decided to leave them out of TRIP for now.
> 
> Sure, and this philosophy is clearly mentioned throughout the TRIP spec.
> I was just trying to figure out whether we could find a way of deciding
> to include some gateway(s) in the media path, based on advertised
> capabilities (a source route maybe?) I am referring to the generalised
> generalised concept of a gateway here, where you would expect even the
> signalling servers to be part of a gateway (its MGC, to be exact)...
> 
> Cheers,
> 
> Dimitris
> 
> ---------------------------------------------------------------
>     Dimitris Terzis
>     PhD Candidate - Department of Computer Science, UCL
>     E-Mail: D.Terzis@cs.ucl.ac.uk
>     URL : http://www.cs.ucl.ac.uk/staff/D.Terzis/
> ---------------------------------------------------------------
> 
> _______________________________________________
> IPTEL mailing list
> IPTEL@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/iptel

-- 
Hussein F. Salama
Cisco Systems
Mail Stop SJC6/3, 170 W. Tasman Drive, San Jose, CA 95134
Voice: +1 (408) 527-7147, Fax: +1 (408) 527-1714


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James Yu wrote:
> 
> Jonathan,
> 
> By glancing through the document quickly, I did not find any text describing
> the nature/type of the phone numbers or number ranges mentioned in the TRIP
> document.  For calls terminated to those countries that support number
> portability (NP), the server (in Internet) may have both the subscriber
> phone number and the network routing number (or routing number) after
> retrieving the NP information.  How to retrieve the NP information is
> outside the scope of the TRIP discussion.   For calls terminated to
> countries that do not support NP, the subscriber phone numbers are used as
> the network routing number for call routing.   IMHO, the network routing
> number instead of the subscriber phone number should be used for call
> routing (e.g., TRIP table lookup) because the "network routing number"
> applies to both NP and non-NP environments.

From TRIP's perspective all numbers/prefixes it advertises are routing
numbers, otherwise these numbers/prefixes should not have been injected
into TRIP in first place. This has been stated clearly in Section 11 of
the "Framework for Telephony Routing over IP" draft. It says: "Since
TRIP is a routing protocol, the routes it propagates should be to
routable numbers, not to names which are eventually translated to
routable numbers."

> 
> In the North America, the network routing numbers are called Location
> Routing Numbers (LRNs).  They "look" like subscriber phone numbers (e.g.,
> the same length) and can overlap with the subscriber phone numbers.  In some
> European countries, special digit (e.g., A and B) are used to differentiate
> the network routing numbers from the subscriber phone numbers.  Those

I think TRIP should be extended advertising these European formats. Is
this an extended form of E.164, or should it be a separate address
family?

> network routing numbers may have shorter length than their national
> subscriber phone numbers and their values do not overlap with the subscriber
> phone numbers.   The TRIP document should be specific about the nature/type
> of number, the network routing number, that is to be advertised.  This is
> particularly important for the North American markets.  So if a GW can reach
> a particular GSTN switch or network that is allocated a network routing
> number (every switch or network is allocated a network routing number in the
> NP environment).   It can use the TRIP to broadcast its reachability to that
> switch or network.
> 
> The network routing numbers for NP are usually national numbers.  I assume
> that the number ranges broadcasted using TRIP are "international" numbers
> (e.g., with country code).
> 

Is there a reason that prevents expanding routing numbers to full E.164
numbers before injecting them into TRIP?

Hussein


> If TRIP broadcasts the network routing numbers or the prefixes of the
> network routing numbers (e.g., the first six digits of the LRN is unique),
> the information in the TRIP tables will remain "static."  There won't  be
> many changes due to the changes to the network routing numbers used by the
> GSTN switches or networks.  The likely changes would be from the GWs
> themselves because they may add or delete the GWs they can reach.
> 
> If TRIP is used to broadcast reachability of subscriber phone numbers in the
> NP environment, the TRIP broadcast will propagate through various domains
> everytime there is a number porting event.  For countries that have
> implmented NP and the network routing numbers overlap with the subscriber
> phone numbers, it is very important for TRIP to specify the nature/type of
> the numbers to be broadcasted.  Otherwise, some may broadcast the network
> routing numbers and others may broadcast the subscriber phone numbers.
> 
> James
> 
> James Yu
> Principal Technical Industry Liaison
> NeuStar Inc.
> 1120 Vermont Avenue, NW,
> Suite 550
> Washington, D.C. 20005
> USA
> +1-202-533-2814 (B)
> +1-202-533-2976 (Fax)
> +1-202-256-1200 (Mobile)
> james.yu@neustar.com
> http://www.neustar.com
> 
> _______________________________________________
> IPTEL mailing list
> IPTEL@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/iptel

-- 
Hussein F. Salama
Cisco Systems
Mail Stop SJC6/3, 170 W. Tasman Drive, San Jose, CA 95134
Voice: +1 (408) 527-7147, Fax: +1 (408) 527-1714


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Dimitris Terzis wrote:
> 
> Jonathan,
> 
> Thanks for your comments. I have a rather different model of operation in
> mind, which (on a second reading), I have to admit made what I was asking
> much less clear (mea culpa). So, I 'll try to give it another run...
> 
> > The purpose of the signaling messages is to exchange, between end
> > systems (PCs, softphones, softswitches, or whatever), the IP addresses
> > to use for media. Thus, "routing" of media itself is handled directly by
> > the IP network.
> 
> Of course, but what happens _above_ the network layer? Suppose that, for
> some reason, I want to have a _specific_ set of gateways in my media path
> during a call. Is there any way to build such a path with TRIP information?
> This rather gets closer to source routing...

What do you mean by set of gateways? IP-PSTN gateways? The TRIP model is
a call that begins in some entity with IP access (could be a gateway
itself), terminates on one IP to PSTN gateway, and then ends within the
PSTN. There is no provision for routing calls that are
IP->PSTN->IP->PSTN->IP..... this is nearly impossible since it requires
integration with PSTN routing.

If you mean location servers - well, you could have them modify the
signaling messages to force the media through them, as Hussein suggests.
I still argue that in most cases this is a bad thing.

> > TRIP would not help in a GSTN to IP call, rather in an IP to GSTN call.
> > How many gateways in such a scenario? Not sure what that means. Clearly
> > only one gateway terminates the call. The number of gateways that can be
> > propagated through TRIP itself is huge, and thats the point for having
> > TRIP in the first place.
> 
> Of course only one gateway terminates the call, but would you completely
> rule out the possibility of needing more than one gateway on an end-to-end
> call path? Or a "pure IP" call, where the gateway would be needed for, say,
> media conversion only? (You can also see this as a special case of
> GSTN-to-IP, or GSTN-toIP-toIP if you prefer, although I did not mean any
> direction in the original message).

Then these are not gateways in the sense as defined by TRIP. Maybe you
can explain the application you have in mind?

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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From iptel-admin@lists.bell-labs.com  Sat Jun  3 06:16:41 2000
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Subject: Re: [IPTEL] TRIP questions
Date: Sat, 3 Jun 2000 11:15:14 +0100
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> > Of course, but what happens _above_ the network layer? Suppose that, for
> > some reason, I want to have a _specific_ set of gateways in my media
path
> > during a call. Is there any way to build such a path with TRIP
information?
> > This rather gets closer to source routing...
>
> What do you mean by set of gateways? IP-PSTN gateways?

More closer to MEGACO gateways (see also description at the end of this
message).


> The TRIP model is
> a call that begins in some entity with IP access (could be a gateway
> itself), terminates on one IP to PSTN gateway, and then ends within the
> PSTN.

Is therefore TRIP only "unidirectional"? We already could do the other way
using phone numbers for terminals (which doesn't sound a bad idea at all to
me), but also I don't see why an expansion of the supported advertised
address families to IP would harm the protocol...


> There is no provision for routing calls that are
> IP->PSTN->IP->PSTN->IP..... this is nearly impossible since it requires
> integration with PSTN routing.

Absolutely...


> If you mean location servers - well, you could have them modify the
> signaling messages to force the media through them, as Hussein suggests.
> I still argue that in most cases this is a bad thing.

For Location Servers, I agree, if only because they 're not supposed to do
that... I actually have in mind gateways, but more general than the ones we
currently have in mind (see text a few paragraphs below).


> > Of course only one gateway terminates the call, but would you completely
> > rule out the possibility of needing more than one gateway on an
end-to-end
> > call path? Or a "pure IP" call, where the gateway would be needed for,
say,
> > media conversion only? (You can also see this as a special case of
> > GSTN-to-IP, or GSTN-toIP-toIP if you prefer, although I did not mean any
> > direction in the original message).
>
> Then these are not gateways in the sense as defined by TRIP. Maybe you
> can explain the application you have in mind?

Yes. I am thinking of a (perhaps theoretical) global IP Telephony Network,
where terminals (either GSTN or IP) are reachable in any combination.
Addressing is dual: There is one E.164 address per IP destination, and every
GSTN number is reachable via a corresponding IP address (the one of its
access gateway).

The concept of a gateway also puzzles me. One good thing in the plain
vanilla telephony network is that all the "gateways" (i.e., the switches)
are integrated (or tightly coupled) devices, by contrast to IP, where, at
the application layer, we have Location Servers (to use TRIP terminology - I
prefer the expression "Call Routers"), Signalling Servers, Signalling
Gateways (with varying opinions as to what they do), etc., etc., etc., let
alone the multiplicity of devices at the network layer and below (which we
can ignore, though).

My opinion is that, if we are ever to see global IP Telephony (which I
personally doubt), we need a more clear framework, at least for the core
components of the network, which means, among other things, unifying all
these devices according to their role. Now, it might not be conceivable
making a combined LS + gateway (i.e, an IP Telephony corresponding of the
GSTN switch), but at least we could do that about the gateways.

Given the different types of signalling and media entities in an IP
Telephony call, a unified gateway (a gateway framework , if you prefer) can
only be perceived as an entity general enough to enclose all these entities
in one "package". This should be generic enough as to allow us, with very
little additional logic, to "mutate" it to anything from a single SIP Proxy
to a full-blown network of multiple signalling and media components,
advertised to the outside world as a unique entity. There is, of course, the
concept of the MEGACO gateway, but this is so closely modelled after the
H.323 one, that I am not absolutely convinced it is the answer (possibly
with a bit more work on it).

In the above (perhaps ideal) model, the type of terminals on either end of a
connection is irrelevant and you might expect that, for a "long-distance"
call, you will have _several_ gateways on the path, the same way it
currently happens in GSTN.

Of course, I realise that TRIP's model is much more specific (down to earth,
if you like) than the one I describe (hence the double figure of "this is
outside the scope of TRIP" comments I 've already got in this thread! :-),
but the original intention of the message was to verify how much of this we
could do with TRIP. I 've been watching this whole effort since TBGP and GLP
and went through all TRIP's drafts, but (to the expense of being considered
an... alien) I thought I could resort to the source for best information!
:-)

Thanks for the time and the very useful comments.

Regards,

Dimitris

---------------------------------------------------------------
    Dimitris Terzis
    PhD Candidate - Department of Computer Science, UCL
    E-Mail: D.Terzis@cs.ucl.ac.uk
    URL : http://www.cs.ucl.ac.uk/staff/D.Terzis/
---------------------------------------------------------------





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From iptel-admin@lists.bell-labs.com  Sun Jun  4 23:12:19 2000
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Subject: Re: [IPTEL] TRIP questions
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Dimitris Terzis wrote:
> 
> > The TRIP model is
> > a call that begins in some entity with IP access (could be a gateway
> > itself), terminates on one IP to PSTN gateway, and then ends within the
> > PSTN.
> 
> Is therefore TRIP only "unidirectional"? We already could do the other way
> using phone numbers for terminals (which doesn't sound a bad idea at all to
> me),

In this case it is not a routing problem, but a naming problem. See the
ENUM working group for a solution to that.

> but also I don't see why an expansion of the supported advertised
> address families to IP would harm the protocol...

It would not; and the set of families is indeed extensible.

> > > Of course only one gateway terminates the call, but would you completely
> > > rule out the possibility of needing more than one gateway on an
> end-to-end
> > > call path? Or a "pure IP" call, where the gateway would be needed for,
> say,
> > > media conversion only? (You can also see this as a special case of
> > > GSTN-to-IP, or GSTN-toIP-toIP if you prefer, although I did not mean any
> > > direction in the original message).
> >
> > Then these are not gateways in the sense as defined by TRIP. Maybe you
> > can explain the application you have in mind?
> 
> Yes. I am thinking of a (perhaps theoretical) global IP Telephony Network,
> where terminals (either GSTN or IP) are reachable in any combination.
> Addressing is dual: There is one E.164 address per IP destination, and every
> GSTN number is reachable via a corresponding IP address (the one of its
> access gateway).
> 
> The concept of a gateway also puzzles me. One good thing in the plain
> vanilla telephony network is that all the "gateways" (i.e., the switches)
> are integrated (or tightly coupled) devices, by contrast to IP, where, at
> the application layer, we have Location Servers (to use TRIP terminology - I
> prefer the expression "Call Routers"), Signalling Servers, Signalling
> Gateways (with varying opinions as to what they do), etc., etc., etc., let
> alone the multiplicity of devices at the network layer and below (which we
> can ignore, though).
> 
> My opinion is that, if we are ever to see global IP Telephony (which I
> personally doubt), we need a more clear framework, at least for the core
> components of the network, which means, among other things, unifying all
> these devices according to their role. Now, it might not be conceivable
> making a combined LS + gateway (i.e, an IP Telephony corresponding of the
> GSTN switch), but at least we could do that about the gateways.

?? THis is exactly the opposite of the trend, which is to decompose
these monster gateways into now FOUR components - signaling gateway,
media gateway, gateway controller, and applications server.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From iptel-admin@lists.bell-labs.com  Mon Jun  5 07:06:56 2000
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From: Dimitris Terzis <D.Terzis@cs.ucl.ac.uk>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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References: <002e01bfcc8c$ff195040$9f036cd5@kallisto> <3937DC81.1CB9748A@dynamicsoft.com> <002b01bfcce1$7cb35a60$b7036cd5@kallisto> <393883AC.3C7AEA6@dynamicsoft.com> <001601bfcd44$ca7fb320$a2096cd5@kallisto> <393B1A71.58739DF2@dynamicsoft.com>
Subject: Re: [IPTEL] TRIP questions
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Hi...

> > My opinion is that, if we are ever to see global IP Telephony (which I
> > personally doubt), we need a more clear framework, at least for the core
> > components of the network, which means, among other things, unifying all
> > these devices according to their role. Now, it might not be conceivable
> > making a combined LS + gateway (i.e, an IP Telephony corresponding of
the
> > GSTN switch), but at least we could do that about the gateways.
>
> ?? THis is exactly the opposite of the trend, which is to decompose
> these monster gateways into now FOUR components - signaling gateway,
> media gateway, gateway controller, and applications server

I know... Well, it's another point of view. I think the important part is
the larger picture, i.e., how to provide for a trully large-scale ISTN (IP
Switched Telephone Network), and this rather depends more on routing, than
on how you define the gateway _internally_...

Cheers,

Dimitris

---------------------------------------------------------------
    Dimitris Terzis
    PhD Candidate - Department of Computer Science, UCL
    E-Mail: D.Terzis@cs.ucl.ac.uk
    URL : http://www.cs.ucl.ac.uk/staff/D.Terzis/
---------------------------------------------------------------




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Subject: [IPTEL] Regarding Interoperability between VoIP Protocol stacks
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I would need your help to understand the interoperability issue between
different VoIP stacks.

Are there any design issues to be considered , while developing an
independent stack , so that we do not have problems later when you try to
introduc einteroperability of this stack with any other stack. Moreover,
does Interoperability functionality sits on top of the stacks. What I mean
is, can we develop two stacks independently and work on interoperability
between the two at  a later period of time.

Are there any guidelines on that ???


Regards
Rishi Choudhary

Wipro Technologies
Bangalore



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Subject: [IPTEL] Telephone newtork statistics
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Hi...

I am looking for good referrence(s) with very recent statistics on the
international GSTN, and particularly for things like number of switches per
hierarchical level, proportion of switches
per level (e.g., SSP/STP/SCP for SS7), and similar information. Generic web
searches I 've tried proved of not much help, and the few telecoms
newsgroups in Usent are so busy moderating themselves that no similar
question can get through.

Does anyone know of pointers to information like this?

Thanks,

Dimitris

---------------------------------------------------------------
    Dimitris Terzis
    PhD Candidate - Department of Computer Science, UCL
    E-Mail: D.Terzis@cs.ucl.ac.uk
    URL : http://www.cs.ucl.ac.uk/staff/D.Terzis/
---------------------------------------------------------------




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> > Of course, but what happens _above_ the network layer? Suppose that, for
> > some reason, I want to have a _specific_ set of gateways in my
> media path
> > during a call. Is there any way to build such a path with TRIP
> information?
> > This rather gets closer to source routing...
>
> What do you mean by set of gateways? IP-PSTN gateways? The TRIP model is
> a call that begins in some entity with IP access (could be a gateway
> itself), terminates on one IP to PSTN gateway, and then ends within the
> PSTN. There is no provision for routing calls that are
> IP->PSTN->IP->PSTN->IP..... this is nearly impossible since it requires
> integration with PSTN routing.
>
Well, if you think of the PSTN as either a single hop for which you inject a
static route into TRIP, or an IGP-based thing where you get the information
separately (e.g. from an SCP) and inject it, then actually this isn't so
hard. In fact, I think it's kind of neat that TRIP could do this fairly
easily as long as you make sure that each signaling hop is in the media path
and capable of packet/circuit conversion.

Of course the practical value of this is highly questionable, because the
intermediate transcoding and de-jittering at each PSTN interconnect point
will quickly add up to pretty poor voice quality and rather long delays.

Dave.



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To: "Choudhary, Rishi [Partner:WBNG:7P81]" <rishi.choudhary@wipro.com>,
        iptel <iptel@lists.bell-labs.com>
Subject: RE: [IPTEL] Regarding Interoperability between VoIP Protocol stac ks
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A stack by definition is a layered collection of protocol entities.  Each
layer has its role in terms of signalling performance, reliability, and
security.  I assume from your question that you are talking about
application-layer interoperability, with an application-layer interworking
function to provide it.  From a semantic point of view this can be developed
without worrying about what is down below, but you have to make sure your
new stack as a whole meets the engineering assumptions which the other one
has built in.

> -----Original Message-----
> From:	Choudhary, Rishi [Partner:WBNG:7P81] 
> Sent:	Monday, June 05, 2000 7:12 AM
> To:	iptel
> Subject:	[IPTEL] Regarding Interoperability between VoIP Protocol
> stacks
> 
> I would need your help to understand the interoperability issue between
> different VoIP stacks.
> 
> Are there any design issues to be considered , while developing an
> independent stack , so that we do not have problems later when you try to
> introduc einteroperability of this stack with any other stack. Moreover,
> does Interoperability functionality sits on top of the stacks. What I mean
> is, can we develop two stacks independently and work on interoperability
> between the two at  a later period of time.
> 
> Are there any guidelines on that ???
> 
> 
> Regards
> Rishi Choudhary
> 
> Wipro Technologies
> Bangalore
> 
> 
> 
> _______________________________________________
> IPTEL mailing list
> IPTEL@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/iptel

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
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<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2651.65">
<TITLE>RE: [IPTEL] Regarding Interoperability between VoIP Protocol =
stacks</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">A stack by =
definition is a layered collection of protocol entities.&nbsp; Each =
layer has its role in terms of signalling performance, reliability, and =
security.&nbsp; I assume from your question that you are talking about =
application-layer interoperability, with an application-layer =
interworking function to provide it.&nbsp; From a semantic point of =
view this can be developed without worrying about what is down below, =
but you have to make sure your new stack as a whole meets the =
engineering assumptions which the other one has built in.</FONT></P>
<UL>
<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Choudhary, Rishi [Partner:WBNG:7P81] </FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, June 05, 2000 7:12 AM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">iptel</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">[IPTEL] Regarding Interoperability =
between VoIP Protocol stacks</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">I would need your help to understand =
the interoperability issue between</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">different VoIP stacks.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Are there any design issues to be =
considered , while developing an</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">independent stack , so that we do not =
have problems later when you try to</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">introduc einteroperability of this =
stack with any other stack. Moreover,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">does Interoperability functionality =
sits on top of the stacks. What I mean</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">is, can we develop two stacks =
independently and work on interoperability</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">between the two at&nbsp; a later =
period of time.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Are there any guidelines on that =
???</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Arial">Regards</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Rishi Choudhary</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Arial">Wipro Technologies</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">Bangalore</FONT>
</P>
<BR>
<BR>

<P><FONT SIZE=3D2 =
FACE=3D"Arial">_______________________________________________</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">IPTEL mailing list</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Arial">IPTEL@lists.bell-labs.com</FONT>
<BR><U><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial"><A =
HREF=3D"http://lists.bell-labs.com/mailman/listinfo/iptel" =
TARGET=3D"_blank">http://lists.bell-labs.com/mailman/listinfo/iptel</A><=
/FONT></U>
</P>
</UL>
</BODY>
</HTML>
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From iptel-admin@lists.bell-labs.com  Mon Jun  5 22:55:06 2000
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From: James Yu <james.yu@neustar.com>
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Cc: "'jdrosen@dynamicsoft.com'" <jdrosen@dynamicsoft.com>,
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> 
> From TRIP's perspective all numbers/prefixes it advertises are routing
> numbers, otherwise these numbers/prefixes should not have 
> been injected
> into TRIP in first place. This has been stated clearly in 
> Section 11 of
> the "Framework for Telephony Routing over IP" draft. It says: "Since
> TRIP is a routing protocol, the routes it propagates should be to
> routable numbers, not to names which are eventually translated to
> routable numbers."
> 

I'm glad to see that this particular point (roting number vs. subscriber
number) has been clarified in the TRIP document.   Thanks for pointing that
out.  May be that statement should be put in the abstract to make that point
clear.

> > 
> > In the North America, the network routing numbers are 
> called Location
> > Routing Numbers (LRNs).  They "look" like subscriber phone 
> numbers (e.g.,
> > the same length) and can overlap with the subscriber phone 
> numbers.  In some
> > European countries, special digit (e.g., A and B) are used 
> to differentiate
> > the network routing numbers from the subscriber phone 
> numbers.  Those
> 
> I think TRIP should be extended advertising these European formats. Is
> this an extended form of E.164, or should it be a separate address
> family?

Some of the routing numbers (or routig prefixes, as they are usually
prefixes to the dialed called party number) contain the digits outside 0-9
(e.g., A, B, C, D, E, and F).  They are valid routing numbers (national
numbers) either within one network (used by a particular network) or between
networks.  Since they don't just contain digits 0-9, we may not say that
they follow E.164 format.  But they should be viewed as valid routing
numbers.  In this case, a valid digit in a routng number can be from 0
(0000) to F (1111).  After adding the country code to the routing numbers in
Europe, those "international" routing numbers probably can be used for
routing.  The country code will allow the call to be routed to that country.
After removing the country code by a gateway at that country, the "national"
routing number can then be used for routing.  

Since the routing number is cancatenated with the dialed called party number
(they are carried in the Called Party Number paramter)in some NP
implementations in Europe, the "routing number" we are discussing should be
limited to the "routing number" or "routing prefix" instead of the "routing
number + dialed called party number."   For example, UK uses "5xxxxx" as a
routing number/prefix where "xxxxx" identifies the carrier.  A carrier can
assign additional numbers after "5xxxxx" which can point to a network, a
switch/exchange, or even a line card at a switch/exchange.  In that case,
"5xxxxx" plus the "addtional digits" would be treated us the routing number.
I believe that this routing number is not longer than 15 digits. 

Any comment from people in Europe?

> 
> Is there a reason that prevents expanding routing numbers to 
> full E.164
> numbers before injecting them into TRIP?
> 

If "5xxxxx+x....x" is treated as a routing prefix (no exception within the
range), then there is no need to expand a routing number to full E.164
numbers.  In some countries, the E.164 numbers have variable length of
digits.  They don't have a fixed length of digits (in North America, we have
11 digits for all "international" NANP numbers).

James


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Subject: RE: [IPTEL] Regarding Interoperability between VoIP Protocol stacks
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The APIs and message sets on top of the stacks are not
standarized, but I do suggest you look at the JAIN work
which is attempting to do that for Java applications.

<troll> Of course, for really high-performance applications, Java
is not appropriate. </troll>

We have been able in our softswitch system
to not only abstract between different H.323 implementations,
but also between H.323 and MGCP stack implementations,
so it can be done.

-Bob


-----------
Bob Wise
Chief Architect
Vsys, Inc. - "The Softswitch Company"
www.vsys.com


-----Original Message-----
From: iptel-admin@lists.bell-labs.com
[mailto:iptel-admin@lists.bell-labs.com]On Behalf Of Rishi Choudhary
Sent: Monday, June 05, 2000 5:12 AM
To: iptel
Subject: [IPTEL] Regarding Interoperability between VoIP Protocol stacks


I would need your help to understand the interoperability issue between
different VoIP stacks.

Are there any design issues to be considered , while developing an
independent stack , so that we do not have problems later when you try to
introduc einteroperability of this stack with any other stack. Moreover,
does Interoperability functionality sits on top of the stacks. What I mean
is, can we develop two stacks independently and work on interoperability
between the two at  a later period of time.

Are there any guidelines on that ???


Regards
Rishi Choudhary

Wipro Technologies
Bangalore



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Date: Tue, 06 Jun 2000 19:40:08 -0400
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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Subject: [IPTEL] Availability of RTP library source code in C
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Folks,

I am pleased to finally announce the availability of source code for
RTPlib, and RTP library in C developed jointly by myself, Daniel
Rubenstein from U.Mass Amherst, and Henning Schulzrinne and Jonathan
Lennox from Columbia University. The library is fairly complete,
implementing some of the harder things, such as SSRC collision, mixers,
and RTCP timer reconsideration, in a transparent manner. It comes with
HTML documentation and example applications. 

The library is free for non-commercial and research usage. It has been
run and tested primarily on Solaris, but has also been used on NT.

You can obtain the software at:
http://www.bell-labs.com/topic/swdist/

Thanks!

-Jonathan R.
-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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    When do we use ASN Compiler in H.323 Application and how we use it ? =

If you have an ASN Compiler, please send it to me and example of using =
it.

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From iptel-admin@lists.bell-labs.com  Thu Jun  8 17:56:55 2000
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To: James Yu <james.yu@neustar.com>
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        "'iptel@lists.bell-labs.com'" <iptel@lists.bell-labs.com>
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James Yu wrote:
> 
> > >
> > > In the North America, the network routing numbers are
> > called Location
> > > Routing Numbers (LRNs).  They "look" like subscriber phone
> > numbers (e.g.,
> > > the same length) and can overlap with the subscriber phone
> > numbers.  In some
> > > European countries, special digit (e.g., A and B) are used
> > to differentiate
> > > the network routing numbers from the subscriber phone
> > numbers.  Those
> >
> > I think TRIP should be extended advertising these European formats. Is
> > this an extended form of E.164, or should it be a separate address
> > family?
> 
> Some of the routing numbers (or routig prefixes, as they are usually
> prefixes to the dialed called party number) contain the digits outside 0-9
> (e.g., A, B, C, D, E, and F).  They are valid routing numbers (national
> numbers) either within one network (used by a particular network) or between
> networks.  Since they don't just contain digits 0-9, we may not say that
> they follow E.164 format. 

Oh yuck.

I'm curious how this impacts the aggregatability of such numbers? Are
they assigned in such a way that they can still be aggregated even
though there are these random letters in the phone number? Is there any
kind of hierarchy in these numbers?

-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com


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From iptel-admin@lists.bell-labs.com  Fri Jun  9 00:40:39 2000
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From: James Yu <james.yu@neustar.com>
To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>
Cc: "'hsalama@cisco.com'" <hsalama@cisco.com>,
        "'iptel@lists.bell-labs.com'" <iptel@lists.bell-labs.com>
Subject: RE: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
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My understanding about the reason of using the digit A, B, ... or F is to
avoid taking away the valid E.164 numbers from being assigned as subscriber
numbers.  In ISUP Called Party Number parameter, there is a Nature of
Address (NOA) field.  Some countries assigned a new NOA value
(country-specific) to indicate that routing number is concatenated with the
dialed called party number and may implicitly indicate that NP database dip
has been performed.  If we use "rn" parameter in tel URL to carry the
routing number, we may need to define the rn in a way to allow 0-9 and A-F.

The routing number, for example, "5xxxxx+ additional digits defined by the
assigned carrier" in UK, may have hierarchy where "5" indicates a routing
number for NP and "xxxxx" identifies the carrier.  The "additional digits"
may have hierarchy.  For example, BT may define the routing number to
identify the line card (concentrator) at the switch.  This implies that
those "additional digits" can be divided at least into two levels: one
identifies the switch and the other the line card.

James

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, June 08, 2000 5:38 PM
> To: James Yu
> Cc: 'hsalama@cisco.com'; 'iptel@lists.bell-labs.com'
> Subject: Re: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
> 
> 
> 
> 
> James Yu wrote:
> > 
> > > >
> > > > In the North America, the network routing numbers are
> > > called Location
> > > > Routing Numbers (LRNs).  They "look" like subscriber phone
> > > numbers (e.g.,
> > > > the same length) and can overlap with the subscriber phone
> > > numbers.  In some
> > > > European countries, special digit (e.g., A and B) are used
> > > to differentiate
> > > > the network routing numbers from the subscriber phone
> > > numbers.  Those
> > >
> > > I think TRIP should be extended advertising these 
> European formats. Is
> > > this an extended form of E.164, or should it be a separate address
> > > family?
> > 
> > Some of the routing numbers (or routig prefixes, as they are usually
> > prefixes to the dialed called party number) contain the 
> digits outside 0-9
> > (e.g., A, B, C, D, E, and F).  They are valid routing 
> numbers (national
> > numbers) either within one network (used by a particular 
> network) or between
> > networks.  Since they don't just contain digits 0-9, we may 
> not say that
> > they follow E.164 format. 
> 
> Oh yuck.
> 
> I'm curious how this impacts the aggregatability of such numbers? Are
> they assigned in such a way that they can still be aggregated even
> though there are these random letters in the phone number? Is 
> there any
> kind of hierarchy in these numbers?
> 
> -Jonathan R.
> 
> -- 
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
> 


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From iptel-admin@lists.bell-labs.com  Fri Jun  9 07:46:25 2000
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From: M.Muench@alcatel.de
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Subject: RE: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
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I can agree with the explanation of the use of hexadecimal digits in the
Routing Number.
But not all values "A-F" can be used. Value "B and C" are standardised as
digits for special operator functions.
Value "F" has never been used for the Routing Number or other applications,
because this value is standardised and used as the "End of Signal"
indication in the routing process. That means that subsequent digits are
not accepted and therefore discarded.

So for the Routing Number the following digits can be used:

1) hexadecimal digits A, D, E;
   the use of B and C is country specific

or

2) free number blocks of the numbering range

Please take this correction into account if the relevant text will be
included in the next draft of TRIP.

Kind regards,

Monika Muench
Alcatel SEL AG, Germany


----------
From: james.yu@neustar.com
To: jdrosen@dynamicsoft.com
Cc: hsalama@cisco.com; iptel@lists.bell-labs.com
Subject: RE: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
Date: Friday, June 09, 2000 6:38AM

My understanding about the reason of using the digit A, B, ... or F is to
avoid taking away the valid E.164 numbers from being assigned as subscriber
numbers.  In ISUP Called Party Number parameter, there is a Nature of
Address (NOA) field.  Some countries assigned a new NOA value
(country-specific) to indicate that routing number is concatenated with the
dialed called party number and may implicitly indicate that NP database dip
has been performed.  If we use "rn" parameter in tel URL to carry the
routing number, we may need to define the rn in a way to allow 0-9 and A-F.

The routing number, for example, "5xxxxx+ additional digits defined by the
assigned carrier" in UK, may have hierarchy where "5" indicates a routing
number for NP and "xxxxx" identifies the carrier.  The "additional digits"
may have hierarchy.  For example, BT may define the routing number to
identify the line card (concentrator) at the switch.  This implies that
those "additional digits" can be divided at least into two levels: one
identifies the switch and the other the line card.

James

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> Sent: Thursday, June 08, 2000 5:38 PM
> To: James Yu
> Cc: 'hsalama@cisco.com'; 'iptel@lists.bell-labs.com'
> Subject: Re: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
>
>
>
>
> James Yu wrote:
> >
> > > >
> > > > In the North America, the network routing numbers are
> > > called Location
> > > > Routing Numbers (LRNs).  They "look" like subscriber phone
> > > numbers (e.g.,
> > > > the same length) and can overlap with the subscriber phone
> > > numbers.  In some
> > > > European countries, special digit (e.g., A and B) are used
> > > to differentiate
> > > > the network routing numbers from the subscriber phone
> > > numbers.  Those
> > >
> > > I think TRIP should be extended advertising these
> European formats. Is
> > > this an extended form of E.164, or should it be a separate address
> > > family?
> >
> > Some of the routing numbers (or routig prefixes, as they are usually
> > prefixes to the dialed called party number) contain the
> digits outside 0-9
> > (e.g., A, B, C, D, E, and F).  They are valid routing
> numbers (national
> > numbers) either within one network (used by a particular
> network) or between
> > networks.  Since they don't just contain digits 0-9, we may
> not say that
> > they follow E.164 format.
>
> Oh yuck.
>
> I'm curious how this impacts the aggregatability of such numbers? Are
> they assigned in such a way that they can still be aggregated even
> though there are these random letters in the phone number? Is
> there any
> kind of hierarchy in these numbers?
>
> -Jonathan R.
>
> --
> Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> Chief Scientist                             First Floor
> dynamicsoft                                 East Hanover, NJ 07936
> jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> http://www.dynamicsoft.com
>


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Subject: [IPTEL] How to use ASN.1 Compiler
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What is ASN.1 Compiler and when do we use ASN.1 Compiler in H.323 =
application ? If you have ASN.1 Compiler,
please send it to me with some example of using it

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From: "Suntichai Chuaywong" <s4110400@maliwan.psu.ac.th>
To: <iptel@lists.research.bell-labs.com>
Date: Wed, 7 Jun 2000 23:00:19 +0700
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Subject: [IPTEL] How to use ASN.1 Compiler
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I wonder what ASN.1 Compiler use for and when to use it in H.323 =
application ? If you have ASN.1 compiler, please
send it to me with some example of using it
                                                                    =
Thank you very much
                                                                        =
Suntichai Chuaywong

------=_NextPart_000_001F_01BFD0D4.27321120
Content-Type: text/html;
	charset="windows-874"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dwindows-874" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I wonder what ASN.1 Compiler use for =
and=20
when&nbsp;to use it in H.323 application ? If you have ASN.1 compiler,=20
please</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>send it to me with some example of =
using=20
it</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; Thank you very=20
much</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
Suntichai Chuaywong</FONT></DIV></BODY></HTML>

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From iptel-admin@lists.bell-labs.com  Fri Jun  9 12:56:17 2000
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Date: Thu, 8 Jun 2000 10:56:45 -0400 (EDT)
From: Bancroft Scott <baos@oss.com>
To: "Nadathur, Srivathsan" <snadathur@bell-labs.com>,
        iptel@lists.research.bell-labs.com
Cc: "'support@oss.com'" <support@oss.com>
Subject: Re: FW: [IPTEL] How to use ASN Compiler
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On Thu, 8 Jun 2000, Nadathur, Srivathsan wrote:

>     When do we use ASN Compiler in H.323 Application and how we use it ? 
> If you have an ASN Compiler, please send it to me and example of using it.

You use the ASN.1 compiler to compile the H.323 ASN.1-defined messages.
The output of the ASN.1 compiler consist of files that contain C structs,
C++ classes or Java classes, depending on whether you are using the OSS
ASN.1/C, ASN.1/C++ or ASN.1/Java product.  You then use the
classes/structs defined in these files in writing your application.

Let me know if the above is not sufficiently clear to you.  You can also
bet a better idea as to the possibilities by looking at:

	http://www.oss.com/products/tools.html
	http://www.oss.com/products/application.html
	http://www.oss.com/products/asn1cpp/tools_cpp.html
	http://www.oss.com/products/asn1java/javatools.html
	http://www.oss.com/products/asn1java/javaapp.html

If you would like a trial license of the OSS ASN.1 Tools please send email
to info@oss.com.

If you have any technical questions, please send email to support@oss.com.

-------------------------------------------------------------------------
Bancroft Scott                               Toll Free    :1-888-OSS-ASN1
OSS Nokalva                                  International:1-732-302-0750
baos@oss.com                                 Tech Support :1-732-302-9669 x-200
http://www.oss.com                           Fax          :1-732-302-0023




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From: Jonathan Lennox <lennox@cs.columbia.edu>
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Subject: [IPTEL] Re: CPL Timezone offset
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On Thu, June 8 2000, "Amit Bhadoria" wrote to "iptel@lists.research.bell-labs.com, lennox@cs.columbia.edu, schulzrinne@opus.cs.columbia.edu" saying:

> hi,
> 
> i'm working on an implementation of CPL (draft-ietf-iptel-cpl-01). The
> reffered document specifies a syntax for the "standard" and "daylight"
> offset tags of "timezone" (section 9 and fig 17).
> 
> presently, the document specifies that the parameters of theses offset
> tags specify "... a set of instants when the time zone rule takes effect,
> I N    T H E    L O C A L   T I M E   O F   T H E  ``O T H E R''  
> O F F S E T ..."
> 
> if there are more than two offsets defined for a particular timezone, 
> keeping a reference to this ``OTHER'' is difficult (if not ambiguous!),
> making the implementation very ugly (due to brute force approach).
> 
> in my opinion, there needs to be one more parameter in the offset tags,
> that referes to the "abbr" field of the ``OTHER'' offset (after which it
> becomes effective).
> 
> please suggest if there is some other elegant implementation poosible.

At the IPTel meeting at the last IETF, it seemed that rough consensus was
that no one liked the way the most recent CPL draft defined timezones.
After investigating the issue a fair amount, I've determined that a) it's
much harder than I thought, and b) other groups have faced the same problem.
My current thought is to simply reference the timezones defined by iCal;
this is the best way of leveraging others' work, though it does
unfortunately impose a requirement that CPL implementors understand a rather
complex separate protocol language.

The other possibility would be to reference the Olson TZ database
<http://www.twinsun.com/tz/tz-link.htm>, but unfortunately that isn't a
normative reference.  There've been some efforts to generate a normative set
of time zone names from that database for use in iCal, but I don't believe
very much has come of this so far.

The suggestion you gave above -- of referencing the abbreviation of the
other zone -- doesn't unfortunately work, due to things like the Australian
timezones "Eastern Standard Time" and "Eastern Summer Time" which have the
same abbreviation.

-- 
Jonathan Lennox
lennox@cs.columbia.edu



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Date: Fri, 09 Jun 2000 10:52:33 -0400
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Subject: [IPTEL] 3rd CFP (EXTENDED DEADLINE): IP Telecom Services Workshop 2000
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[Please accept our apologies for multiple copies of this call.]

-------------   THIRD CALL FOR PAPERS  ---------------
-------------     EXTENDED DEADLINE    ---------------

       IP TELECOM SERVICES WORKSHOP 2000 (IPTS 2000)

                    co-located with
             Fall 2000 Voice on the Net (VON)

                     organized by
                  AT&T Labs Research
                      pulver.com

         Cobb Galleria, Atlanta, Georgia, U.S.A
                  September 11, 2000

            submission deadline: June 23, 2000
         http://www.research.att.com/conf/ipts2000/

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

IPTS 2000 is a new workshop dedicated to the important, emerging field of IP
Telecom Services research.

In contrast to the services available on the PSTN, a new world of telecom
services is possible on an IP-based infrastructure. This world presents new
challenges for service development, deployment and management. This one-day
workshop will serve as a forum for the dissemination of research relating to
these challenges.

The IPTS 2000 Program Committee invites submission of papers on substantial,
original and previously unpublished research in IP telecom services, including,
but not limited to:

- service architectures
  (e.g. AIN, SoftSwitch, JAIN, Parlay, DFC)
- service creation environments
- protocols, standards and interoperability
  (e.g. H.323, SIP, MGCP, Megaco/H.248, etc.)
- service programming languages
- service management - billing, fault tolerance, monitoring
- novel services

We welcome papers describing original research; position papers describing
research interests in the field, work in progress, or future directions of
research; project or system descriptions.


IMPORTANT DATES

June 23, 2000        Submission deadline
July 21, 2000        Notification of acceptance
August 25, 2000      Final version of papers
September 11, 2000   IPTS 2000


PAPER SUBMISSIONS

Papers must describe original, previously unpublished work that has not been
simultaneously submitted for publication elsewhere. They must be written in
English, not exceeding 6 pages including figures, tables and references in 10-12
point font on 8.5x11-inch (US letter) paper. The first page must include an
abstract of up to 200 words, keywords, and email address of the corresponding
author.

All papers must be submitted electronically unless specifically approved by the
Workshop Co-Chairs. Submissions in PostScript or PDF format should be sent to
mailto:ipts2000@research.att.com.

Authors of accepted papers are expected to present their contribution at the
workshop.  The proceedings will be published in hardcopy and on the web.


REVIEW PROCEDURE

All submissions will be subject to academic peer review by the IPTS 2000 Program
Committee under the chairmanship of the Workshop Co-Chairs. Each paper will
receive three written reviews, and reviews will be returned to the authors.  The
Workshop Co-Chairs have final authority over the review process and all
decisions relating to acceptance of papers. Review criteria include originality
of ideas, technical soundness, significance of results, and quality of
presentation.


CONFERENCE VENUE AND RELATED EVENTS

IPTS 2000 will be held on September 11, 2000 in Atlanta, Georgia. The workshop
will be co-located with Fall 2000 Voice on the Net (VON).  Details about Fall
2000 VON can be found at http://pulver.com.

IPTS 2000 is organized by AT&T Labs Research (http://www.research.att.com) and
pulver.com (http://pulver.com).


WORKSHOP CO-CHAIRS

Greg Bond, AT&T Labs Research
Eric Cheung, AT&T Labs Research

PROGRAM COMMITTEE

Mauricio Arango, Sun Microsystems Labs
Joanne Atlee, University of Waterloo
Ralph Blumenthal, Daewoo Telecom
Scott Hoffpauir, BroadSoft
Evan Magill, University of Strathclyde
Peter Mataga, dynamicsoft
Bernie Pagurek, Carleton University
Jonathan Rosenberg, dynamicsoft
Henning Schulzrinne, Columbia University
Greg Utas, Nortel Networks
Pamela Zave, AT&T Labs Research


FURTHER GENERAL INFORMATION

http://www.research.att.com/conf/ipts2000



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Hi,
MGCP/Megaco Conference 14 to 17 November 2000. Second edition: =
deployment experiences, operational issues, interoperability status.
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hi,

i'm working on an implementation of CPL (draft-ietf-iptel-cpl-01). The
reffered document specifies a syntax for the "standard" and "daylight"
offset tags of "timezone" (section 9 and fig 17).

presently, the document specifies that the parameters of theses offset
tags specify "... a set of instants when the time zone rule takes effect,
I N    T H E    L O C A L   T I M E   O F   T H E  ``O T H E R''  
O F F S E T ..."

if there are more than two offsets defined for a particular timezone, 
keeping a reference to this ``OTHER'' is difficult (if not ambiguous!),
making the implementation very ugly (due to brute force approach).

in my opinion, there needs to be one more parameter in the offset tags,
that referes to the "abbr" field of the ``OTHER'' offset (after which it
becomes effective).

please suggest if there is some other elegant implementation poosible.

thanks and regards,
-amit.
<amitb@vovida.com>




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From iptel-admin@lists.bell-labs.com  Tue Jun 13 07:41:36 2000
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Hi,

Why is it needed the "optional parameter length" in the OPEN message if
the length of the whole message is already present in the fixed-sized
header?

Thanks,

Gonzalo
-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland



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From iptel-admin@lists.bell-labs.com  Tue Jun 13 10:59:46 2000
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Some important information with regard to the Routeing Number used in
telecommunication networks, if Number Portability has to be supported:
======================================================================

In the existing networks, routeing is a quite complex procedure. Several
input data are used (e.g. information from and on the origin, information
on the destination, signalling capabilities, bearer requirements). Before
Number Portability, there was no real distinction between the Directory
Number and the Routeing Number; now, for a call without Number Portability,
the same still applies - there is no real distinction between the Directory
Number and the Routeing Number. Both are represented by one and the same
E.164-number which is transferred in one parameter (Called Party Number
parameter).

By the introduction of Number Portability, this has changed. There is a need
for a split into a Directory Number (DN) and a Routeing Number (RN), also
from a protocol point of view. The DN does not change, while the RN is used
for routeing purposes.

From an ISUP perspective, three addressing mechanisms for the RN have
been specified in Q.769.1. ETSI has endorsed Q.769.1 with one
exception: The NP-method "Dropback" is not supported. Again - as
far as the addressing mechanisms are concerned, there is no
difference between ITU-T and ETSI.

Number Portability is used only in a national environment and has never been
planned for international use, i.e. in the case of international calls,
it is always seen as the responsibility of the destination country to
perform Number Portability (especially the data base query).
Example: An "All Call Query" in the origination country for an international
call is not planned to be used for Number Portability.
When the call enters the network of the destination network then e.g.
the "All Call Query" solution can be used to retrieve the Routeing Number.

If the data base query for international calls should be performed in
the origination country, several problems have to be solved, e.g.
administrating ALL routing numbers (worldwide!) and keeping them up to date.

In general for the coding of the Routeing Number in ISUP the same principles
apply as for the called party number otherwise a worldwide function of
telecommunication would not work if basic routeing procedures would be
changed. However, the Number Portability relevant parts of the ISUP
protocol are for "national use". Therefore, (bilateral?) agreements
on the ISUP protocol issues are needed as well.

NOTE: The coding and the rules can be found in ITU-T ISUP
Recommendation Q.763, Q.764 (12/99), and Q.769.1. As far as ETSI is
concerned: see above.


Kind regards,

Monika Muench
Alcatel SEL AG, Germany



----------
From: hsalama@cisco.com
To: Muench, Monika / selh2
Subject: Re: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
Date: Monday, June 12, 2000 9:05AM

Monika,

Are the formats of European routning numbers standardized somewhere?
Could you please point me to the relevant standards?

Thank you.

Hussein


M.Muench@alcatel.de wrote:
>
> I can agree with the explanation of the use of hexadecimal digits in the
> Routing Number.
> But not all values "A-F" can be used. Value "B and C" are standardised as
> digits for special operator functions.
> Value "F" has never been used for the Routing Number or other applications,
> because this value is standardised and used as the "End of Signal"
> indication in the routing process. That means that subsequent digits are
> not accepted and therefore discarded.
>
> So for the Routing Number the following digits can be used:
>
> 1) hexadecimal digits A, D, E;
>    the use of B and C is country specific
>
> or
>
> 2) free number blocks of the numbering range
>
> Please take this correction into account if the relevant text will be
> included in the next draft of TRIP.
>
> Kind regards,
>
> Monika Muench
> Alcatel SEL AG, Germany
>
> ----------
> From: james.yu@neustar.com
> To: jdrosen@dynamicsoft.com
> Cc: hsalama@cisco.com; iptel@lists.bell-labs.com
> Subject: RE: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
> Date: Friday, June 09, 2000 6:38AM
>
> My understanding about the reason of using the digit A, B, ... or F is to
> avoid taking away the valid E.164 numbers from being assigned as subscriber
> numbers.  In ISUP Called Party Number parameter, there is a Nature of
> Address (NOA) field.  Some countries assigned a new NOA value
> (country-specific) to indicate that routing number is concatenated with the
> dialed called party number and may implicitly indicate that NP database dip
> has been performed.  If we use "rn" parameter in tel URL to carry the
> routing number, we may need to define the rn in a way to allow 0-9 and A-F.
>
> The routing number, for example, "5xxxxx+ additional digits defined by the
> assigned carrier" in UK, may have hierarchy where "5" indicates a routing
> number for NP and "xxxxx" identifies the carrier.  The "additional digits"
> may have hierarchy.  For example, BT may define the routing number to
> identify the line card (concentrator) at the switch.  This implies that
> those "additional digits" can be divided at least into two levels: one
> identifies the switch and the other the line card.
>
> James
>
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
> > Sent: Thursday, June 08, 2000 5:38 PM
> > To: James Yu
> > Cc: 'hsalama@cisco.com'; 'iptel@lists.bell-labs.com'
> > Subject: Re: [IPTEL] I-D ACTION:draft-ietf-iptel-trip-02.txt
> >
> >
> >
> >
> > James Yu wrote:
> > >
> > > > >
> > > > > In the North America, the network routing numbers are
> > > > called Location
> > > > > Routing Numbers (LRNs).  They "look" like subscriber phone
> > > > numbers (e.g.,
> > > > > the same length) and can overlap with the subscriber phone
> > > > numbers.  In some
> > > > > European countries, special digit (e.g., A and B) are used
> > > > to differentiate
> > > > > the network routing numbers from the subscriber phone
> > > > numbers.  Those
> > > >
> > > > I think TRIP should be extended advertising these
> > European formats. Is
> > > > this an extended form of E.164, or should it be a separate address
> > > > family?
> > >
> > > Some of the routing numbers (or routig prefixes, as they are usually
> > > prefixes to the dialed called party number) contain the
> > digits outside 0-9
> > > (e.g., A, B, C, D, E, and F).  They are valid routing
> > numbers (national
> > > numbers) either within one network (used by a particular
> > network) or between
> > > networks.  Since they don't just contain digits 0-9, we may
> > not say that
> > > they follow E.164 format.
> >
> > Oh yuck.
> >
> > I'm curious how this impacts the aggregatability of such numbers? Are
> > they assigned in such a way that they can still be aggregated even
> > though there are these random letters in the phone number? Is
> > there any
> > kind of hierarchy in these numbers?
> >
> > -Jonathan R.
> >
> > --
> > Jonathan D. Rosenberg                       72 Eagle Rock Ave.
> > Chief Scientist                             First Floor
> > dynamicsoft                                 East Hanover, NJ 07936
> > jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
> > http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
> > http://www.dynamicsoft.com
> >
>
> _______________________________________________
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--
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Mail Stop SJC6/3, 170 W. Tasman Drive, San Jose, CA 95134
Voice: +1 (408) 527-7147, Fax: +1 (408) 527-1714



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Gonzalo Camarillo wrote:
> 
> Hi,
> 
> Why is it needed the "optional parameter length" in the OPEN message if
> the length of the whole message is already present in the fixed-sized
> header?

TRIP inherited this format from BGP. I don't know why the "optional
parameters length" was included in BGP's OPEN message header. However,
unless I am missing something, removing this two-byte field will not
save us much, because the OPEN message only occurs once a session
anyways.

Hussein

> 
> Thanks,
> 
> Gonzalo
> --
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> Finland
> 
> _______________________________________________
> IPTEL mailing list
> IPTEL@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/iptel

-- 
Hussein F. Salama
Cisco Systems
Mail Stop SJC6/3, 170 W. Tasman Drive, San Jose, CA 95134
Voice: +1 (408) 527-7147, Fax: +1 (408) 527-1714


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Where can I get free ASN.1 Compiler?


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From iptel-admin@lists.bell-labs.com  Wed Jun 14 12:44:56 2000
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From: "Erickson Trejo-Reyes" <eenet@electeng.leeds.ac.uk>
To: "James Yu" <james.yu@neustar.com>, <hsalama@cisco.com>
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Hi everybody,

=20

Does any of you have some references for specific call admission control =
techniques that might have been considered for traffic engineering in =
the IP for avoiding congestion in networks carrying telephony flows? =
Thanks in advance,


Erickson Trejo-Reyes
University of Leeds
Institute of Integrated Information Systems
eenet@electeng.leeds.ac.uk
Tel: +44 (113) 233-2019
Fax: +44 (113) 233-2032

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<P class=3DMsoNormal><SPAN lang=3DEN-US=20
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mso-ansi-language: EN-US">Hi=20
everybody,</SPAN><SPAN lang=3DEN-US=20
style=3D"mso-ansi-language: EN-US"><O:P></O:P></SPAN></P>
<P class=3DMsoNormal><SPAN lang=3DEN-US=20
style=3D"mso-ansi-language: EN-US">&nbsp;<O:P></O:P></SPAN></P><SPAN =
lang=3DEN-US=20
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mso-ansi-language: EN-US; mso-fareast-font-family: Times New Roman; =
mso-bidi-font-family: Times New Roman; mso-fareast-language: ES; =
mso-bidi-language: AR-SA">Does=20
any of you have some references for specific call admission control =
techniques=20
that might have been considered for traffic engineering in the IP for =
avoiding=20
congestion in networks carrying telephony flows? Thanks in=20
advance,</SPAN></FONT></DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2>Erickson =
Trejo-Reyes</FONT></DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2></FONT>
<DIV><FONT face=3D"Bookman Old Style" size=3D2>University of =
Leeds</FONT></DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2>Institute of Integrated =
Information=20
Systems</FONT></DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2><A=20
href=3D"mailto:eenet@electeng.leeds.ac.uk">eenet@electeng.leeds.ac.uk</A>=
</FONT></DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2>Tel: +44 (113) =
233-2019</FONT></DIV>
<DIV><FONT face=3D"Bookman Old Style" size=3D2>Fax: +44 (113)=20
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From iptel-admin@lists.bell-labs.com  Wed Jun 14 12:50:30 2000
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From: "Lesley Scott" <lesley.scott@aculab.com>
To: <iptel@lists.bell-labs.com>
Date: Wed, 14 Jun 2000 17:50:16 +0100
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Subject: [IPTEL] h.245 logical channel conflict resolution
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Hi,

I'm looking for some input into the area of conflict resolution with
unidirectional logical channel signalling procedures.  My understanding of
the H.245 specification is that the master terminal is expected to recognize
conflicts between simultaneous open logical channel requests.  If the master
receives an open logical channel request from the slave which it determines
to be incompatible with an open logical channel request that it has sent, it
must reject the request from the slave.  This being the case, when an
OpenLogicalChannelReject is received by the slave terminal, how should it
respond ???  Should it reattempt to open a logical channel that is
compatible ??  The H.245 spec doesn't seem to address this.

I hope someone can help me to clarify this somewhat shady area.

thanks in advance
Lesley Scott
Aculab plc



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From iptel-admin@lists.bell-labs.com  Wed Jun 14 13:59:46 2000
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Date: Wed, 14 Jun 2000 13:57:56 -0400 (EDT)
From: "Mills, Jeff" <6jpm1@qlink.queensu.ca>
To: iptel@lists.bell-labs.com
Subject: [IPTEL] DTMF and packet loss
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Hello,

I am looking for information regarding the effects of DTMF tones on packet
loss in IP networks.  As well, I'm also looking for more information about
the distortion of DTMF tones during coding and packetization.  Basically
anything to do with DTMF and IP networks.

Any help would be very much appreciated, even any suggestions as to where
I could find good information would be appreciated.


Thanks very much,

Jeff Mills
Queen's University
Kingston, Ontario, Canada



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From iptel-admin@lists.bell-labs.com  Wed Jun 14 15:36:50 2000
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Date: Wed, 14 Jun 2000 17:54:49 -0500
From: Henry Sinnreich <Henry.Sinnreich@WCom.com>
Subject: RE: [IPTEL] Call admission control
In-reply-to: <01bfd61f$b5ab5ce0$dbb00b81@EENET.leeds.ac.uk>
To: Erickson Trejo-Reyes <eenet@electeng.leeds.ac.uk>,
        James Yu <james.yu@neustar.com>, hsalama@cisco.com
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Having enough bandwidth is a first step that comes to my mind. Predications
show voice traffic will be only several % points on IP networks and the
Internet, so why bother to complicate life just for voice. My two cents.

Henry
  -----Original Message-----
  From: iptel-admin@lists.bell-labs.com
[mailto:iptel-admin@lists.bell-labs.com]On Behalf Of Erickson Trejo-Reyes
  Sent: Wednesday, June 14, 2000 11:44 AM
  To: James Yu; hsalama@cisco.com
  Cc: iptel@lists.bell-labs.com
  Subject: [IPTEL] Call admission control


  Hi everybody,



  Does any of you have some references for specific call admission control
techniques that might have been considered for traffic engineering in the IP
for avoiding congestion in networks carrying telephony flows? Thanks in
advance,


  Erickson Trejo-Reyes
  University of Leeds
  Institute of Integrated Information Systems
  eenet@electeng.leeds.ac.uk
  Tel: +44 (113) 233-2019
  Fax: +44 (113) 233-2032

--Boundary_(ID_03xdaFEZr9Zb6FHCHPF6hg)
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<BODY bgColor=3D#ffffff>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D850275222-14062000>Having=20
enough bandwidth is a first step that comes to my mind. Predications =
show voice=20
traffic will be only several % points on IP networks and the Internet, =
so why=20
bother to complicate life just for voice. My two =
cents.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D850275222-14062000></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D850275222-14062000>Henry</SPAN></FONT></DIV>
<BLOCKQUOTE=20
style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: =
0px; PADDING-LEFT: 5px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B>=20
  iptel-admin@lists.bell-labs.com =
[mailto:iptel-admin@lists.bell-labs.com]<B>On=20
  Behalf Of </B>Erickson Trejo-Reyes<BR><B>Sent:</B> Wednesday, June 14, =
2000=20
  11:44 AM<BR><B>To:</B> James Yu; hsalama@cisco.com<BR><B>Cc:</B>=20
  iptel@lists.bell-labs.com<BR><B>Subject:</B> [IPTEL] Call admission=20
  control<BR><BR></DIV></FONT>
  <DIV><FONT color=3D#000000 face=3D"Bookman Old Style" size=3D2>
  <P class=3DMsoNormal><SPAN lang=3DEN-US=20
  style=3D"COLOR: black; FONT-FAMILY: Bookman Old Style; FONT-SIZE: =
10pt; mso-ansi-language: EN-US">Hi=20
  everybody,</SPAN><SPAN lang=3DEN-US=20
  style=3D"mso-ansi-language: EN-US"><O:P></O:P></SPAN></P>
  <P class=3DMsoNormal><SPAN lang=3DEN-US=20
  style=3D"mso-ansi-language: EN-US">&nbsp;<O:P></O:P></SPAN></P><SPAN =
lang=3DEN-US=20
  style=3D"FONT-FAMILY: Bookman Old Style; FONT-SIZE: 10pt; =
mso-ansi-language: EN-US; mso-fareast-font-family: Times New Roman; =
mso-bidi-font-family: Times New Roman; mso-fareast-language: ES; =
mso-bidi-language: AR-SA">Does=20
  any of you have some references for specific call admission control =
techniques=20
  that might have been considered for traffic engineering in the IP for =
avoiding=20
  congestion in networks carrying telephony flows? Thanks in=20
  advance,</SPAN></FONT></DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2>Erickson =
Trejo-Reyes</FONT></DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2></FONT>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2>University of =
Leeds</FONT></DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2>Institute of Integrated =
Information=20
  Systems</FONT></DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2><A=20
  =
href=3D"mailto:eenet@electeng.leeds.ac.uk">eenet@electeng.leeds.ac.uk</A>=
</FONT></DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2>Tel: +44 (113)=20
233-2019</FONT></DIV>
  <DIV><FONT face=3D"Bookman Old Style" size=3D2>Fax: +44 (113)=20
  233-2032</FONT></DIV></DIV></BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_03xdaFEZr9Zb6FHCHPF6hg)--



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At 05:43 PM 6/14/00 +0100, Erickson Trejo-Reyes wrote: 

> Hi everybody,
>
>
>Does any of you have some references for specific call admission control
>techniques that might have been considered for traffic engineering in the
>IP for avoiding congestion in networks carrying telephony flows?
>Thanks in advance,

As you are probably already aware, the problem is both very serious and
rather difficult to resolve. In a public Internet environment, even with
significant free bandwidth, packet voice is frequently completely
incomprehensible. Packet jitter can turn voice into nearly completely
random noise. At present, the preferred procedure is to create a separate
network dedicated to voice. The problem with this is a complete negation of
the primary reason for using VoIP. If you have a separate network anyway,
the existing circuit switched network is a known quantity with very high
reliability and voice over ATM is more efficient than voice over IP.
RSVP-TE, Diffserv, and MPLS appear to be the direction that the industry is
going. The combination will allow for traffic engineering, priority of
service for time critical streams, and failure recovery.

You may want to check the MPLS list. They are working with the issues
related to latency and jitter. They are also working toward traffic
engineering and resolving the failure recovery problem. send an email to
<mpls-request@uu.net>. I do not remember if subscribe needs to be in the
subject line or the body. I think that I sent both.

Larry



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From: Billy Biggs <vektor@div8.net>
To: Larry Cox <lcox@neta.com>
Cc: iptel@lists.bell-labs.com
Subject: Re: [IPTEL] Call admission control
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Larry Cox (lcox@neta.com):

> >Does any of you have some references for specific call admission control
> >techniques that might have been considered for traffic engineering in the
> >IP for avoiding congestion in networks carrying telephony flows?
> >Thanks in advance,
> 
> As you are probably already aware, the problem is both very serious and
> rather difficult to resolve. In a public Internet environment, even with
> significant free bandwidth, packet voice is frequently completely
> incomprehensible. Packet jitter can turn voice into nearly completely
> random noise.

  Larry,

  Not that this is the right forum or even the right thread, I'm very
curious about your statements on the unreliability of packet voice over
the public Internet.  I've now had several trans-atlantic telephone calls
over the public Internet, and often use packet voice for shorter range
(100ms avg lag, 20 hops) Internet calls.  64kbps u-law, RTP encapsulated
in each direction.

  I have never experienced the audio to be 'incomprehensible'.  Even with
no traffic shaping or control, and even when talking to people on cable
modems with reasonable amounts of traffic, it's still much better than
most cellphones.

  Do you have any references to papers or web pages which indicate that
strong admission control or traffic management is useful beyond very
extreme networks or specifc cases of heavy bandwidth applications?

  In general, TCP seems back off quite happily.

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca


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From iptel-admin@lists.bell-labs.com  Thu Jun 15 18:28:11 2000
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Folks,

Some colleagues and I have put together a suite of protocols for IMPP,
based partly on SIP. Its actually seven separate drafts, althouth the
core is just two. They have been submitted to the I-D editor, but you
can grab them right now from:

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-presence-00.txt

   Core presence protocol. We've written it to be understandable by non
SIP experts,
   and to also indicate what parts of SIP you do and don't need for
presence.
   Don't be scared by the size (70 pages), nearly half are example
message flows.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-im-00.txt

   Core IM protocol. Also written to be both a tutorial on SIP and
explain how
   to do IM.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-qauth-00.txt

   A mini-protocol for authorizing subscriptions, to support, among
other things,
   the ability for subscriptions to happen when the presentity is
offline.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-pidf-00.txt

   An XML PIDF format, optional in our architecture.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-lpidf-00.txt

   A really thin pidf, good for wireless devices, since it uses the same
parser
   SIP uses. This is mandatory to support in our architecture.

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-watcherinfo-00.txt

   A format for watcher info. 

http://www.cs.columbia.edu/~jdrosen/papers/draft-rosenberg-impp-buddylist-00.txt

   Something we never discussed on the list before; a buddylist
document, much 
   akin to bookmarks, which you can store on a server. This way, when
you go
   to a different machine, you can fetch your buddylist, and get your
presence
   services there.


Comments and questions are welcome, of course.



-Jonathan R.

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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Why stop at XML for IMPP? Why not XML for SIP? Simple Object Access Protocol
may be the right way. Call it SIP Control by Unified Methods.

SOAP SCUM 


Best regards,
Mike Fox



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From iptel-admin@lists.bell-labs.com  Thu Jun 15 22:35:36 2000
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Hi, 

Call admission control is a traditional concept derived from PSTN, ISDN, SONET and ATM networks. Because there is no global resource management on IP network, any effort on call-based resource allocation or traffic engineering in network layer (IP) is opposite to the fundament of Internet. Existing technologies for providing QoS guaranteed call on Internet is session layer control such as RSVP, RTP and MPLS. But they assume that the network has unlimited resource. If you desire to make traffic engineering on resource-limited Internet and refuse some new calls while the traffic is in congestion somewhere, you need a global resource management server, which surveys the traffic on routers and links, as well as free process capacity in routers and residual bandwidth on links. All access points act as the agent of the server and execute admission control functions. Is it possible? No. The only solution for this issue is hierarchical dynamic statistical routing optimization protocol. It intends to update routing tables dynamically in order to reserve enough free resource on routers and links for further use. It can be used with any resource reservation protocol. I have taken a very careful study on it and propose a framework for it, but I don't know whether it has been realized and implemented by now. 

James

-----Original Message-----
From:	Billy Biggs [SMTP:vektor@div8.net]
Sent:	Wednesday, June 14, 2000 1:50 PM
To:	Larry Cox
Cc:	iptel@lists.bell-labs.com
Subject:	Re: [IPTEL] Call admission control

Larry Cox (lcox@neta.com):

> >Does any of you have some references for specific call admission control
> >techniques that might have been considered for traffic engineering in the
> >IP for avoiding congestion in networks carrying telephony flows?
> >Thanks in advance,
> 
> As you are probably already aware, the problem is both very serious and
> rather difficult to resolve. In a public Internet environment, even with
> significant free bandwidth, packet voice is frequently completely
> incomprehensible. Packet jitter can turn voice into nearly completely
> random noise.

  Larry,

  Not that this is the right forum or even the right thread, I'm very
curious about your statements on the unreliability of packet voice over
the public Internet.  I've now had several trans-atlantic telephone calls
over the public Internet, and often use packet voice for shorter range
(100ms avg lag, 20 hops) Internet calls.  64kbps u-law, RTP encapsulated
in each direction.

  I have never experienced the audio to be 'incomprehensible'.  Even with
no traffic shaping or control, and even when talking to people on cable
modems with reasonable amounts of traffic, it's still much better than
most cellphones.

  Do you have any references to papers or web pages which indicate that
strong admission control or traffic management is useful beyond very
extreme networks or specifc cases of heavy bandwidth applications?

  In general, TCP seems back off quite happily.

-- 
Billy Biggs                         vektor@div8.net
http://www.div8.net/billy       wbiggs@uwaterloo.ca







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From: "Suntichai Chusywong" <s4110400@maliwan.psu.ac.th>
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I wonder when Master - Slave Determinaiton message wil be sent, before =
or after sending Terminal Capability Set message? After Determining =
which Endpoint is master, what do we use this result for?

 I don't understand how to encoding H.245 message, what will I do if I =
want to encode these message easily. If you have any tool to help me, =
please send me
       =20
                                                                Thank =
you very much=20
                                                                         =
   Suntichai Chuaywong

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</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I wonder when Master - Slave=20
Determinaiton&nbsp;message wil be sent, before or after sending Terminal =

Capability Set message? After Determining which Endpoint is master, what =
do we=20
use this result for?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D3>&nbsp;I don't understand how to =
encoding H.245=20
message, what will I do if I want to encode these message easily. If you =
have=20
any tool to help me, please send me</FONT></DIV>
<DIV><FONT face=3DArial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
</FONT></DIV>
<DIV><FONT face=3DArial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; Thank you very much </FONT></DIV>
<DIV><FONT face=3DArial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
Suntichai Chuaywong</FONT></DIV></BODY></HTML>

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To: "'Suntichai Chusywong'" <s4110400@maliwan.psu.ac.th>,
        iptel <iptel@lists.research.bell-labs.com>
Subject: RE: [IPTEL] Master - Slave Determination
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Terminal Capability Set comes first.  See Rec. H.323 section 8.2.  Encoding
H.245 requires detailed understanding of ASN.1 Packed Encoding Rules (very
few people are willing to undertake this task) or use of a commercial
compiler provided by OSS.  You can also buy stacks from various providers.
If you want to acquire the documents defining ASN.1 Packed Encoding Rules
(ITU-T Recommendations X.680 and X.691), you can purchase them from the
ITU-T at http://www.itu.int/ITU-T/index.html 

> -----Original Message-----
> From:	Suntichai Chusywong [SMTP:s4110400@maliwan.psu.ac.th]
> Sent:	Monday, June 19, 2000 2:20 PM
> To:	iptel
> Cc:	s4110400
> Subject:	[IPTEL] Master - Slave Determination
> 
> I wonder when Master - Slave Determinaiton message wil be sent, before or
> after sending Terminal Capability Set message? After Determining which
> Endpoint is master, what do we use this result for?
>  
>  I don't understand how to encoding H.245 message, what will I do if I
> want to encode these message easily. If you have any tool to help me,
> please send me
>         
>                                                                 Thank you
> very much 
>  
> Suntichai Chuaywong

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<TITLE>RE: [IPTEL] Master - Slave Determination</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" SIZE=3D2 FACE=3D"Arial">Terminal Capability =
Set comes first.&nbsp; See Rec. H.323 section 8.2.&nbsp; Encoding H.245 =
requires detailed understanding of ASN.1 Packed Encoding Rules (very =
few people are willing to undertake this task) or use of a commercial =
compiler provided by OSS.&nbsp; You can also buy stacks from various =
providers.&nbsp; If you want to acquire the documents defining ASN.1 =
Packed Encoding Rules (ITU-T Recommendations X.680 and X.691), you can =
purchase them from the ITU-T at<U> <A =
HREF=3D"http://www.itu.int/ITU-T/index.html" =
TARGET=3D"_blank">http://www.itu.int/ITU-T/index.html</A></U> =
</FONT></P>
<UL>
<P><FONT SIZE=3D1 FACE=3D"Arial">-----Original Message-----</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">From:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Suntichai Chusywong =
[SMTP:s4110400@maliwan.psu.ac.th]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Monday, June 19, 2000 2:20 PM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">iptel</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Cc:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">s4110400</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">Subject:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT>=
</B> <FONT SIZE=3D1 FACE=3D"Arial">[IPTEL] Master - Slave =
Determination</FONT>
</P>

<P><FONT SIZE=3D4 FACE=3D"Arial">I wonder when Master - Slave =
Determinaiton</FONT><FONT SIZE=3D4 FACE=3D"Arial">=A0message wil be =
sent, before or after sending Terminal Capability Set message? After =
Determining which Endpoint is master, what do we use this result =
for?</FONT></P>

<P><FONT SIZE=3D5 FACE=3D"Cordia New">=A0</FONT>
<BR><FONT SIZE=3D5 FACE=3D"Arial">=A0I don't understand how to encoding =
H.245 message, what will I do if I want to encode these message easily. =
If you have any tool to help me, please send me</FONT></P>

<P><FONT SIZE=3D5 FACE=3D"Arial">=A0=A0=A0 =A0=A0=A0</FONT>=20
<BR><FONT SIZE=3D5 FACE=3D"Arial">=A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =
=A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =
=A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 Thank you =
very much</FONT>=20
<BR><FONT SIZE=3D5 FACE=3D"Arial">=A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =
=A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =
=A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =A0=A0=A0 =
=A0=A0=A0 =A0=A0=A0 Suntichai Chuaywong</FONT>
</P>
</UL>
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</HTML>
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From: "Suntichai Chusywong" <s4110400@maliwan.psu.ac.th>
To: <iptel@lists.research.bell-labs.com>
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I wonder when Master - Slave Determinaiton message wil be sent, before =
or after sending Terminal Capability Set message? After Determining =
which Endpoint is master, what do we use this result for?

       =20
                                                                Thank =
you very much=20
                                                                         =
   Suntichai Chuaywong

------=_NextPart_000_0015_01BFDB0D.B8B216C0
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</HEAD>
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<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>I wonder when Master - Slave=20
Determinaiton&nbsp;message wil be sent, before or after sending Terminal =

Capability Set message? After Determining which Endpoint is master, what =
do we=20
use this result for?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
</FONT></DIV>
<DIV><FONT face=3DArial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; Thank you very much </FONT></DIV>
<DIV><FONT face=3DArial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; =
&nbsp;&nbsp;&nbsp;=20
Suntichai Chuaywong</FONT></DIV></FONT></DIV></BODY></HTML>

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May have an answer to this - would be interested in any other input though:

Referencing H.245 (February 6, 1998) - it appears that when the slave sends
out the OpenLogicalChannel, it then goes into the Awaiting Establishment
state - (refer to Figure 10 in section 8.4.2.5.)  Upon receipt of the
OpenLogicalChannelRej - the slave should reset T103, send a Release
Indication to the protocol user, and go into the Released state (Figure 13
section 8.4.4.5).  At this point, it appears from figure 10 that it is
"legal" from the H.245 protocol standpoint to reenter the awaiting
establishment state, which would imply that the protocol user has the option
to resend another OpenLogicalChannel.

-----Original Message-----
From: Lesley Scott [mailto:lesley.scott@aculab.com]
Sent: Wednesday, June 14, 2000 9:50 AM
To: iptel@lists.bell-labs.com
Subject: [IPTEL] h.245 logical channel conflict resolution


Hi,

I'm looking for some input into the area of conflict resolution with
unidirectional logical channel signalling procedures.  My understanding of
the H.245 specification is that the master terminal is expected to recognize
conflicts between simultaneous open logical channel requests.  If the master
receives an open logical channel request from the slave which it determines
to be incompatible with an open logical channel request that it has sent, it
must reject the request from the slave.  This being the case, when an
OpenLogicalChannelReject is received by the slave terminal, how should it
respond ???  Should it reattempt to open a logical channel that is
compatible ??  The H.245 spec doesn't seem to address this.

I hope someone can help me to clarify this somewhat shady area.

thanks in advance
Lesley Scott
Aculab plc



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To: "'Suntichai Chusywong'" <s4110400@maliwan.psu.ac.th>,
        "'iptel@lists.bell-labs.com'" <iptel@lists.bell-labs.com>
Subject: RE: [IPTEL] Master - Slave Determination
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Since the portion of your question regarding message order has already been
answered - I will take a crack at the second part regarding what the result
is used for. 
 
There are cases where two or more endpoints/terminals may initiate events
simultaneously where there is a resource conflict - in this case it is
necessary for the master endpoint to in order to resolve the conflict.  A
very good example is the logical channel determination, where each endpoint
needs to establish a unique channel for the data streams, and since they are
half-duplex, each two way "conversation" requires two channels.  Thus - if
both endpoints choose the same logical channel for it's part of the
conversation, this results in a conflict.  How this is "resolved" is an
issue of another message on this board as well.  
 
Additionally - in H.235, the master is also responsible for the distribution
of the encryption keys for media channels to other terminals.  

-----Original Message-----
From: Suntichai Chusywong [mailto:s4110400@maliwan.psu.ac.th]
Sent: Tuesday, June 20, 2000 9:18 AM
To: iptel@lists.research.bell-labs.com
Subject: [IPTEL] Master - Slave Determination



I wonder when Master - Slave Determinaiton message wil be sent, before or
after sending Terminal Capability Set message? After Determining which
Endpoint is master, what do we use this result for?
 
        
                                                                Thank you
very much 
 
Suntichai Chuaywong


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<META content="MSHTML 5.00.2614.3500" name=GENERATOR>
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<BODY bgColor=#ffffff>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN class=990555223-20062000>Since 
the portion of your question regarding message order has already been answered - 
I will take a crack at the second part regarding what the result is used for. 
</SPAN></FONT></FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN 
class=990555223-20062000></SPAN></FONT>&nbsp;</FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN class=990555223-20062000>There 
are cases where two or more endpoints/terminals may initiate events 
simultaneously where there is a resource conflict - in this case it is necessary 
for&nbsp;the master endpoint to in order to resolve&nbsp;the conflict.&nbsp; A 
very good example is the logical channel determination, where each endpoint 
needs to establish a unique channel for the data streams, and since they are 
half-duplex, each two way "conversation" requires two channels.&nbsp; Thus - if 
both endpoints choose the same logical channel for it's part of the 
conversation, this results in a conflict.&nbsp;&nbsp;How&nbsp;this 
is&nbsp;"resolved" is&nbsp;an issue of another message on this board as 
well.&nbsp; </SPAN></FONT></FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN 
class=990555223-20062000></SPAN></FONT>&nbsp;</FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN 
class=990555223-20062000>Additionally - in H.235, the master is also responsible 
for the distribution of the encryption keys for media channels to other 
terminals.&nbsp; </SPAN></FONT></FONT></DIV>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT SIZE=5><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Suntichai Chusywong 
  [mailto:s4110400@maliwan.psu.ac.th]<BR><B>Sent:</B> Tuesday, June 20, 2000 
  9:18 AM<BR><B>To:</B> iptel@lists.research.bell-labs.com<BR><B>Subject:</B> 
  [IPTEL] Master - Slave Determination<BR><BR></FONT></DIV></FONT>
  <DIV><FONT SIZE=5><FONT size=5><FONT face=Arial size=2></FONT>
  </FONT><DIV><FONT face=Arial size=2>I wonder when Master - Slave 
  Determinaiton&nbsp;message wil be sent, before or after sending Terminal 
  Capability Set message? After Determining which Endpoint is master, what do we 
  use this result for?</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=Arial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; </FONT></DIV>
  <DIV><FONT face=Arial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; Thank you very much </FONT></DIV>
  <DIV><FONT face=Arial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  Suntichai Chuaywong</FONT></DIV></FONT></DIV></BLOCKQUOTE></BODY></HTML>

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From: Ron Traficante <ront@bzln.com>
To: "'Suntichai Chusywong'" <s4110400@maliwan.psu.ac.th>,
        iptel@lists.research.bell-labs.com
Subject: RE: [IPTEL] Master - Slave Determination
Date: Tue, 20 Jun 2000 17:04:08 -0700
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Since the portion of your question regarding message order has already been
answered - I will take a crack at the second part regarding what the result
is used for. 
 
There are cases where two or more endpoints/terminals may initiate events
simultaneously where there is a resource conflict - in this case it is
necessary for the master endpoint to in order to resolve the conflict.  A
very good example is the logical channel determination, where each endpoint
needs to establish a unique channel for the data streams, and since they are
half-duplex, each two way "conversation" requires two channels.  Thus - if
both endpoints choose the same logical channel for it's part of the
conversation, this results in a conflict.  How this is "resolved" is an
issue of another message on this board as well.  
 
Additionally - in H.235, the master is also responsible for the distribution
of the encryption keys for media channels to other terminals.  

-----Original Message-----
From: Suntichai Chusywong [mailto:s4110400@maliwan.psu.ac.th]
Sent: Tuesday, June 20, 2000 9:18 AM
To: iptel@lists.research.bell-labs.com
Subject: [IPTEL] Master - Slave Determination



I wonder when Master - Slave Determinaiton message wil be sent, before or
after sending Terminal Capability Set message? After Determining which
Endpoint is master, what do we use this result for?
 
        
                                                                Thank you
very much 
 
Suntichai Chuaywong


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<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN class=990555223-20062000>Since 
the portion of your question regarding message order has already been answered - 
I will take a crack at the second part regarding what the result is used for. 
</SPAN></FONT></FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN 
class=990555223-20062000></SPAN></FONT>&nbsp;</FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN class=990555223-20062000>There 
are cases where two or more endpoints/terminals may initiate events 
simultaneously where there is a resource conflict - in this case it is necessary 
for&nbsp;the master endpoint to in order to resolve&nbsp;the conflict.&nbsp; A 
very good example is the logical channel determination, where each endpoint 
needs to establish a unique channel for the data streams, and since they are 
half-duplex, each two way "conversation" requires two channels.&nbsp; Thus - if 
both endpoints choose the same logical channel for it's part of the 
conversation, this results in a conflict.&nbsp;&nbsp;How&nbsp;this 
is&nbsp;"resolved" is&nbsp;an issue of another message on this board as 
well.&nbsp; </SPAN></FONT></FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN 
class=990555223-20062000></SPAN></FONT>&nbsp;</FONT></DIV>
<DIV><FONT SIZE=5><FONT color=#0000ff face=Arial size=2><SPAN 
class=990555223-20062000>Additionally - in H.235, the master is also responsible 
for the distribution of the encryption keys for media channels to other 
terminals.&nbsp; </SPAN></FONT></FONT></DIV>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
  <DIV align=left class=OutlookMessageHeader dir=ltr><FONT SIZE=5><FONT face=Tahoma 
  size=2>-----Original Message-----<BR><B>From:</B> Suntichai Chusywong 
  [mailto:s4110400@maliwan.psu.ac.th]<BR><B>Sent:</B> Tuesday, June 20, 2000 
  9:18 AM<BR><B>To:</B> iptel@lists.research.bell-labs.com<BR><B>Subject:</B> 
  [IPTEL] Master - Slave Determination<BR><BR></FONT></DIV></FONT>
  <DIV><FONT SIZE=5><FONT size=5><FONT face=Arial size=2></FONT>
  </FONT><DIV><FONT face=Arial size=2>I wonder when Master - Slave 
  Determinaiton&nbsp;message wil be sent, before or after sending Terminal 
  Capability Set message? After Determining which Endpoint is master, what do we 
  use this result for?</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=Arial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; </FONT></DIV>
  <DIV><FONT face=Arial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; Thank you very much </FONT></DIV>
  <DIV><FONT face=Arial>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; 
  Suntichai Chuaywong</FONT></DIV></FONT></DIV></BLOCKQUOTE></BODY></HTML>

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From iptel-admin@lists.bell-labs.com  Wed Jun 21 12:10:31 2000
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Date: Wed, 21 Jun 2000 17:08:27 +0000 (GMT)
From: James Undery <jundery@ubiquity.net>
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Hi,

	Just a quick observation, I know The CPL DTD got a mauling on the
iptel mailing group a while back, but I think Nilo missed a mistake in the
DTD,

	<!ELEMENT timezone ( standard+, daylight* ) >

is what the text describes, rather than

	<!ELEMENT timezone ( standard, daylight? ) >

in the draft's DTD.

I am also interested in how seriously XML schemas are being considered as
I think there are some areas where the DTD could be tightened up, but,
don't want to waste time if the DTD is binned in favour of a schema.

James Undery



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To: Suntichai Chusywong <s4110400@maliwan.psu.ac.th>
Cc: iptel@lists.research.bell-labs.com
Subject: Re: [IPTEL] Master - Slave Determination
References: <002f01bfda1b$1489a380$8300a8c0@coe.psu.ac.th>
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This list is not meant for H.323 questions. Please use the H.323
implementors list (which I always lose my reference to --- someone?)

-Jonathan R.

> Suntichai Chusywong wrote:
> 
> I wonder when Master - Slave Determinaiton message wil be sent, before
> or after sending Terminal Capability Set message? After Determining
> which Endpoint is master, what do we use this result for?
> 
>  I don't understand how to encoding H.245 message, what will I do if I
> want to encode these message easily. If you have any tool to help me,
> please send me
> 
>                                                                 Thank
> you very much
> 
>         Suntichai Chuaywong

-- 
Jonathan D. Rosenberg                       72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
http://www.dynamicsoft.com



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To: lennox@cs.columbia.edu
Cc: iptel@lists.bell-labs.com, "Trip Ingle" <trip@nortelnetworks.com>
Subject: [IPTEL] draft-ietf-cpl-01 the DTD
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Hi,

I need a few clarifications regarding the timezone element. 

The timezone element as described in the text seems to be

	<!ELEMENT timezone ( standard, daylight? )+ > 

rather than

	<!ELEMENT timezone ( standard, daylight? ) >

in the draft's DTD.

My understanding is that the standard and daylight element(if present) have
to occur as a pair with atmost one occurence for a  particular year and
timezone. The DTD, however, does allow the possibility of defining more than
one set of  (standard & daylight) per year per timezone. Is this required ?

Also the entity ZoneParams has  attribute 'offset' missing from the
attribute list in the DTD. 


Thanks,
Shantala.

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charset=3Diso-8859-1">
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5.5.2651.65">
<TITLE>[IPTEL] draft-ietf-cpl-01 the DTD</TITLE>
</HEAD>
<BODY>

<P><FONT SIZE=3D2>Hi,</FONT>
</P>

<P><FONT SIZE=3D2>I need a few clarifications regarding the timezone =
element. </FONT>
</P>

<P><FONT SIZE=3D2>The timezone element as described in the text seems =
to be</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&lt;!ELEMENT timezone ( standard, daylight? )+ &gt; </FONT>
</P>

<P><FONT SIZE=3D2>rather than</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT =
SIZE=3D2>&lt;!ELEMENT timezone ( standard, daylight? ) &gt;</FONT>
</P>

<P><FONT SIZE=3D2>in the draft's DTD.</FONT>
</P>

<P><FONT SIZE=3D2>My understanding is that the standard and daylight =
element(if present) have to occur as a pair with atmost one occurence =
for a&nbsp; particular year and timezone. The DTD, however, does allow =
the possibility of defining more than one set of&nbsp; (standard &amp; =
daylight) per year per timezone. Is this required ?</FONT></P>

<P><FONT SIZE=3D2>Also the entity ZoneParams has&nbsp; attribute =
'offset' missing from the attribute list in the DTD. </FONT>
</P>
<BR>

<P><FONT SIZE=3D2>Thanks,</FONT>
<BR><FONT SIZE=3D2>Shantala.</FONT>
</P>

</BODY>
</HTML>
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On Thu, 22 Jun 2000, Shantala Manchenahally wrote:

> Hi,
> 
> I need a few clarifications regarding the timezone element. 
> 
> The timezone element as described in the text seems to be
> 
> 	<!ELEMENT timezone ( standard, daylight? )+ > 

This is stricter than my version and doesn't allow a universal 'standard'
with 'daylight' different for different years. 

> 
> rather than
> 
> 	<!ELEMENT timezone ( standard, daylight? ) >
> 
> in the draft's DTD.
> 
> My understanding is that the standard and daylight element(if present) have
> to occur as a pair with atmost one occurence for a  particular year and
> timezone. The DTD, however, does allow the possibility of defining more than
> one set of  (standard & daylight) per year per timezone. Is this required ?
> 

Your version doesn't provide that, your reading should be expressed as

	<!ELEMENT timezone ( standard | ( standard, daylight)+ ) >

obviously some semantics are lost here.

> Also the entity ZoneParams has  attribute 'offset' missing from the
> attribute list in the DTD. 
> 

not in 01

James Undery



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Subject: Re: [IPTEL] H.323 Reflector [Was: Master - Slave Determination]
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Here it is.

Orit Levin
RADVision Inc.
575 Corporate Drive Suite 420
Mahwah, NJ 07430
Tel: 1 201 529 4300  (230)
Fax: 1 201 529 3516
www.radvision.com
orit@radvision.com

Below is the description for two mailing lists:
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----------------
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-------------------------------------------------


-----Original Message-----
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
To: Suntichai Chusywong <s4110400@maliwan.psu.ac.th>
Cc: iptel@lists.research.bell-labs.com <iptel@lists.research.bell-labs.com>
Date: Thursday, June 22, 2000 10:12 AM
Subject: Re: [IPTEL] Master - Slave Determination


>This list is not meant for H.323 questions. Please use the H.323
>implementors list (which I always lose my reference to --- someone?)
>
>-Jonathan R.
>
>> Suntichai Chusywong wrote:
>> 
>> I wonder when Master - Slave Determinaiton message wil be sent, before
>> or after sending Terminal Capability Set message? After Determining
>> which Endpoint is master, what do we use this result for?
>> 
>>  I don't understand how to encoding H.245 message, what will I do if I
>> want to encode these message easily. If you have any tool to help me,
>> please send me
>> 
>>                                                                 Thank
>> you very much
>> 
>>         Suntichai Chuaywong
>
>-- 
>Jonathan D. Rosenberg                       72 Eagle Rock Ave.
>Chief Scientist                             First Floor
>dynamicsoft                                 East Hanover, NJ 07936
>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
>http://www.dynamicsoft.com
>
>
>
>_______________________________________________
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-----Original Message-----
From: Ron Traficante [mailto:ront@bzln.com]
Sent: 20 June 2000 21:34
To: 'Suntichai Chusywong'; iptel@lists.research.bell-labs.com
Subject: RE: [IPTEL] Master - Slave Determination


Since the portion of your question regarding message order has already been
answered - I will take a crack at the second part regarding what the result
is used for.

There are cases where two or more endpoints/terminals may initiate events
simultaneously where there is a resource conflict - in this case it is
necessary for the master endpoint to in order to resolve the conflict.  A
very good example is the logical channel determination, where each endpoint
needs to establish a unique channel for the data streams, and since they are
half-duplex, each two way "conversation" requires two channels.  Thus - if
both endpoints choose the same logical channel for it's part of the
conversation, this results in a conflict.  How this is "resolved" is an
issue of another message on this board as well.

Additionally - in H.235, the master is also responsible for the distribution
of the encryption keys for media channels to other terminals.

-----Original Message-----
From: Suntichai Chusywong [mailto:s4110400@maliwan.psu.ac.th]
Sent: Tuesday, June 20, 2000 9:18 AM
To: iptel@lists.research.bell-labs.com
Subject: [IPTEL] Master - Slave Determination



I wonder when Master - Slave Determinaiton message wil be sent, before or
after sending Terminal Capability Set message? After Determining which
Endpoint is master, what do we use this result for?


                                                                Thank you
very much

Suntichai Chuaywong








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Date: Thu, 22 Jun 2000 14:34:06 -0400
From: Dave Walker <drwalker@ss8networks.com>
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Subject: [IPTEL] TRIP tie-breakers
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In section 10.2.2.1 of draft-ietf-iptel-trip-02.txt, there are
three means of breaking ties among equally preferred routes to 
the same destination.  One is using MED.  The other two use the 
TRIP ID of some LS:

a) If the local LS is configured to use the MultiExitDisc
   attribute to break ties, and the candidate routes differ 
   in the value of the MultiExitDisc attribute, then select the 
   route that has the larger value of MultiExitDisc.     
b) If at least one of the routes was advertised by an LS in a
   neighboring ITAD, then select the route that was advertised 
   by the LS that has the smallest TRIP ID.
c) Otherwise, select the route that was advertised by the internal
   LS that has the lowest TRIP ID.

I have a simple question on (a).  Since this algorithm is pretty
much lifted from BGP, why does it give preference to the largest
value of MED, when BGP uses the smallest?  In other respects, the
usage of MED in TRIP is the same as in BGP (i.e. routes with lower
MED value are preferred).

I don't understand the difference between (b) and (c).  It seems 
to me that they're the same.  In both cases, the lowest TRIP ID 
is selected.  It's possible that (b) intends that the lowest 
TRIP ID be selected from among the external routes, but that's 
not what it says (according to my interpretation).  BGP's algorithm
for breaking phase 2 ties explicitly says that the selection is 
only from among the external routes.  Because of the change in
the use of MED for tie-breaking (see above), I'm uneasy about 
using BGP for what I think is undefined functionality in TRIP.

Now if the intention for (b) is that the preferred route is
the one that was advertised by an external LS (as per BGP), my 
next question is why?  It seems to me that if I were a service 
provider, I'd prefer to use my own gateways (internal routes) 
over someone else's.

Regards,

Dave Walker
SS8 Networks
Ottawa, Canada


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From iptel-admin@lists.bell-labs.com  Thu Jun 22 15:18:27 2000
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From: Dhaval Shah <dhaval@cisco.com>
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Subject: Re: [IPTEL] Length in OPEN
To: hsalama@cisco.com
Date: Thu, 22 Jun 2000 12:18:01 -0700 (PDT)
Cc: Gonzalo.Camarillo@lmf.ericsson.se (Gonzalo Camarillo),
        iptel@lists.bell-labs.com (iptel)
In-Reply-To: <39467363.22030676@cisco.com> from "Hussein F. Salama" at Jun 13, 2000 10:46:11 AM
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> Gonzalo Camarillo wrote:
> > 
> > Hi,
> > 
> > Why is it needed the "optional parameter length" in the OPEN message if
> > the length of the whole message is already present in the fixed-sized
> > header?
> 
> TRIP inherited this format from BGP. I don't know why the "optional
> parameters length" was included in BGP's OPEN message header. However,
> unless I am missing something, removing this two-byte field will not
> save us much, because the OPEN message only occurs once a session
> anyways.

Is there any reason to keep it if TRIP doesn't need it? Also, the optional 
parameter length in BGP is 1 byte. Any reason why TRIP has 2 bytes?

Thanx,
Dhaval

> 
> Hussein
> 
> > 
> > Thanks,
> > 
> > Gonzalo
> > --
> > Gonzalo Camarillo         Phone :  +358  9 299 33 71
> > Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> > Telecom R&D               Fax   :  +358  9 299 30 52
> > FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
> > Finland
> > 
> > _______________________________________________
> > IPTEL mailing list
> > IPTEL@lists.bell-labs.com
> > http://lists.bell-labs.com/mailman/listinfo/iptel
> 
> -- 
> Hussein F. Salama
> Cisco Systems
> Mail Stop SJC6/3, 170 W. Tasman Drive, San Jose, CA 95134
> Voice: +1 (408) 527-7147, Fax: +1 (408) 527-1714
> 
> 
> _______________________________________________
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> http://lists.bell-labs.com/mailman/listinfo/iptel
> 
> 



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From iptel-admin@lists.bell-labs.com  Fri Jun 23 20:06:06 2000
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From: "Hussein F. Salama" <hsalama@cisco.com>
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To: Dave Walker <drwalker@ss8networks.com>
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Subject: Re: [IPTEL] TRIP tie-breakers
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Dave Walker wrote:
> 
> In section 10.2.2.1 of draft-ietf-iptel-trip-02.txt, there are
> three means of breaking ties among equally preferred routes to
> the same destination.  One is using MED.  The other two use the
> TRIP ID of some LS:
> 
> a) If the local LS is configured to use the MultiExitDisc
>    attribute to break ties, and the candidate routes differ
>    in the value of the MultiExitDisc attribute, then select the
>    route that has the larger value of MultiExitDisc.

This statement needs to be tightened up, such that only routes from the
same neighboring ITAD are compared with each other in step a).

> b) If at least one of the routes was advertised by an LS in a
>    neighboring ITAD, then select the route that was advertised
>    by the LS that has the smallest TRIP ID.
> c) Otherwise, select the route that was advertised by the internal
>    LS that has the lowest TRIP ID.
> 
> I have a simple question on (a).  Since this algorithm is pretty
> much lifted from BGP, why does it give preference to the largest
> value of MED, when BGP uses the smallest?  In other respects, the
> usage of MED in TRIP is the same as in BGP (i.e. routes with lower
> MED value are preferred).

Thanks for pointing this out. The purpose of this divergence from BGP
was to make the LocalPreference attribute and the MED attribute
consistent: higher value means higher preference. It seems that the
change was made in this paragraph only. This will be fixed in the next
revision.

> 
> I don't understand the difference between (b) and (c).  It seems
> to me that they're the same.

They're different. Step b) eliminates all internal routes and breaks the
tie between the external routes. TRIP gets to Step c) only if the tie is
among internal routes only. Step c) breaks the tie between these
internal routes.

> In both cases, the lowest TRIP ID
> is selected.  It's possible that (b) intends that the lowest
> TRIP ID be selected from among the external routes,

Correct.

> but that's
> not what it says (according to my interpretation).  BGP's algorithm
> for breaking phase 2 ties explicitly says that the selection is
> only from among the external routes.  Because of the change in
> the use of MED for tie-breaking (see above), I'm uneasy about
> using BGP for what I think is undefined functionality in TRIP.

We had a discussion about the use of MED in TRIP before (check iptel
archives), but we decided at the end to use it exactly the same way it
is defined in BGP, with the exception mentioned above on how to
interpret the MED value.

> 
> Now if the intention for (b) is that the preferred route is
> the one that was advertised by an external LS (as per BGP), my
> next question is why?  It seems to me that if I were a service
> provider, I'd prefer to use my own gateways (internal routes)
> over someone else's.

In BGP, when a BGP speaker receives a route from an external peer and
another route from an internal peer, both to the same destination, this
means that the destination must be located outside the local AS. The
route from an external peer is preferred, because it is the shortest
path out of the local AS.

The wording may not be clear in TRIP. However, TRIP's approach is
different from BGP's. A "route advertised by an internal LS" is a route
to a local GW in the same ITAD. A "route advertised by an LS in a
neighboring ITAD" is a route to a GW outside the local ITAD. It doesn't
matter whether the route was received from an internal peer or from an
external peer, what matters is the location of the terminating GW. 

What you're proposing makes sense. I think, there will be more ITADs
which encourage calls to terminate on their local than ITADs which
prefer to terminate calls on external GWs and keep their local GWs as a
last resort only. So it should be preferable to route calls to local GWs
and not to external GWs.

Hussein

> 
> Regards,
> 
> Dave Walker
> SS8 Networks
> Ottawa, Canada
> 
> _______________________________________________
> IPTEL mailing list
> IPTEL@lists.bell-labs.com
> http://lists.bell-labs.com/mailman/listinfo/iptel


-- 
Hussein F. Salama
Cisco Systems
Mail Stop SJC6/3, 170 W. Tasman Drive, San Jose, CA 95134
Voice: +1 (408) 527-7147, Fax: +1 (408) 527-1714


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--=====================_1302240==_.ALT
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Mike

How long did it take you to think of this acronym? If less than an hour -- you
need a life, man!

;-)

cheers

At 03:18 PM 6/15/2000 -0700, Fox, Mike wrote:
>Why stop at XML for IMPP? Why not XML for SIP? Simple Object Access Protocol
>may be the right way. Call it SIP Control by Unified Methods.
>
>SOAP SCUM 
>
>
>Best regards,
>Mike Fox
>
> 
*************************************
"At the end of the day... the most committed win!"

James M. Polk
Sr. Product Manager, Multiservice Architecture and Standards
Enterprise Voice Business Unit
Cisco Systems
Dallas, Texas
w) 972.813.5208
f)  972.813.5280
www.cisco.com
--=====================_1302240==_.ALT
Content-Type: text/html; charset="us-ascii"

<html><div>Mike</div>
<br>
<div>How long did it take you to think of this acronym? If less than an
hour -- you need a life, man!</div>
<br>
<div>;-)</div>
<br>
<div>cheers</div>
<br>
<div>At 03:18 PM 6/15/2000 -0700, Fox, Mike wrote:</div>
<div>&gt;Why stop at XML for IMPP? Why not XML for SIP? Simple Object
Access Protocol</div>
<div>&gt;may be the right way. Call it SIP Control by Unified
Methods.</div>
<div>&gt;</div>
<div>&gt;SOAP SCUM </div>
<div>&gt;</div>
<div>&gt;</div>
<div>&gt;Best regards,</div>
<div>&gt;Mike Fox</div>
<div>&gt;</div>
&gt;
<br>

<div align="center">
*************************************<br>
&quot;At the end of the day... the most committed win!&quot;<br>
<br>
</div>
James M. Polk<br>
Sr. Product Manager, Multiservice Architecture and Standards<br>
Enterprise Voice Business Unit<br>
Cisco Systems<br>
Dallas, Texas<br>
w) 972.813.5208<br>
f)&nbsp; 972.813.5280<br>
<a href="http://www.cisco.com/" eudora="autourl">www.cisco.com</a></html>

--=====================_1302240==_.ALT--




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