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From: "Hussein F. Salama" <hsalama@cisco.com>
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Matt,

Please find some comments and questions on your analysis below.

Hussein


Matt Squire wrote:
> 
> >
> >2. Matt raised numerous transport issues (TCP or not TCP). I think its
> >fair to say we want to avoid designing a new transport protocol. There
> >is work underway elsewhere (sigtran and slums) to do this, which we can
> >leverage. It would seem to me that if something new is needed, this
> >would also be independent of whether the protocol baseline is BGP or
> >SCSP, since presumably one could use this new transport protocol with
> >either.
> >
> 
> I certainly agree that we don't want a new transport protocol.  But well
> short of designing a new transport is to design reliability into our
> application protocol (ie with acks, timeouts, re-transmissions, etc), just
> like most other application protocols.

I prefer to view TBGP/GLP as a call routing protocol not as an
application protocol. A call routing protocol bears closer resemblance
to layer 3 routing protocols than it has to application protocols. In
order to scale well and not to overload the network with too much
control traffic, routing protocols may have to tolerate a lower level of
reliability than some application protocols. 

> 
> >3. To me, the MAIN differences between a BGP based approach and SCSP
> >based approach are:
> >  a. how the data is synchronized between peers
> >  b. the intra-domain topology
> >
> >I think we are clear on (b), so (a) is the issue to explore. Hussein had
> >made reference to a 2-stage exchange only being needed with BGP, while
> >its three with SCSP. These are the kind of differences I think we need
> >to explore here. I'd invite both authors to comment on the differences
> >in how the data is exchanged and updated between peers. Lets stick with
> >inter-domain for the moment...
> >
> 
> Here's a brief synopsis of the data exchanges for each alternative.  In
> both cases, the exchanges are just peer-to-peer.
> 
> BGP
> ***
> 
>   BGP1) OPEN.  OPEN messages are exchanged to bring the link up.
> Basically, two peers connect and send an open message to the other.  A peer
> either acks the OPEN with a KEEP-ALIVE, or nacks the OPEN with a
> NOTIFICATION.  When your OPEN gets ack'd, the peer connection is considered
> to be established (ie have bi-directional connectivity).  BGP can also
> perform version negotiation and capabilities exchanges with the OPEN
> messages.
> 
>   BGP2) UPDATE.  Once a peer connection is established (via ack'd OPENs),
> BGP peers exchange UPDATE messages.  UPDATE messages are the messages that
> contain the routing information.  Each UPDATE message may contain a number
> of withdrawn routes and a single feasible route.  A withdrawn route is a
> route that should be removed from the peer's RIB-IN.  A withdrawn route is
> defined by a address/prefix-length.  A feasible route is defined as the
> NLRI (network layer reachability information, ie a list of
> address/prefix-length combinations) and its attributes.   A key mandatory
> attribute is the NEXT-HOP attribute which gives the next-hop for the NLRI.
> Basically, you send an UPDATE for each possible NEXT-HOP in your routing
> database.

You are talking here strictly about BGP for layer 3 routing. Even in
this case, a single UPDATE message may contain multiple NLRI
(destination prefixes) under the condition that they share the same
attributes (NEXT_HOP, AS_PATH, ...) . TBGP uses the MP_REACH_NLRI
attribute to advertise prefixes. Similar to BGP, the MP_REACH_NLRI
contains a list of NLRIs which share the same attributes.

The original BGP NEXT_HOP attribute will not be used in call routing.

The above discussion brings up the question I had for some time: Does
GLP/SCSP define a mechanism for explicitly withdrawing a route and
removing it from the call routing databases?

> 
> KEEP-ALIVE messages are periodically going in the background to ensure that
> the link is up.
> 
> BGP UPDATE and NOTIFICATION messages are not acknowledged as TCP
> acknowledgment is considered sufficient.

I'll comment on this further down.

> 
> SCSP
> ****
>   SCSP1) Hello.  Peers exchange Hello messages to ensure bi-directional
> connectivity between peers.  This is equivalent to the BGP OPEN exchanges
> but without anything like version negotiation or capabilities exchange.
> Hello messages are also periodically exchanged in the background to ensure
> the link us still up.
> 
>   SCSP2) Cache Aligment.  SCSP distinguishes between the state where two
> peers have had their databases synchronized and when they have not.  When a
> peer connection first comes up (via Hello), the peers enter the Cache
> Alignment phase where the entire databases must be exchanged.  First, there
> is a negotiation to determine who is the master and slave of the peering
> relationship.  Next, the peers exchange summary records with each other,
> where a summary record is a short-version of the record - enough to
> identify it, but doesn't have the real data.  This phase has no parallel in
> BGP.

And I don't see what advantage call routing can gain from this phase?

> 
>   SCSP3) Cache State Update.  A server maintiains a list of the records
> that it needs to get from its peer (the summary records in the previous
> phase).  A server requests these records via a Solicit message (ie I need
> the full record for these summary records).  The response to the the
> request is a Update Request message containing some/all of the records.  A
> server continues to send further Solicit messages for records that it has
> not yet received.  A server ack's that the records have been received with
> an Update Reply containing the summaries of recieved records.  The Update
> Request messages are similar to the UPDATE messages of BGP.

From reading the SCSP spec, I thought that in the Cache State Update
phase no CSA
Solicit messages are required? Just CSA Request/CSA Reply messages. If a
CSA Solicit message is required, this makes SCSP's overhead three
messages per update as compared to BGP's one message per update.

> 
> Acknowledgements in SCSP are done per-record (ie requests for a record are
> acknowledged by sending the record, record transmissions are acknowledged
> by sending a record summary in an Update Reply).
> 
> SOME DIFFERENCES
> ****************
> 
> I think Hussein's "2 vs 3" stages are identified above.  SCSP has the added
> stage of Cache Alignment where database summaries are exchanged.
> 
> BGP uses TCP, SCSP uses X (pick your transport).
> 
> BGP exchanges and message formats are certainly more efficient, SCSP more
> generic.  In the peer-to-peer case (ie inter-domain), some SCSP header
> fields are redundant (example: reciever and sender ID).
> 
> SCSP does per-record acks and requests, which is flexible, but can be a
> pain in the ass and inefficient.  BGP UPDATE/NOTIFICATION messages are
> unacknowledged and assume TCP provides reliability for the application.
> 

BGP does not provide a mechanism to positively Ack UPDATE messages.
However the NOTIFICATION message is used to provide a negative Ack in
response to errors in the contents of, or in the processing of, an
UPDATE message. This may be looked at as a more efficient mechanism for
reliability than positive Acks. Since transport reliability is already
guaranteed by TCP, error Notification usually indicates fatal problems
with the BGP FSM, and therefore is followed by closing the peer
connection. I can see situations where, in response to an error in
processing an UPDATE message, it should be sufficient to send an error
NOTIFICATION and to ignore that UPDATE, but not closing the entire
connection. But that's the way BGP works.

> SCSP Update Requests can contain many records.  BGP UPDATEs contain a
> single route (maybe multiple withdrawn routes).

I commented on this earlier.

> 
> SCSP bases record identification on record summaries (source identifier,
> cache 'key' for record), so to replace a record you keeep the record
> summary and replace the data contents.  BGP identifies records based on the
> contents (ie NLRI), so the data content is examined to determine new vs
> replacement.
> 

From discussions with people who implemented IDRP in the past, my
understanding is that
it relied on a key for record location, which proved to be "difficult to
implement with little practical gain."

-- 
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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From: ninguno <ip@excite.com>
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base de datos.


AHORA USTED PUEDE HABLAR LARGA DISTANCIA comercialmente o en forma particular 
.....GRATIS!!!!!!.... Llamando a los numeros gratuitos tipo 0-800 del mundo.

SI, GRATIS !!!! o con aranceles equivalentes a los u$s 0.10 el minuto llamando 
a USA, Australia, Inglaterra, Francia, Alemania, Israel, Suiza, etc.... a cualquier 
hora del dia si restricciones y sin importar si pertenece a cualquiera de las 
companias telefonicas de Argentina

La nueva "interface IP", de INTERNATIONAL PHONE le permiten establecer comunicaciones 
de VOZ DE ALTA CALIDAD en  comunicaciones de telefono analogico comun (con el 
IP instalado a su PC) a telefono standard en cualquier parte del mundo sin necesidad 
de intermediar PC alguna en el lugar al que llama.

Esta "interface IP" requieren una conexion a Internet y una maquina de requerimientos 
minimos (desde equipos Pentium)

Para más información visite nuestro site www.orbita.starmedia.com/via/~international-phone/ip.html 
o bien contáctese con nuestros ejecutivos de cuenta y será para nosotros un placer 
ofrecerle una demostración gratuita.

INTERNATIONAL PHONE
Depto. de ventas

ip-ar@excite.com

Tel : (011) 4 786-1857
 
Sus datos fueron obtenidos de bases publicas de diversas fuentes.
Si usted no desea volver a recibir informacion de productos novedosos por favor 
responda este mail a 
no-e-mail@excite.com y sera eliminado automaticamente de nuestra base de datos.




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From: Jose Costa Requena <jose@tct.hut.fi>
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Hi Hussein,


>You are talking here strictly about BGP for layer 3 routing. Even in
>this case, a single UPDATE message may contain multiple NLRI
>(destination prefixes) under the condition that they share the same
>attributes (NEXT_HOP, AS_PATH, ...) . TBGP uses the MP_REACH_NLRI
>attribute to advertise prefixes. Similar to BGP, the MP_REACH_NLRI
>contains a list of NLRIs which share the same attributes.
>
>The original BGP NEXT_HOP attribute will not be used in call routing.
>
>The above discussion brings up the question I had for some time: Does
>GLP/SCSP define a mechanism for explicitly withdrawing a route and
>removing it from the call routing databases?

The SCSP itself has a mechanism to notify the databases in the rest of DCSs 
forming the SG that the server has received a request from a client or User 
Agent to remove an entry from its cache or routing database.In this case, 
when the LS which contains the entry that is going to disappear and receive 
Cache State Update Solicit (CSUS) from other server, this LS will respond 
copying the CSAS from the CSUS and will set the N bit (described in the CSAS 
record specification). The N bit signifies that this CSAS Record is actually 
a Null record and this entry in the database does not exist any longer.

I wonder if is that the question that you were asking?
and I hope that I have clarified the idea.

Best regards.

Jose



Helsinki University of Technology
Laboratory of Telecommunications Technology
Jose Costa Requena
Research Scientist
Room: SG 227B
Tel: + 358 9 451 2475
email(internet): jose@tct.hut.fi
http://keskus.tct.hut.fi/tutkimus/ipana/pers/jose.shtml



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Hi Hussein,

Here is another comment that I couldn't answer in the last mail.
Regarding the Cache Alignment state you said:

>And I don't see what advantage call routing can gain from this phase?

This phase could seems that will be inefficient but the utility appears when 
one of the connections between a pair of servers is down.The rest of servers 
in the server Group are continuing their transactions and updating their 
databases. The FSM which takes care of that pair will check the connection 
after the Time out has expired(Hello Interval) and it has to restart all the 
process till it reaches again the bidirectional state. In this moment, the 
server that was isolated from the SG has to update its database. When the 
server is again reachable can be long time ago from the last update or it 
can be after a short interval, anyway in the Cache Alignment State that 
Server can ask only for the CSA records that have changed since the last 
time that it was connected. It means that in that state there will be 
required only the records that are out of date and not the rest. But if the 
failure was only during a short interval the server does not have to update 
any record and it transitions immediately to the Cache Update State and it 
means that the protocol will perform nearly only two phases from the Hello 
directly to the Cache State Update.



Another question is about your comment:


>From reading the SCSP spec, I thought that in the Cache State Update
>phase no CSA Solicit messages are required? Just CSA Request/CSA Reply 
>messages. If a CSA Solicit message is required, this makes SCSP's overhead 
>three messages per update as compared to BGP's one message per update.

In this case, the CSUS are required for specific situations, one example 
could be the moment that the entry that the server is asking for is no 
longer in the database.As I comment in the last mail in that situation the 
Server will receive the same CSAS that it sent in the CSUS but with the N 
bit on. In this case we get the benefit of not sending and receiving all the 
record, only the notification saying that it was a null entry. I guess that 
in this and other cases is useful to send only the summary and compare if it 
is  newest or not, instead of compare all the record. I guess that when the 
data is quite big is better to send and receive only the summaries.

Sorry for so long explanation but I have tried to describe my point of view, 
maybe I'm wrong and is better only use two phases in the updating and only 
two kind of messages for that, but I hope some comments about my idea.
Thanks very much.
Best regards.
Jose

Helsinki University of Technology
Laboratory of Telecommunications Technology
Jose Costa Requena
Research Scientist
Room: SG 227B
Tel: + 358 9 451 2475
email(internet): jose@tct.hut.fi
http://keskus.tct.hut.fi/tutkimus/ipana/pers/jose.shtml



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Jose Costa Requena wrote:
> 
> Hi Hussein,
> 
> Here is another comment that I couldn't answer in the last mail.
> Regarding the Cache Alignment state you said:
> 
> >And I don't see what advantage call routing can gain from this phase?
> 
> This phase could seems that will be inefficient but the utility appears when
> one of the connections between a pair of servers is down.The rest of servers
> in the server Group are continuing their transactions and updating their
> databases. The FSM which takes care of that pair will check the connection
> after the Time out has expired(Hello Interval) and it has to restart all the
> process till it reaches again the bidirectional state. In this moment, the
> server that was isolated from the SG has to update its database. When the
> server is again reachable can be long time ago from the last update or it
> can be after a short interval, anyway in the Cache Alignment State that
> Server can ask only for the CSA records that have changed since the last
> time that it was connected. It means that in that state there will be
> required only the records that are out of date and not the rest. But if the
> failure was only during a short interval the server does not have to update
> any record and it transitions immediately to the Cache Update State and it
> means that the protocol will perform nearly only two phases from the Hello
> directly to the Cache State Update.

I think in call routing, when the connection between LS A and LS B is
disconnected (due to any reason), LS A should immediately delete any
call routes it has previously learned from LS B. As a result, when the
connection between LS A and LS B is restored, both LSs will have to
exchange their entire call routing databases. 

For call routing, and layer 3 routing, purposes, I don't think it's wise
for an LS A to keep a call route X after LS B,  which advertised X to A,
disappears. The reason is: how will we A learn about changes to the
status of X while B is down? 

> 
> Another question is about your comment:
> 
> >From reading the SCSP spec, I thought that in the Cache State Update
> >phase no CSA Solicit messages are required? Just CSA Request/CSA Reply
> >messages. If a CSA Solicit message is required, this makes SCSP's overhead
> >three messages per update as compared to BGP's one message per update.
> 
> In this case, the CSUS are required for specific situations, one example
> could be the moment that the entry that the server is asking for is no
> longer in the database.As I comment in the last mail in that situation the
> Server will receive the same CSAS that it sent in the CSUS but with the N
> bit on. In this case we get the benefit of not sending and receiving all the
> record, only the notification saying that it was a null entry. I guess that
> in this and other cases is useful to send only the summary and compare if it
> is  newest or not, instead of compare all the record. I guess that when the
> data is quite big is better to send and receive only the summaries.

Please see my previous message.

Hussein


> 
> Sorry for so long explanation but I have tried to describe my point of view,
> maybe I'm wrong and is better only use two phases in the updating and only
> two kind of messages for that, but I hope some comments about my idea.
> Thanks very much.
> Best regards.
> Jose
> 
> Helsinki University of Technology
> Laboratory of Telecommunications Technology
> Jose Costa Requena
> Research Scientist
> Room: SG 227B
> Tel: + 358 9 451 2475
> email(internet): jose@tct.hut.fi
> http://keskus.tct.hut.fi/tutkimus/ipana/pers/jose.shtml

-- 
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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From owner-iptel-outgoing@lists.research.bell-labs.com  Wed May  5 20:05:37 1999
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Jose,

Thanks for the clarification.

Jose Costa Requena wrote:
> 
> Hi Hussein,
> 
> >You are talking here strictly about BGP for layer 3 routing. Even in
> >this case, a single UPDATE message may contain multiple NLRI
> >(destination prefixes) under the condition that they share the same
> >attributes (NEXT_HOP, AS_PATH, ...) . TBGP uses the MP_REACH_NLRI
> >attribute to advertise prefixes. Similar to BGP, the MP_REACH_NLRI
> >contains a list of NLRIs which share the same attributes.
> >
> >The original BGP NEXT_HOP attribute will not be used in call routing.
> >
> >The above discussion brings up the question I had for some time: Does
> >GLP/SCSP define a mechanism for explicitly withdrawing a route and
> >removing it from the call routing databases?
> 
> The SCSP itself has a mechanism to notify the databases in the rest of DCSs
> forming the SG that the server has received a request from a client or User
> Agent to remove an entry from its cache or routing database.In this case,
> when the LS which contains the entry that is going to disappear and receive
> Cache State Update Solicit (CSUS) from other server, this LS will respond
> copying the CSAS from the CSUS and will set the N bit (described in the CSAS
> record specification). The N bit signifies that this CSAS Record is actually
> a Null record and this entry in the database does not exist any longer.

Assume server A advertises a record X to server B. Then, at some later
point in time, record X is deleted. Server B won't learn that record X
has seized to exist until it solicits an update from server A for that
record? This could be a very slow operation. How frequently should B
solicit updates from A, in order to minimize the probability of using
outdated or deleted records?

Advertising route withdrawals to a server which is multiple hops away
will be very slow using this mechanism. Let's say every server
periodically solicits updates every time interval t. It will take a
server, C, n*t time units, in the worst case, to learn that a record has
been withdrawn, where n is the number of hops between server C and the
server which has withdrawn the record. 

What's wrong with conventional routing logic: when a server detects a
change to the state of a record, it immediately triggers an unsolicited
update to its peers. When a peer receives the update, if it causes any
changes to that peer's records, it immediately triggers an update to the
peer's peers. And so on. Wouldn't that be faster?

The SCSP RFC says: "The "N" bit signifies that this CSAS Record is
actually a Null record.  This bit is only used in a CSAS Record
contained in a CSU Request/Reply which is sent in response to a CSUS
message." This means that an update advertising a record deletion must
be solicited. How about other changes to the attributes of an existing
record? Can they trigger an unsolicited update, CSU Request?

Hussein

> 
> I wonder if is that the question that you were asking?
> and I hope that I have clarified the idea.
> 
> Best regards.
> 
> Jose
> 
> Helsinki University of Technology
> Laboratory of Telecommunications Technology
> Jose Costa Requena
> Research Scientist
> Room: SG 227B
> Tel: + 358 9 451 2475
> email(internet): jose@tct.hut.fi
> http://keskus.tct.hut.fi/tutkimus/ipana/pers/jose.shtml
> 
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.

-- 
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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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May  6 03:28:06 1999
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Hi Hussein,

Thanks for your comments.
I have implemented the SCSP and actually I am testing the first version in 
the Lab., I am glad to receive some comments to change some details and 
polish the implementation.

>Assume server A advertises a record X to server B. Then, at some later
>point in time, record X is deleted. Server B won't learn that record X
>has seized to exist until it solicits an update from server A for that
>record? This could be a very slow operation. How frequently should B
>solicit updates from A, in order to minimize the probability of using
>outdated or deleted records?

You are right, it is a good point because in the case that we fixe a short 
time out in the cache entry, all the database will be updated in a short 
interval but the drawback is that it will increase the traffic.Otherwise, if 
we fixed a long time out it will take a long time to update all the records.

Professor Kantola and me are working in the feature provided by the Family 
Identifier to split the database up in different sections. The part of the 
routing table that is changing dynamically will be fixed with a shorter time 
out than the other.Thus, the former part will be updated more frequently. 
With the Family ID we can distinguish two instances of the same SCSP running 
in the same Sever Group and maybe that is the key to differentiate a section 
in the routing table that is more dynamic than another section that is more 
static and requires less maintenance.


>Advertising route withdrawals to a server which is multiple hops away
>will be very slow using this mechanism. Let's say every server
>periodically solicits updates every time interval t. It will take a
>server, C, n*t time units, in the worst case, to learn that a record has
>been withdrawn, where n is the number of hops between server C and the
>server which has withdrawn the record. 

>What's wrong with conventional routing logic: when a server detects a
>change to the state of a record, it immediately triggers an unsolicited
>update to its peers. When a peer receives the update, if it causes any
>changes to that peer's records, it immediately triggers an update to the
>peer's peers. And so on. Wouldn't that be faster?


>The SCSP RFC says: "The "N" bit signifies that this CSAS Record is
>actually a Null record.  This bit is only used in a CSAS Record
>contained in a CSU Request/Reply which is sent in response to a CSUS
>message." This means that an update advertising a record deletion must
>be solicited. How about other changes to the attributes of an existing
>record? Can they trigger an unsolicited update, CSU Request?

>Hussein

I guess that the conventional routing logic that you mention in the last 
paragraph is performed by the SCSP on its last phase. When the peer has 
reached the Cache State Update phase they will perform a mirroring state.It 
means that when any cache entry has timed out, or it has been modified, this 
LS will send a CSU Request to each of its DCSs. In this Request we can 
advertise either that this record is newest or is going to be deleted 
setting the bit N. Thus, we don't have to wait to Solicit and update message 
for that record because is just the LS that has the record which is going to 
be removed who will send the Request to all the DCSs advertising that this 
record is going to be deleted from the routing table using the N bit.

Thanks for your comments, I am looking forward to hear from you about my 
ideas and what is your opinion.



Helsinki University of Technology
Laboratory of Telecommunications Technology
Jose Costa Requena
Research Scientist
Room: SG 227B
Tel: + 358 9 451 2475
email(internet): jose@tct.hut.fi
http://keskus.tct.hut.fi/tutkimus/ipana/pers/jose.shtml



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---
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Hussein - 

I included answers/comments interspersed with the original message.  I
tried to delete anything not relevant to your comments.

- Matt


>In
>order to scale well and not to overload the network with too much
>control traffic, routing protocols may have to tolerate a lower level of
>reliability than some application protocols. 
>

I'm not sure what you mean by a "lower level of reliability".  If the
routing information is not transmitted reliably, then routing loops can
occur.  All peers in a BGP administrative domain must have the same view of
the network.  Otherwise they can make independent decisions that cause
loops.  Reliability is a requirement, the question is where reliability
comes from.  


>>  Each UPDATE message may contain a number
>> of withdrawn routes and a single feasible route.  A withdrawn route is a
>> route that should be removed from the peer's RIB-IN.  A withdrawn route is
>> defined by a address/prefix-length.  A feasible route is defined as the
>> NLRI (network layer reachability information, ie a list of
>> address/prefix-length combinations) and its attributes.   A key mandatory
>> attribute is the NEXT-HOP attribute which gives the next-hop for the NLRI.
>> Basically, you send an UPDATE for each possible NEXT-HOP in your routing
>> database.
>
>You are talking here strictly about BGP for layer 3 routing. Even in
>this case, a single UPDATE message may contain multiple NLRI
>(destination prefixes) under the condition that they share the same
>attributes (NEXT_HOP, AS_PATH, ...) . 

I think I said that.  


> TBGP uses the MP_REACH_NLRI
>attribute to advertise prefixes. Similar to BGP, the MP_REACH_NLRI
>contains a list of NLRIs which share the same attributes.
>
>The original BGP NEXT_HOP attribute will not be used in call routing.

Right, but the restriction is the same, one next-hop per message.  There is
a list of addr prefixes with each next-hop.  


>
>>From reading the SCSP spec, I thought that in the Cache State Update
>phase no CSA
>Solicit messages are required? Just CSA Request/CSA Reply messages. If a
>CSA Solicit message is required, this makes SCSP's overhead three
>messages per update as compared to BGP's one message per update.
>

The solicit messages are used in recovery and link activation.  When a link
comes up between two peers, the peers exchange summaries of all of their
data.  If they need the full record then they ask for it via a solicit.
Suppose you have the servers A-B-C all synchronized, with A-B being peers,
and B-C being peers.  Then a link is brought up between A and C.  The
servers exchange summaries, realize that they already have all of the
records that the other guy has, and are synchronized without ever
synchronizing the real data.  

When restricted to two peers, the solicit adds an extra message during the
initial synchronization.  Solicits are not used after things are up.  

In steady state (after the peers are synchronized), the local server sends
requests to the peer containing the data, and the peer acknowledges via a
reply.  So the general sequence is information followed by ack.  Data is
not solicited.  


>
>BGP does not provide a mechanism to positively Ack UPDATE messages.
>However the NOTIFICATION message is used to provide a negative Ack in
>response to errors in the contents of, or in the processing of, an
>UPDATE message. This may be looked at as a more efficient mechanism for
>reliability than positive Acks. Since transport reliability is already
>guaranteed by TCP, error Notification usually indicates fatal problems
>with the BGP FSM, and therefore is followed by closing the peer
>connection. I can see situations where, in response to an error in
>processing an UPDATE message, it should be sufficient to send an error
>NOTIFICATION and to ignore that UPDATE, but not closing the entire
>connection. But that's the way BGP works.
>

And if the route process is busy or its queues are full or for some reason
it just can't take the socket upcall from the TCP layer at this time, then
the information is lost.  And that is a problem.  Hence one has to hack the
TCP stack and the OS to ensure this never happens.  


>
>>From discussions with people who implemented IDRP in the past, my
>understanding is that
>it relied on a key for record location, which proved to be "difficult to
>implement with little practical gain."
>

When the "uniqueness" information in a record is an 4-byte IP address, then
a key is useless.  When the uniqueness information is more (ie an IPv6 addr
or an E164 addr), then there is benefit from using a short key.  What is so
difficult to implement about a key?


- Matt

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>
>Assume server A advertises a record X to server B. Then, at some later
>point in time, record X is deleted. Server B won't learn that record X
>has seized to exist until it solicits an update from server A for that
>record? This could be a very slow operation. How frequently should B
>solicit updates from A, in order to minimize the probability of using
>outdated or deleted records?
>

If X is deleted at A, then A sends an request to B indicating the deletion.
 B acknowledges the deletion.  The message from A is unsolicited.  

Information is solicited only during link activation and initial
synchronization.  

- Matt

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>I think in call routing, when the connection between LS A and LS B is
>disconnected (due to any reason), LS A should immediately delete any
>call routes it has previously learned from LS B. As a result, when the
>connection between LS A and LS B is restored, both LSs will have to
>exchange their entire call routing databases. 
>

If the next hop on a call route received from B is B, then the route should
be deleted when B goes away.  But in when the next hop is not B, we might
consider having LS A keep the route around for "a little while."  Haven't
really given this enough thought yet, but it might help with transient
failures (ie B is just being rebooted and will be up again in two minutes).  

Just a thought...

- Matt


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Date: Wed, 12 May 1999 12:14:38 -0700 (PDT)
From: Tom Gray <tom_gray_intp@yahoo.com>
Subject: CPL Features
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Has anyone considered what would be the practical  maximum feature set
that could  be programmed by a set of CPL scripts? This would be a set
of features that could be programmed to interact in a deterministic
manner and whose operation would not be a source of confusion to the
user. This bears directly on the customer value of the application.

Has any thought gone into asking the calling party about his
preferences about the disposition of a call by CPL. For example the
calling party could be asked if he wishes to camp on to the called
party and wait for him to become free or to forward to voice mail.
These options could be set up as both acceptable by the terminating
party.





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From owner-iptel-outgoing@lists.research.bell-labs.com  Wed May 12 23:29:41 1999
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Date: Wed, 12 May 1999 23:23:27 -0400
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Tom Gray wrote:
> 
> Has any thought gone into asking the calling party about his
> preferences about the disposition of a call by CPL. For example the
> calling party could be asked if he wishes to camp on to the called
> party and wait for him to become free or to forward to voice mail.
> These options could be set up as both acceptable by the terminating
> party.

Yes. See:

http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sip-caller-00.txt

This draft proposes defining caller preferences to be included in SIP.
The draft does not go so far as defining how the CPL might be structured
to explicitly take these into account, though.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen

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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May 13 12:58:47 1999
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Date: Thu, 13 May 1999 08:53:57 -0700 (PDT)
From: Tom Gray <tom_gray_intp@yahoo.com>
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--- Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com> wrote:
> Tom Gray wrote:
> > 
> > Has any thought gone into asking the calling party
> about his
> > preferences about the disposition of a call by
> CPL. For example the
> > calling party could be asked if he wishes to camp
> on to the called
> > party and wait for him to become free or to
> forward to voice mail.
> > These options could be set up as both acceptable
> by the terminating
> > party.
> 
> Yes. See:
> 
>
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sip-caller-00.txt
> 
> This draft proposes defining caller preferences to
> be included in SIP.
> The draft does not go so far as defining how the CPL
> might be structured
> to explicitly take these into account, though.
> 

The draft is very interesting in its possibilities. In my
interpretation of it, the caller is able state his preferences for call
handling at the called party end of the call. Are there mechanisms for
this to be extended so that the user can  indicate his preferences on a
call by call basis taking into account iformation from the called
party. For example, if the called party could indicate that he could
accept the call after a brief delay the caller may wish to camp on (ie
queue) rather than go to voice mail or he may indicate a preference for
an automatic call back when free. This would best be done during call
set up rather than requiring a user to identify all possibiities before
the call is attempted.

_________________________________________________________
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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May 13 16:18:48 1999
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From: Jonathan Lennox <lennox@cs.columbia.edu>
To: Tom Gray <tom_gray_intp@yahoo.com>
Cc: iptel@lists.research.bell-labs.com
Subject: Re: CPL Features
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On Wed, May 12 1999, "Tom Gray" wrote to "iptel@lists.research.bell-labs.com" saying:

> Has anyone considered what would be the practical  maximum feature set
> that could  be programmed by a set of CPL scripts? This would be a set
> of features that could be programmed to interact in a deterministic
> manner and whose operation would not be a source of confusion to the
> user. This bears directly on the customer value of the application.

Since the point of the CPL is to allow features to be assembled out of CPL
scripts, I don't think that a "maximum feature set" is really possible; the
whole point of the CPL is that it's extensible.  The feature interaction
problem is discussed somewhat in the CPL requirements I-D.  (This has
expired from the I-D archive, by now, but it's available off my web page --
<http://www.cs.columbia.edu/~lennox/>).  I'm planning on issuing an updated
version of this document soon, so any comments on it are appreciated.
 
> Has any thought gone into asking the calling party about his
> preferences about the disposition of a call by CPL. For example the
> calling party could be asked if he wishes to camp on to the called
> party and wait for him to become free or to forward to voice mail.
> These options could be set up as both acceptable by the terminating
> party.

Jonathan Rosenberg mentioned his caller-preferences draft.  There are two
places where this interacts with the CPL -- potentially honoring caller
preferences in a called-party-side CPL script, and adding caller preferences
in a caller-side CPL script.  Both of these are issues I want to look at in
a future revision of the CPL.

-- 
Jonathan Lennox
lennox@cs.columbia.edu

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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May 13 17:11:11 1999
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From: "Puthenkulam, Jose P" <jose.p.puthenkulam@intel.com>
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Subject: XML parser 
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Does anybody know any web sites where public domain XML parser source code 
is available.

thanks,
Jose Puthenkulam
Senior Software Engineer,
Broadband and IP Telephony Labs,
Intel Architecture Labs.
Hillsboro, OR 97124


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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May 13 17:24:03 1999
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From: Jonathan Lennox <lennox@cs.columbia.edu>
To: "Puthenkulam, Jose P" <jose.p.puthenkulam@intel.com>
Cc: iptel@lists.research.bell-labs.com
Subject: Re: XML parser 
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On Thu, May 13 1999, "Jose P. Puthenkulam" wrote to "iptel@lists.research.bell-labs.com" saying:

> Does anybody know any web sites where public domain XML parser source code 
> is available.

There are a number of freely-available parsers listed at
<http://www.xml.com/xml/pub/Guide/XML_Parsers>.  If you're programming in C,
expat is certainly the one to use.  In Java, I used IBM's xml4j to develop
the CPL DTD and validate the sample scripts.  (I assume the CPL is why
you're asking about XML on the IPtel list?)

-- 
Jonathan Lennox
lennox@cs.columbia.edu

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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May 13 17:34:40 1999
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From: "Hoefelmeyer, Ralph" <Ralph.Hoefelmeyer@wcom.com>
Subject: RE: XML parser
To: "'Puthenkulam, Jose P'" <jose.p.puthenkulam@intel.com>,
        iptel@lists.research.bell-labs.com
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go look at 
http://tecfa2.unige.ch/guides/xml/pointers.html
maybe something there

Ralph

> -----Original Message-----
> From:	Puthenkulam, Jose P [SMTP:jose.p.puthenkulam@intel.com]
> Sent:	Thursday, May 13, 1999 3:04 PM
> To:	iptel@lists.research.bell-labs.com
> Subject:	XML parser
> 
> Does anybody know any web sites where public domain XML parser source code
> 
> is available.
> 
> thanks,
> Jose Puthenkulam
> Senior Software Engineer,
> Broadband and IP Telephony Labs,
> Intel Architecture Labs.
> Hillsboro, OR 97124
> 
> 
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
5.5.2569.0">
<TITLE>RE: XML parser</TITLE>
</HEAD>
<BODY>

<P><FONT COLOR=3D"#0000FF" FACE=3D"Arial">go look at </FONT>
<BR><U><FONT COLOR=3D"#0000FF" FACE=3D"Arial"><A =
HREF=3D"http://tecfa2.unige.ch/guides/xml/pointers.html" =
TARGET=3D"_blank">http://tecfa2.unige.ch/guides/xml/pointers.html</A></F=
ONT></U>
<BR><FONT COLOR=3D"#0000FF" FACE=3D"Arial">maybe something there</FONT>
</P>

<P><FONT COLOR=3D"#0000FF" FACE=3D"Arial">Ralph</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">Puthenkulam, Jose P =
[SMTP:jose.p.puthenkulam@intel.com]</FONT>
<BR><B><FONT SIZE=3D1 FACE=3D"Arial">Sent:&nbsp;&nbsp;</FONT></B> <FONT =
SIZE=3D1 FACE=3D"Arial">Thursday, May 13, 1999 3:04 PM</FONT>
<BR><B><FONT SIZE=3D1 =
FACE=3D"Arial">To:&nbsp;&nbsp;&nbsp;&nbsp;</FONT></B> <FONT SIZE=3D1 =
FACE=3D"Arial">iptel@lists.research.bell-labs.com</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">XML parser</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">Does anybody know any web sites =
where public domain XML parser source code </FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">is available.</FONT>
</P>

<P><FONT SIZE=3D2 FACE=3D"Courier New">thanks,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Jose Puthenkulam</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Senior Software =
Engineer,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Broadband and IP Telephony =
Labs,</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Intel Architecture Labs.</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">Hillsboro, OR 97124</FONT>
</P>
<BR>

<P><FONT SIZE=3D2 FACE=3D"Courier New">---------</FONT>
<BR><FONT SIZE=3D2 FACE=3D"Courier New">This message came from the IETF =
IPTEL Working Group Mailing List.</FONT>
</P>
</UL>
</BODY>
</HTML>
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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May 13 17:58:06 1999
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Date: Thu, 13 May 1999 14:51:55 -0700 (PDT)
From: Tom Gray <tom_gray_intp@yahoo.com>
Subject: Re: CPL Features
To: Jonathan Lennox <lennox@cs.columbia.edu>
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--- Jonathan Lennox <lennox@cs.columbia.edu> wrote:
> On Wed, May 12 1999, "Tom Gray" wrote to
> "iptel@lists.research.bell-labs.com" saying:
> 
> > Has anyone considered what would be the practical 
> maximum feature set

...

> > Since the point of the CPL is to allow features to
> be assembled out of CPL
> scripts, I don't think that a "maximum feature set"
> is really possible; the
> whole point of the CPL is that it's extensible.  The
> feature interaction
> problem is discussed somewhat in the CPL
> requirements I-D.  (This has
> expired from the I-D archive, by now, but it's
> available off my web page --
> <http://www.cs.columbia.edu/~lennox/>).  I'm
> planning on issuing an updated
> version of this document soon, so any comments on it
> are appreciated.
>  

The issue as I see it is the practical maximum extensibility of the
system. This would be the number and type of features that could be
made to work in a determinsitic manner. This is the feature
ineteraction problem and any call control system should be assessed
against its ability to handle this problem. This is especially true in
a system in which it is intended that the end user will be able to add
features at will. I suppose what I am saying is that the feature set
which will be enabled or made possible by CPL  should be identified. It
is not clear to me that some useful features of current systems are
possible with the SIP/CPL combination. This may be only the refelction
of  my lack of familiarity with this system. I have read the
specifications for CPL but the connection to SIP and the rationale of
how features are created do not seem to be spelled out clearly in these
documents.


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From owner-iptel-outgoing@lists.research.bell-labs.com  Thu May 13 21:47:48 1999
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Tom Gray wrote:
> 
> The issue as I see it is the practical maximum extensibility of the
> system. This would be the number and type of features that could be
> made to work in a determinsitic manner. This is the feature
> ineteraction problem and any call control system should be assessed
> against its ability to handle this problem. This is especially true in
> a system in which it is intended that the end user will be able to add
> features at will. I suppose what I am saying is that the feature set
> which will be enabled or made possible by CPL  should be identified. It
> is not clear to me that some useful features of current systems are
> possible with the SIP/CPL combination. 

CPL definitely does not handle certain features - call waiting, for
example. In particular, features which require user input after the call
is established, or features which require call state, are not possible.
In an IP telephony context, many, if not all, of these features are most
easily supported in end systems. CPL is best at providing call routing
related services; the infinite flavors of forwarding, follow-me,
screening, and so on. These are services which require network support,
often since the user may not be logged in or otherwise available. I
don't think we can enumerate them all. If we could, may as well number
them, do away with CPL, and have users select a service by number. But,
after all, CPL is a scripting language of sorts, and the point is we can
use it to develop new services (at least those supportable within the
primitives CPL provides. Its not Turing complete, so everything is not
possible).

This more restricted scope helps mitigate many feature interaction
problems. There are some we know of (the call forward/call screening
interaction), and these are documented in the framework. If you know of
any other interactions you see, please post them.

It would also be helpful if you could point out particular features you
feel are useful yet not supported by CPL.

-Jonathan R.

-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen

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--- Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com> wrote:

> 
> CPL definitely does not handle certain features -
> call waiting, for
> example. In particular, features which require user
> input after the call
> is established, or features which require call
> state, are not possible.
> In an IP telephony context, many, if not all, of
> these features are most
> easily supported in end systems. CPL is best at
> providing call routing
> related services; the infinite flavors of
> forwarding, follow-me,
> screening, and so on. These are services which
> require network support,
> often since the user may not be logged in or
> otherwise available. I
> don't think we can enumerate them all. If we could,
> may as well number
> them, do away with CPL, and have users select a
> service by number. But,
> after all, CPL is a scripting language of sorts, and
> the point is we can
> use it to develop new services (at least those
> supportable within the
> primitives CPL provides. Its not Turing complete, so
> everything is not
> possible).
> 
> This more restricted scope helps mitigate many
> feature interaction
> problems. There are some we know of (the call
> forward/call screening
> interaction), and these are documented in the
> framework. If you know of
> any other interactions you see, please post them.
> 
> It would also be helpful if you could point out
> particular features you
> feel are useful yet not supported by CPL.
> 

The CPL documents describe features like Call Forward Busy, Call
Forward No Answer and the refusal of anonymous calls. These certainly
are features that have meaning only if the user is present.
Additionally a useful feature to have is the ability to set an
automatic call back on a user who is not registered or present and
another is to camp on to a busy line (ie queue till the user becomes
free.  These are features that would interact with the call forwarding
features. The resolution of these interactions is  something that a
user would like to decide upon during a call with the knowledge that he
has at the time. It is not something that he could set up in a
preference table.

_________________________________________________________
Do You Yahoo!?
Free instant messaging and more at http://messenger.yahoo.com

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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri May 14 15:41:15 1999
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Matt Squire wrote:

......

> >
> >BGP does not provide a mechanism to positively Ack UPDATE messages.
> >However the NOTIFICATION message is used to provide a negative Ack in
> >response to errors in the contents of, or in the processing of, an
> >UPDATE message. This may be looked at as a more efficient mechanism for
> >reliability than positive Acks. Since transport reliability is already
> >guaranteed by TCP, error Notification usually indicates fatal problems
> >with the BGP FSM, and therefore is followed by closing the peer
> >connection. I can see situations where, in response to an error in
> >processing an UPDATE message, it should be sufficient to send an error
> >NOTIFICATION and to ignore that UPDATE, but not closing the entire
> >connection. But that's the way BGP works.
> >
> 
> And if the route process is busy or its queues are full or for some reason
> it just can't take the socket upcall from the TCP layer at this time, then
> the information is lost.  And that is a problem.  Hence one has to hack the
> TCP stack and the OS to ensure this never happens.
> 

I'd appreciate pointers to specific implementations where the OS and the
TCP stack has been hacked. I am not aware of any. Please also check the
the NSFNET backbone (phase II) for BGP implementations on standard OSes
and TCP stacks. And what about the Gated implementation? 

> >
> >>From discussions with people who implemented IDRP in the past, my
> >understanding is that
> >it relied on a key for record location, which proved to be "difficult to
> >implement with little practical gain."
> >
> 
> When the "uniqueness" information in a record is an 4-byte IP address, then
> a key is useless.  When the uniqueness information is more (ie an IPv6 addr
> or an E164 addr), then there is benefit from using a short key.  What is so
> difficult to implement about a key?
> 

A cache key's benefits will be quite limited, I think. True, it may
simplify the search for an older version of the same record received in
an Update message. However, a routing/call routing protocol does not
just search for older versions of the same record. It also searches for
other existing records having the same destination prefix, in order to
do route selection amongst them. So a prefix-based search is needed
anyways, because this can't be done using the cache key. 

Hussein 


> - Matt
> 
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.

-- 
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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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri May 14 16:13:25 1999
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Matt Squire wrote:
> 
> >I think in call routing, when the connection between LS A and LS B is
> >disconnected (due to any reason), LS A should immediately delete any
> >call routes it has previously learned from LS B. As a result, when the
> >connection between LS A and LS B is restored, both LSs will have to
> >exchange their entire call routing databases.
> >
> 
> If the next hop on a call route received from B is B, then the route should
> be deleted when B goes away.  But in when the next hop is not B, we might
> consider having LS A keep the route around for "a little while."  Haven't
> really given this enough thought yet, but it might help with transient
> failures (ie B is just being rebooted and will be up again in two minutes).

What if B advertises the addresses of the gateways in its domain as the
next hops, not its own address. When B goes away, there will be no way
to advertise to other domains updated information about these gateways
and the prefixes they can reach (assuming B is the only LS in that
domain).

This is an issue where we can present good examples for both approaches.
I think deleting all call routes learned from B immediately when B is
disconnected is the safer approach. We will probably re-learn the routes
which are still valied through alternate paths.

Hussein


> 
> Just a thought...
> 
> - Matt
> 
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.

-- 
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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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri May 14 16:45:30 1999
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Matt Squire wrote:
> 
> >
> >Assume server A advertises a record X to server B. Then, at some later
> >point in time, record X is deleted. Server B won't learn that record X
> >has seized to exist until it solicits an update from server A for that
> >record? This could be a very slow operation. How frequently should B
> >solicit updates from A, in order to minimize the probability of using
> >outdated or deleted records?
> >
> 
> If X is deleted at A, then A sends an request to B indicating the deletion.
>  B acknowledges the deletion.  The message from A is unsolicited.
> 
> Information is solicited only during link activation and initial
> synchronization.

Since I am no expert on SCSP, I'll accept your statement. However, the
description provided in Appendix B.2.0.2 of the SCSP RFC  about the
usage of the "N" bit in the CSAS record is quite confusing. The "N" bit
is the way to indicate a record deletion.

Appendix B.2.0.2 of the SCSP RFC says:
     "The "N" bit signifies that this CSAS Record is actually a Null
     record.  This bit is only used in a CSAS Record contained in a CSU
     Request/Reply which is sent in response to a CSUS message. ....."

There was no mention of phases. So I assumed the above statement applies
to both phases (cache alignment and cache state update), not just to the
cache alignment phase.

Hussein


> 
> - Matt
> 
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.

-- 
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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From owner-iptel-outgoing@lists.research.bell-labs.com  Sat May 15 08:45:43 1999
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To: "Puthenkulam, Jose P" <jose.p.puthenkulam@intel.com>
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Subject: RE: XML parser 
Date: Sat, 15 May 1999 08:37:39 -0400
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That's probably the best one.

http://www.alphaworks.ibm.com/tech/xml4c

-=Francois=-

> -----Original Message-----
> From: owner-iptel@lists.research.bell-labs.com
> [mailto:owner-iptel@lists.research.bell-labs.com]On Behalf Of
> Puthenkulam, Jose P
> Sent: May 13, 1999 5:04 PM
> To: iptel@lists.research.bell-labs.com
> Subject: XML parser 
> 
> 
> Does anybody know any web sites where public domain XML parser 
> source code 
> is available.
> 
> thanks,
> Jose Puthenkulam
> Senior Software Engineer,
> Broadband and IP Telephony Labs,
> Intel Architecture Labs.
> Hillsboro, OR 97124
> 
> 
> ---------
> This message came from the IETF IPTEL Working Group Mailing List.
> 

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From owner-iptel-outgoing@lists.research.bell-labs.com  Sat May 15 11:04:08 1999
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Hello,
 
> Matt Squire wrote:
> > 
> > >I think in call routing, when the connection between LS A and LS B is
> > >disconnected (due to any reason), LS A should immediately delete any
> > >call routes it has previously learned from LS B. As a result, when
>  the
> > >connection between LS A and LS B is restored, both LSs will have to
> > >exchange their entire call routing databases.
> > >
> > 
> > If the next hop on a call route received from B is B, then the route
>  should
> > be deleted when B goes away.  But in when the next hop is not B, we
>  might
> > consider having LS A keep the route around for "a little while." 
>  Haven't
> > really given this enough thought yet, but it might help with transient
> > failures (ie B is just being rebooted and will be up again in two
>  minutes).
> 
> What if B advertises the addresses of the gateways in its domain as the
> next hops, not its own address. When B goes away, there will be no way
> to advertise to other domains updated information about these gateways
> and the prefixes they can reach (assuming B is the only LS in that
> domain).
> 
> This is an issue where we can present good examples for both approaches.
> I think deleting all call routes learned from B immediately when B is
> disconnected is the safer approach. We will probably re-learn the routes
> which are still valied through alternate paths.
> 
> Hussein
> 
There are something that I cannot understand:

If the LS B is the unique server in that domain which can advertise about 
the gateways in its domain as the next hops, then if it goes down there will 
be no other way to advertise about the gateways of that domain. If I'm right 
in that supposition we cannot re-learn the routes, that are still valid, but 
they cannot be advertised through the B LS because actually it is down.

I think that it would be wiser if the last updates from the B LS are 
maintained until they will be updated from other paths, because as Matt 
wrote it could be either a transient failure of the B LS or maybe the 
connection to this domain through the B server is going to be replaced with 
another path or LS which will advertise about the same gateways that was 
done by the B before.

Best Regards.
Jose





*******************************************************
Jose Costa Requena
Research Scientist
Helsinki University of Technology
Laboratory of Telecommunications Technology
Room: SG 227B, Tel: + 358 9 451 2475
email(internet): jose@tct.hut.fi
http://keskus.tct.hut.fi/tutkimus/ipana/pers/jose.shtml



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From owner-iptel-outgoing@lists.research.bell-labs.com  Tue May 18 00:56:47 1999
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Date: Mon, 17 May 1999 23:53:41 -0400
From: Hans Viens <vieh01@gel.usherb.ca>
Subject: SIP encryption
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Hi Folk!

I have few questions concerning SIP implementation.

1. Is SIP "HAS" to be implement PGP-based to be compatible with other SIP 
engine ?

2. If SIP has to be PGP-based, am I right to assume the Public-key 
encryption has been made using RSA algorithm ?

3. Is SIP use symmetric encryption PGP's IDEA algorithm ?

It's clear in the RFC that the signature could be make by MD5 or SHA.1 
algorithm, but I must admit that I feel confuse about the rest!  Somebody 
could help me ?

Hans...

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Date: Tue, 18 May 1999 01:06:52 -0400
From: Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com>
Organization: Bell Laboratories
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This post belongs on confctrl, not iptel.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
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Date: Mon, 17 May 1999 23:53:41 -0400
From: Hans Viens <vieh01@gel.usherb.ca>
Subject: SIP encryption
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Hi Folk!

I have few questions concerning SIP implementation.

1. Is SIP "HAS" to be implement PGP-based to be compatible with other SIP 
engine ?

2. If SIP has to be PGP-based, am I right to assume the Public-key 
encryption has been made using RSA algorithm ?

3. Is SIP use symmetric encryption PGP's IDEA algorithm ?

It's clear in the RFC that the signature could be make by MD5 or SHA.1 
algorithm, but I must admit that I feel confuse about the rest!  Somebody 
could help me ?

Hans...

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From owner-iptel-outgoing@lists.research.bell-labs.com  Tue May 18 09:14:24 1999
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>> 
>> And if the route process is busy or its queues are full or for some reason
>> it just can't take the socket upcall from the TCP layer at this time, then
>> the information is lost.  And that is a problem.  Hence one has to hack the
>> TCP stack and the OS to ensure this never happens.
>> 
>
>I'd appreciate pointers to specific implementations where the OS and the
>TCP stack has been hacked. I am not aware of any. Please also check the
>the NSFNET backbone (phase II) for BGP implementations on standard OSes
>and TCP stacks. And what about the Gated implementation? 
>

I'm aware of these implementations.  In routers, the routing process is the
highest priority process in the box, because you can't forward the packets
without the control.  But in call routing, GLP may not be the most
important thing in the box.  Call routing is a different problem.

Many devices are going to handle media, call routing (SIP/323), GLP, and
more.  In this case, existing media connections have to have the highest
priority.  When resources are tight, existing media streams have to get
whats available so that already accepted calls get the quality they need.
Other things going on in the device may also have precedence over GLP.  

When a protocol is not the most important thing going on in a device,
things can happen inside the device so that TCP acks something but the
information doesn't get processed by its application.  Queues fill up, you
run out of memory, etc.  It happens all of the time.  

Reliability (timeouts, retries, etc.) is just a good thing to build into a
protocol, regardless of its transport.  

- Matt 


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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri May 21 13:34:17 1999
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From: <dianita_@mixmail.com>
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Date: Vie, 21 May 1999 10:49:45
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Hola,

Hay nuevas fotos en mi pagina!!!

Puedes visitarla en:

http://209.3.241.176/

Espero que te gusten!!

Diana
 
 
 
 
 
 
 
 
 
 

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From owner-iptel-outgoing@lists.research.bell-labs.com  Sat May 22 02:18:32 1999
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From: CAMAS REBATIBLES <gonzalom@ciudad.com.ar>
To: Solucionar problemas de espacio <gonzalom@ciudad.com.ar>
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	Desde hace mas de veinte años estamos en la industria de las CAMAS REBATIBLES 
y sus muebles asociados. Conocemos el valor del espacio.
	Usted sabe, el mayor obstaculo para usar el ambiente para cualquier otra cosa, 
como una oficina o para hacer ejercicios o para que sus hijos jueguen o para 
recibir amigos o simplemente para sentirse comodo, es la CAMA. No importa si 
es de una o dos plazas, cualquier CAMA domina el ambiente y limita la funcionalidad 
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	Podra obtener mayor informacion en nuestro sitio en INTERNET  www.camasrebatibles.com.ar
	Si este mensaje no es de su interes, disculpe la molestia.
	Si nos envia un e-mail a info@camasrebatibles.com.ar  colocando en subject(asunto):borrar, 
su direccion sera borrada de nuestra base de datos.
	atte, CAMAS REBATIBLES 

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From owner-iptel-outgoing@lists.research.bell-labs.com  Wed May 26 05:28:26 1999
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To: 
From: <tecnosi@datamarkets.com.ar>
Date: Wed, 26 May 1999 06:20:59
Subject: CURSOS DE INGLES

INVIERTA EN CAPACITACION O PAGUE EL COSTO DE LA IGNORANCIA
SI NO AHORA, ¿CUANDO?

¿Por que elegirnos para la Capacitación en Inglés? 

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3. RESULTADOS: Realizamos Evaluaciones de Nivel antes de comenzar el curso, para determinar la distribución de los alumnos en grupos homogéneos conforme su nivel de idioma.
Realizamos Evaluaciones y un Informe Personal Trimestral, lo cual le permite medir los Resultados de la Capacitación.

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Arancel por hora y por grupo:	$25	$30	$35	$40		
Los precios no incluyen IVA.
					
ESPERAMOS SU LLAMADO AL 15-4975-0797
ENGLISH SKILLS

Si este mensaje no es de su interés, disculpe la molestia.


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From owner-iptel-outgoing@lists.research.bell-labs.com  Wed May 26 06:02:53 1999
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Date: Wed, 26 May 1999 15:28:13 +0530
From: Venkata Narasimha Murthy Nanduri <nvnmurthy@seeyes.co.in>
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Hai,

1) How can we know whether an RTP packet is encapsulated in the UDP data
field?
2) If it is based on the UDP port numbers, then are there any standard
UDP port
numbers that will be used for RTP?

Thanks in Advance
Murthy NVN


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From owner-iptel-outgoing@lists.research.bell-labs.com  Wed May 26 06:25:00 1999
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Date: Wed, 26 May 1999 11:17:10 +0100
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--> Venkata Narasimha Murthy Nanduri writes:
>1) How can we know whether an RTP packet is encapsulated in the UDP data
>field?

The RTP spec contains code to validate a packet, to ensure that it is RTP
data.

>2) If it is based on the UDP port numbers, then are there any standard
>UDP port numbers that will be used for RTP?

The ports used for RTP traffic are signalled out of band using your
favourite control protocol (SIP, SAP, RTSP, H.whatever, etc...). There
are no standard UDP ports for RTP traffic.

Colin

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From owner-iptel-outgoing@lists.research.bell-labs.com  Fri May 28 03:49:23 1999
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Sorry for the very late reply... forgot I didn't respond to this.

Tom Gray wrote:
> 
> --- Jonathan Rosenberg <jdrosen@dnrc.bell-labs.com> wrote:
> 
> >
> > CPL definitely does not handle certain features -
> > call waiting, for
> > example. In particular, features which require user
> > input after the call
> > is established, or features which require call
> > state, are not possible.
> > In an IP telephony context, many, if not all, of
> > these features are most
> > easily supported in end systems. CPL is best at
> > providing call routing
> > related services; the infinite flavors of
> > forwarding, follow-me,
> > screening, and so on. These are services which
> > require network support,
> > often since the user may not be logged in or
> > otherwise available. I
> > don't think we can enumerate them all. If we could,
> > may as well number
> > them, do away with CPL, and have users select a
> > service by number. But,
> > after all, CPL is a scripting language of sorts, and
> > the point is we can
> > use it to develop new services (at least those
> > supportable within the
> > primitives CPL provides. Its not Turing complete, so
> > everything is not
> > possible).
> >
> > This more restricted scope helps mitigate many
> > feature interaction
> > problems. There are some we know of (the call
> > forward/call screening
> > interaction), and these are documented in the
> > framework. If you know of
> > any other interactions you see, please post them.
> >
> > It would also be helpful if you could point out
> > particular features you
> > feel are useful yet not supported by CPL.
> >
> 
> The CPL documents describe features like Call Forward Busy, Call
> Forward No Answer and the refusal of anonymous calls. These certainly
> are features that have meaning only if the user is present.

Call forward no answer only takes place when the user is *not* present.

> Additionally a useful feature to have is the ability to set an
> automatic call back on a user who is not registered or present and
> another is to camp on to a busy line (ie queue till the user becomes
> free.

The first one (callback) is not one I think CPL is good at doing. The
problem is that in the callback case, the called party calls back the
calling party when available. This is easily done in a user agent
(assuming it has access to call logs when it logs in), but harder in a
SIP server since the SIP server doesn't usually initiate calls or
maintain call state. None of the features we're targeting for CPLv1
require call state or having the server initiate calls, bridge calls,
etc.

The second service is not something I think that we would do in the
first version (it requires call state in the SIP server), but possibly a
later one. In that case, the CPL might look something like:

<proxy>
  <busy>
    <camp-on timeout="1h">
      <failure>
        <respond status="noanswer"/>
      </failure>
    </camp-on>
  </busy>
</proxy>

and for forwarding:

<proxy>
  <busy>
    <location url="sip:forwarded-location@company.com">
      <proxy/>
    </location>
  </busy>
</proxy>

So, if the called party specifies the service ala CPL, there would be no
feature interaction between forwarding and camp-on since they CANNOT
represent both at the same time in the same place in CPL. 

Interactions do arise if the called party specifies one of them (say
camp-on on busy) and the calling party tries to specify a different one
(forward on busy). This is not strictly a CPL issue, since somehow there
is here a way for the caller to ask for this feature for this call (the
caller preferences document is the general mechanism for this). In that
document, we mention that there is this generally hard interaction
problem. I don't have an easy answer for how its done, nor do I think
there is one right way.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       Lucent Technologies
Member of Technical Staff                   101 Crawfords Corner Rd.
High Speed Networks Research                Holmdel, NJ 07733
FAX: (732) 834-5379                         Rm. 4C-526
EMAIL: jdrosen@bell-labs.com
URL: http://www.cs.columbia.edu/~jdrosen

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From owner-iptel-outgoing@lists.research.bell-labs.com  Mon May 31 19:05:22 1999
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Message-ID: <37531225.6DD9@iptel.de>
Date: Tue, 01 Jun 1999 00:50:13 +0200
From: Pascal Merle <Pascal.Merle@iptel.de>
Organization: PAM Software GmbH
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To: iptel@lists.research.bell-labs.com
Subject: iptel trademark
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Hi!

Did anybody register "iptel" as a trademark for the standards
developed by the working group?

We are currently having problems with our domain "iptel.de":

A company named "tiptel" wants to forbid us the use of "iptel"
because they registered the almost similar "tiptel" as
a trademark.

Thanks for your help,
Pascal

-- 
Phone   : +49 6131 41537               PAM Software GmbH
Fax     : +49 6131 466138                Zur Oberlache 5
Mobile  : +49 171 7029415         D-55124 Mainz, Germany
Video   : callto:ils.uni-mainz.de/Pascal.Merle@iptel.de
Look at our new Telephony Soundcard at http://www.iptel.de

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