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Thread-Topic: Scalability of Netconf
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From: "Branislav Meandzija" <bmeandzija@packeteer.com>
To: "Len Nieman" <lwnieman@bellsouth.net>, <netconf@ops.ietf.org>
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Cc: j.schoenwaelder@iu-bremen.de, nmrg@ibr.cs.tu-bs.de
Subject: [nmrg] RE: Scalability of Netconf
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Yes. How about the case where devices update their config autonomously and communicate this back to the NMSs which need to reconcile the updates and re-distribute them?

Branislav

> -----Original Message-----
> From: owner-netconf@ops.ietf.org [mailto:owner-netconf@ops.ietf.org]On
> Behalf Of Len Nieman
> Sent: Monday, August 15, 2005 6:21 PM
> To: netconf@ops.ietf.org
> Cc: j.schoenwaelder@iu-bremen.de; nmrg@ibr.cs.tu-bs.de; Eliot Lear
> Subject: Re: Scalability of Netconf
> 
> 
> More likely a single NMS trying to manage 800 - 1,200 routers.
> 
> Len
> 
> >Date: Wed, 06 Jul 2005 12:15 -0400 (EDT)
> >From: Eliot Lear <lear@cisco.com>
> >To: j.schoenwaelder@iu-bremen.de
> >CC: netconf@ops.ietf.org,
> >    nmrg@ibr.cs.tu-bs.de
> >Subject: Re: Scalability of Netconf
> >
> >Yeah, I just had this funny thought of 1200 NMSes trying to 
> configure a
> >router.  That's the sort of locking contention we certainly did NOT
> >optimize for!
> >
> >Eliot
> >
> >
> >Juergen Schoenwaelder wrote:
> >> On Tue, Jul 05, 2005 at 04:01:17PM +0200, Juergen 
> Schoenwaelder wrote:
> >> 
> >> 
> >>>I am running this script on a 3GHz Pentium with 512 MB Ram 
> (so really
> >>>nothing high end for a management system) running Linux 2.6.10. I 
> >>>keep 300 connections open and the funny thing is that this 
> box does 
> >>>not at really get stressed by this. The load never went above 0.1% 
> >>>and was most of the time significantly below. The memory 
> used excluding
> >>>buffers was ~235 MB.
> >> 
> >> 
> >> Today I was running 1200 ssh connections using the same 
> basic script
> >> on the same box. The load again never went above 0.1 while 
> the memory 
> >> usage went up to ~494 MB. Shall I try 10.000 tomorrow? With 2GB of
> >> memory this should be possible. The two Linux XEON servers 
> that play 
> >> the remote end for 600 connections each do not seem to bother much 
> >> about all this.
> >> 
> >> /js
> >> 
> >
> >--
> >to unsubscribe send a message to netconf-request@ops.ietf.org with
> >the word 'unsubscribe' in a single line as the message text body.
> >archive: <http://ops.ietf.org/lists/netconf/>
> 
> --
> to unsubscribe send a message to netconf-request@ops.ietf.org with
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> 



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Len Nieman wrote:
> More likely a single NMS trying to manage 800 - 1,200 routers.

We'd like to aim for higher.

Eliot


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More likely a single NMS trying to manage 800 - 1,200 routers.

Len

>Date: Wed, 06 Jul 2005 12:15 -0400 (EDT)
>From: Eliot Lear <lear@cisco.com>
>To: j.schoenwaelder@iu-bremen.de
>CC: netconf@ops.ietf.org,
>    nmrg@ibr.cs.tu-bs.de
>Subject: Re: Scalability of Netconf
>
>Yeah, I just had this funny thought of 1200 NMSes trying to configure a
>router.  That's the sort of locking contention we certainly did NOT
>optimize for!
>
>Eliot
>
>
>Juergen Schoenwaelder wrote:
>> On Tue, Jul 05, 2005 at 04:01:17PM +0200, Juergen Schoenwaelder wrote:
>> 
>> 
>>>I am running this script on a 3GHz Pentium with 512 MB Ram (so really
>>>nothing high end for a management system) running Linux 2.6.10. I 
>>>keep 300 connections open and the funny thing is that this box does 
>>>not at really get stressed by this. The load never went above 0.1% 
>>>and was most of the time significantly below. The memory used excluding
>>>buffers was ~235 MB.
>> 
>> 
>> Today I was running 1200 ssh connections using the same basic script
>> on the same box. The load again never went above 0.1 while the memory 
>> usage went up to ~494 MB. Shall I try 10.000 tomorrow? With 2GB of
>> memory this should be possible. The two Linux XEON servers that play 
>> the remote end for 600 connections each do not seem to bother much 
>> about all this.
>> 
>> /js
>> 
>
>--
>to unsubscribe send a message to netconf-request@ops.ietf.org with
>the word 'unsubscribe' in a single line as the message text body.
>archive: <http://ops.ietf.org/lists/netconf/>


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--SUOF0GtieIMvvwua
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Hi.

At the last NMRG meeting, we discussed that it would be useful to
have a paper which describes the various VoIP metrics being defined
(e.g. RTCP XR) and the mechanism to access or transport them. The 
idea was to provide a summary like overview which can serve as an
introductionary reference for people entering the field.

Today, I received the attached call for papers for the Network and
Service Management Series of the IEEE Communications Magazine series.
This seems to be a natural fit and it would be very timely if
something suitable could be produced by the deadline (Sept. 15).
I am happy to review a draft paper in case there is interest in
some review service.

/js

-- 
Juergen Schoenwaelder		    International University Bremen
<http://www.eecs.iu-bremen.de/>	    P.O. Box 750 561, 28725 Bremen, Germany

--SUOF0GtieIMvvwua
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********************************************************************

                           Call for Papers

  IEEE Communications Magazine - Network & Service Management Series

                 Second issue (deadline: September 15, 2005)

         http://www.comsoc.org/pubs/commag/cfpcommag1005.html

********************************************************************


IEEE Communications Magazine announces the creation of a new series on
Network and Service Management. The series will be published twice a
year, with the first issue already under way to be published in October 2005
and the second issue accepting submissions now to be published in March 2006.
The series intends to provide articles on the latest developments in this
well-established and thriving discipline. Published articles are
expected to highlight recent research achievements in this field and
provide insight into theoretical and practical issues related to the
evolution of network and service management from different perspectives.
The series will provide a forum for the publication of both academic and
industrial research, addressing the state of the art, theory and
practice in network and service management. Both original research and
review papers are welcome, in the style expected for IEEE Communications
Magazine. Articles should be of tutorial nature, written in a style
comprehensible to readers outside the speciality of Network and Service
Management. This series therefore complements the newly established IEEE
Electronic Transactions on Network & Service Management (eTNSM) -
http://www.comsoc.org/etnsm/.


General areas include but are not limited to:
- Management models, architectures and frameworks
- Network and service provisioning, reliability and quality assurance
- Management functions
- Management standards, technologies and platforms
- Management policies
- Applications, case studies and experiences

IEEE Communications Magazine is read by tens of thousands of readers
from both academia and industry. The magazine has also been ranked the
number one telecommunications journal according to the ISI citation
database for year 2000, and the number three for year 2001. The
published papers will also be available on-line through Communications
Magazine Interactive, the WWW edition of the magazine. Details about
IEEE Communications Magazine can be found at http://www.comsoc.org/ci/.

Schedule for the first issue:
- Manuscripts due: September 15, 2005
- Acceptance notification: November 30, 2006
- Publication date: March 2006

Series Editors:
Prof. George Pavlou, University of Surrey, UK.
Dr. Aiko Pras, University of Twente, Netherlands.
 

--SUOF0GtieIMvvwua--


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I tried to write down some conclusions from the meeting in Paris that
can go into the next report I have to submit to the IRTF chair. Please
review the following text and share your comments. As usual, concrete
proposals in the form "replace this with that" are especially welcome.

---

[1] The NMRG held its 18th meeting in Nancy on July 30/31. The meeting
    focused on VoIP management. The details can be found at:

    <http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/>

    During the meeting, topics such as measurements and metrics, fault
    isolation and reporting mechanisms, provisioning, data modeling,
    speech quality, and security aspects were discussed in the context
    of VoIP applications. Several observations were made:

    (1) VoIP management requirements largely depend on the underlying
        VoIP business model. There seems to be a continuum of business
        models ranging from classic centralized managed voice services
        to fully distributed P2P voice services where management
        aspects are handled within a distributed P2P infrastructure.
        It might be useful to document the management requirements for
        typical VoIP business models.

    (2) There are several ongoing activities to define VoIP metrics
        and mechanisms to obtain / access / distribute / aggregate
        measurements.  There seems to be a need for a document which
        explains how the various activities fit together and
        complement each other. There also seems to be a need for
        additional research on aggregation mechanisms tailored to VoIP
        metrics.

    (3) For VoIP calls, especially those which cross network domain
        boundaries, it seem to be useful to explore how VoIP endpoints
        and intermediaries can take an active role in fault isolation
        and diagnostics by providing detailed fault reports that
        include information gathered from other nodes in the network.
        This requires the definition and execution of suitable
        diagnostic procedures, potentially executed on nodes in
        different parts of the Internet.

    (4) On a longer perspective, there is interest in more fundamental
        research to replace the classic manager-agent management model
        with a more distributed P2P style management model where nodes
        in the network cooperate to identify and solve problems and to
        exchange management information. Another way to describe this
        would be a self-organizing management overlay. This approach
        seems especially interesting for managing applications
        crossing domain boundaries and which themself organize in a
        P2P fashion (some VoIP solutions being an example for this).

/js

-- 
Juergen Schoenwaelder		    International University Bremen
<http://www.eecs.iu-bremen.de/>	    P.O. Box 750 561, 28725 Bremen, Germany


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On Mon, Aug 01, 2005 at 07:29:31PM +0300, Romascanu, Dan (Dan) wrote:

> Replace: 
> 
> 'FCAPS SIP Data or Information Model'
> 
> By:
> 
> 'SIP-enabled VoIP FCAPS Data or Information Model'

done

/js

-- 
Juergen Schoenwaelder		    International University Bremen
<http://www.eecs.iu-bremen.de/>	    P.O. Box 750 561, 28725 Bremen, Germany


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> -----Original Message-----
> From: nmrg-bounces@ibr.cs.tu-bs.de 
> [mailto:nmrg-bounces@ibr.cs.tu-bs.de] On Behalf Of Juergen 
> Schoenwaelder
...
> That said: Do you have a concrete proposal how to change the 
> draft minutes?
> 

Replace: 

'FCAPS SIP Data or Information Model'

By:

'SIP-enabled VoIP FCAPS Data or Information Model'

Dan






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From: Juergen Schoenwaelder <j.schoenwaelder@iu-bremen.de>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
Subject: Re: [nmrg] draft nancy meeting minutes
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On Mon, Aug 01, 2005 at 07:00:30PM +0300, Romascanu, Dan (Dan) wrote:
> Juergen,
> 
> I remember that we had a discussion wrt. the scope of the model in item
> #1. The outcome was as far as I remember that we will deal with FCAPS
> model for SIP-enable networks rather than 'FCAPS for SIP' the argument
> being that VoIP is not only SIP, and in order to manage a SIP solution
> you need to define management object beyond the borders of the SIP
> protocol itself.  

I think I asked this question, but I am not so sure what the answer
really was. Personally, I believe this work needs to be scoped since
VoIP at then end even runs over Ethernet - but I do consider that to
be out of scope, which you might agree with.

That said: Do you have a concrete proposal how to change the draft
minutes?

/js

-- 
Juergen Schoenwaelder		    International University Bremen
<http://www.eecs.iu-bremen.de/>	    P.O. Box 750 561, 28725 Bremen, Germany


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Juergen,

I remember that we had a discussion wrt. the scope of the model in item
#1. The outcome was as far as I remember that we will deal with FCAPS
model for SIP-enable networks rather than 'FCAPS for SIP' the argument
being that VoIP is not only SIP, and in order to manage a SIP solution
you need to define management object beyond the borders of the SIP
protocol itself.  

Regards,

Dan


> -----Original Message-----
> From: nmrg-bounces@ibr.cs.tu-bs.de 
> [mailto:nmrg-bounces@ibr.cs.tu-bs.de] On Behalf Of Juergen 
> Schoenwaelder
> Sent: Monday, August 01, 2005 6:52 PM
> To: nmrg@ibr.cs.tu-bs.de
> Subject: [nmrg] draft nancy meeting minutes
> 
> Hi!
> 
> Attached are the draft meeting minutes of the 18th NMRG 
> meeting last weekend in Nancy. Aiko, who volunteered to 
> produce the minutes, has already left for vacation and so I 
> am now in charge to finalize them. If you think something is 
> wrong and important things are missing, please let me know 
> quickly since I want to publish the final version of the 
> minutes before the weekend.
> 
> Silence will be counted as agreement.
> 
> /js
> 
> -- 
> Juergen Schoenwaelder		    International University Bremen
> <http://www.eecs.iu-bremen.de/>	    P.O. Box 750 561, 
> 28725 Bremen, Germany
> 



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Hi!

Attached are the draft meeting minutes of the 18th NMRG meeting last
weekend in Nancy. Aiko, who volunteered to produce the minutes, has
already left for vacation and so I am now in charge to finalize
them. If you think something is wrong and important things are
missing, please let me know quickly since I want to publish the final
version of the minutes before the weekend.

Silence will be counted as agreement.

/js

-- 
Juergen Schoenwaelder		    International University Bremen
<http://www.eecs.iu-bremen.de/>	    P.O. Box 750 561, 28725 Bremen, Germany

--gKMricLos+KVdGMg
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Minutes of the 18th NMRG meeting
INRIA/LORIA, Nancy, France
30-31 July 2005
Minutes: Aiko Pras

Participants:
- Michael Alexander (WU Vienna, Austria) 
- Remi Badonnel (LORIA-INRIA, France)
- Vincent Cridlig (LORIA-INRIA, France)
- Olivier Festor (LORIA-INRIA, France) 
- James Hong (Postech, Korea) 
- Christian Hoene (TU Berlin, Germany) 
- Azita Kia (Cisco Systems, USA) 
- Abdelkader Lahmadi (LORIA-INRIA, France)
- Saverio Niccolini (NEC Europe, Germany) 
- Amy Pendleton (Nortel Networks, USA) 
- David Perkins (SNMPInfo, USA) 
- Aiko Pras (University of Twente, the Netherlands) 
- Juergen Quittek (NEC Europe, Germany) 
- Dan Romascanu (Avaya, Israel) 
- Juergen Schoenwaelder (International University Bremen, Germany) 
- Henning Schulzrinne (Columbia University, USA) 
- Radu State (LORIA-INRIA, France) 

Agenda:
Saturday (2005-07-30)
- Welcome (Juergen Schoenwaelder, Olivier Festor)
- Real-time Application Quality of Service Monitoring (Dan Romascanu)
- RTP Control Protocol Extended Reports (Amy Pendleton)
- RTP, RTCP XR and SIP MIB Modules (Dan Romascanu)
- SIP Service Quality Reporting (Amy Pendleton)
- Pre-provisioning, Template, Individual and Per-Subscriber Provisioning 
  for VoIP Services (Michael Alexander)
- Management and QoS for VoIP (Henry Sinnreich / Juergen Schoenwaelder)
- User-oriented Management of VoIP Applications (Henning Schulzrinne)
- Service Provider VoIP OSS - Lessons Learned (Azita Kia)
- Calculation of Speech Quality by Aggregating the Impacts of Individual 
  Frame Losses (Christian Hoene)
- VoIP Security Threat Analysis (Saverio Niccolini)
- VoIP Fuzzer, Cracker and Spammer - incl. demo (Olivier Festor, Radu State)

Sunday (2005-07-31)
- Discussion: Identify main research/engineering questions to be addressed
- Discussion: Develop plans how the research/engineering questions could be 
  addressed
- Wrap Up (Juergen Schoenwaelder, Olivier Festor)


=======================================================================

10:15: Welcome by Juergen Schoenwaelder. Roll call.

The structure of this meeting differs from traditional NMRG meetings,
in the sense that the first day is entirely reserved for presentations.
Many discussions will therefore be postponed until the second
day. Saturday morning started with four presentations that covered
more or less identical topics: monitoring VoIP quality at end devices.

10:25 	Real-time Application Quality of Service Monitoring (RAQMON)
	Dan Romascanu (Avaya, Israel)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/raqmon.pdf 

RAQMON is being developed by the IETF RMON WG. There are three
Internet Drafts, covering the following topics: Framework, PDUs and
MIB. The idea is that every end device includes a RAQMON data source,
which sends RAQMON PDUs to a collector. These PDUs are used by the
collector to maintain "a kind of RMON MIB", which can be queried by
managers via SNMP. Information between data source and collector is
exchanged using TCP or SNMP notifications. Many parameters have been
defined, like addresses of the communicating parties, number of
packets, delay, jitter, loss and CPU utilization; for a complete
overview see the slides.

There are currently two different code bases, and probably 10
applications that use these bases. The question was raised for whom
RAQMON was developed. Dan answered that RAQMON may be interesting for
providers running VoIP or video distribution networks. A discussion
followed on the fact that RAQMON should be implemented in all end
systems. Since there may be millions of them, scalability might become
problematic.

11:00 	RTP Control Protocol Extended Reports (RTCP XR)
	Amy Pendleton (Nortel, USA)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/rtcp-xr.pdf 

Control Protocol Extended Reports (RTCP XR) is being developed by the
IETF Audio/Video Transport (AVT) WG. Its definition can be found in
RFC 3611, which is at Proposed Standard level.

The principle goal is facilitate monitoring / measurements on a per
call basis. The provided metrics include: loss, jitter, high/low
signal level, echo, noise and delay; for a complete overview see the
slides. Possible application scenarios include fault management,
provisioning and accounting. The presentation included a number of
measurement results. One of the interesting conclusions was that it is
not very useful to know only average loss / discard figures; it is
much more important to know the distribution. To determine such
distribution, a four-state Markov Packet Loss model is proposed. There
are two states for Gaps, which indicate minor problems, and two for
Bursts, which indicate serious problem. A window of 16 packets is
needed to maintain these states. Henning questioned the use of four
states, and assumed that two would be sufficient. Amy continued her
presentation with explaining how signal level problems, like clipping,
can be detected, as well as Echo. The last part of her presentation
concentrated on the design philosophy behind RTCP XR, as well as a
framework for VoIP performance management (see slides). The framework
allowed the use of probes to capture VoIP related data. In the
subsequent discussion the problem was raised of how to analyze such
data in case of encryption, as well as the level at which VoIP quality
measurements should take place. Christian believed that such
measurements should be performed at a much higher level, yielding easy
to use metrics. He said that there are many research papers that
demonstrate that VoIP quality can be measured with little knowledge of
network parameters. Henning did not agree on this, and also Azita said
that we still have a long way to go before we understand VoIP quality
measurement like we understand traditional PSTN telephone quality
measurements.

11:45 	RTP, RTCP XR and SIP MIB Modules
	Dan Romascanu (Avaya, Israel)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/voip-mibs.pdf 

Dan gave an overview of VoIP related MIB activities within the IETF.
Three MIB modules are under development; these modules are not
overlapping, but complementary.

The first module is the RTP MIB, which is being revised and currently
and Internet Draft. This module is intended for end systems, and keeps
track of call history. Aiko asked if this module could also be used
for "data retention" purposes. Dan wasn't sure, because the MIBs
structure may be too complex for this purpose.

The second module is the RTCP XR MIB, and is based on the extended
reports technology as defined in RFC 3611 Proposed Standard). The MIB
is structured around the following RTP concepts: Session, Sender and
Receiver. It can be implemented in RTP Host Systems, as well as
Intermediate Systems. The data model includes History as well as Call
Quality parameters.

The third module is the SIP MIB, which is still an Internet Draft and
not on standards track yet. The module can be implemented in all kinds
of SIP entities (see RFC 3261), thus within User Agents, as well as
Servers (Proxy, Redirect and Registrar). The supported operations
include status monitoring, protocol statistics and configuration of
notifications. The structure and indexing of the SIP MIB is based on
the Network-Services MIB (RFC 2788).

It should be noted that all three modules can be used for monitoring
purposes only; they can not be used for VoIP configuration
purposes. For that purpose there are other tools and protocols,
however (see URL in Dan's slides). The level of deployment of the
three MIB modules is unclear; in fact the meeting participants
questioned that SNMP will become the technology of choice for VoIP
management.

12:10 	SIP Service Quality Reporting
	Amy Pendleton (Nortel, USA)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/sip-qr.pdf 

The last presentation of this morning session focused on event
notification capabilities in SIP. These notifications are not related
to the SIP MIB, but are separate messages with a textual syntax
following SIP conventions (which is not XML). The work is performed
within the SIPPING WG and still at Internet Draft level; in February
2005 it went for last call, but there were many comments that a new
draft needs to be created.

12:30 	Lunch Break
	
15:00   Management and QoS for VoIP
        Henry Sinnreich (pulver.com) and Juergen Schoenwaelder (IUB, Germany)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/qos.pdf 

Juergen made two provocative statements to trigger discussion. 

First, QoS is not an issue to worry about for VoIP in general, and for
VoIP management in particular. Second, P2P self organizing networks
will provide the highest possible availability for VoIP services.
Juergen argued that specific management mechanisms for VoIP may not be
necessary, but that we need more generic management mechanisms, which
are also useful for other applications. In addition, we do not need a
centralized "operator", but can have a P2P network to coordinate and
exchange management data (for his precise reasoning: see the slides).

Although some agreed with his statements, others strongly disagreed. A
discussion than started on where you need to measure: in end systems,
or elsewhere. Another discussion popped up whether perceived quality
does not largely depend on the codecs being used; Skype uses high
quality codecs and many therefore perceive Skype as a high quality
VoIP system. Some participants had doubts how easy it is to replace
the common G.7** codecs by higher quality codecs, since most of the
technology is patented and may not become available as open source
codecs.

15:40 	Preprovisioning, Template, Individual and Per-Subscriber 
	Provisioning for VoIP Services
	Michael Alexander (WU Vienna, Austria)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/voip-provisioning.pdf 

Provisioning a VoIP subscriber entails setting a variety of per-line
and per-customer group parameters in several devices and Operation
Support Systems (OSS). In mixed PSTN/VoIP Systems with local PSTN
gateways, at the minimum 3 devices need to be provisioned; for
in-network VoIP the count is two plus OSS - with at the minimum
inventory management being affected. The resulting element and OS
management overhead is considerable especially when considering that
CPE provisioning is expected to be performed primarily by operators
going forward. Michael presented various approaches to lower the
management burden of provisioning VoIP circuits. When utilizing
pre-provisioning and template-based provisioning in addition to
per-subscriber provisioning, he claimed that the management burden
could potentially be reduced. The advantages of the former two
approaches in conjunction with per-circuit provisioning were discussed
and some notions on needs requirements for VoIP management protocols
were derived.

16:40 	User-oriented Management of VoIP Applications
	Henning Schulzrinne (Columbia University, USA)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/dyswis.pdf 

Henning explained that VoIP can fail for any number of reasons, from
low-level connectivity problems to signaling failure, NAT issues,
packet loss and jitter as well as subtle end-system-related problems
that have the same effect as network problems. Often, these problems
are transient and cannot be reproduced. While the user sees one
application, the end user application, user OS, home network, access
network, voice service provider (proxy operator) and the remote party
all need to cooperate to complete a call. Thus, there are probably
half a dozen parties that can be blamed, in mutual finger pointing, if
something goes wrong. Although Henning argued that you need a
relatively small number of "health measurement" tests to detect such
problems, traditional network management tools are of only limited
help in this environment. Henning's proposal was therefore to
introduce a "Do you see what I see" mechanism, which allows the
various parties to query neighboring systems if they experience
similar problems. For example, a system could ask a neighboring system
if it has connectivity, can pass the NAT, and if it is able to get a
certain DNS address. In the discussion the participants agreed that
such "Do you see what I see" mechanism will have certain implications
for security, which require further research.

17:05 	Service Provider VoIP OSS - Lessons Learned
	Azita Kia (Cisco Systems, USA)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/voip-oss.pdf 

Azita presented a short (3 slides) but nice historical overview of
VoIP management. She summarized some lessons learned in 10 years of
VoIP deployments, and identified some areas of standardization that
might help to improve these deployments. One of the recommendations is
to work further on accounting; this recommendation was followed by a
discussion what items we expect operators will charge for. Some argued
that all kind of supplementary services, like Calling Line
Identification Presentation, will be separately charged for. Others
argued that the investment for operators could be too high to charge
for this. In addition, customers may not be willing to pay for this
and move to alternative providers.
	
17:20 	Calculation of Speech Quality by Aggregating the Impacts of
	Individual Frame Losses
	Christian Hoene (TU Berlin, Germany)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/speech.pdf 

Christian gave a similar presentation as its previous IWQoS
presentation in Passau; his findings were part of his PhD work. His
message is that packet loss doesn't say much; it depends on which
packets get lost. There are less important packets, and more important
packets. Christian gave a demonstration, in which he simulated 35%
packet loss, and made clear that if the most important packets get
lost, you can not understand anything. If only the less important
packets get lost, voice is still understandable. Christian's results
may be useful for WIFI environments at home, or if you want to save on
energy of handhelds. Christian expected that the backbone network will
be good enough and does not need to do differentiate between important
and less important packets.

During the discussion, it became clear that there is currently no good
online mechanism to classify VoIP packets into important and unimportant
ones and that this is a subject for further research.

17:45 	VoIP Security Threat Analysis
	Saverio Niccolini (NEC Europe, Germany)

Saverio gave an overview of all kind of security threats for VoIP (see
slides for details). These threats come from the fact that the
signaling is sent using the same network as the multimedia data and
that the traffic is not encrypted. In his talk Saverio also presented
possible countermeasures and the pros and cons of some currently
proposed solutions.

Saverio's work is quite similar to the work of VoIP security Alliance,
although his work is less advanced. Saverio referenced the SIPPING
WG's Internet Draft on SIP SPAM
(http://ietf.org/internet-drafts/draft-ietf-sipping-spam-01.txt), and
told that NEC applied for a patent on this topic. Saverio's intention
is to submit this work as paper to IEEE Network Magazine Special Issue
on Securing Voice over IP.

18:00 	VoIP Fuzzer, Cracker and Spammer (incl. demo)
	Olivier Festor, Radu State (LORIA-INRIA, France)

http://www.ibr.cs.tu-bs.de/projects/nmrg/meetings/2005/nancy/fuzzer.pdf

The first day of the meeting ended with a demo on management of an
Asterix VoIP system, using INRIA/LORIA's netconf implementation. This
demo was followed by a demo of a security assessment tool, called
"fuzzy packet". This tool manipulates and generates data messages by
injecting and capturing packets into the network. It used "random"
users and passwords (generated by script), and is able to inject SPAM
into an ongoing call. Quite interesting!

19:00 	Leaving for Dinner

=======================================================================


July 31 - start at 9:30

Dan: There is a difference in understanding what VoIP management
really means.  Two extremes: 1) alternative / replacement of existing
POTS -> VoIP should comply to same regulatory requirements as VoIP
(legal interception / CDRs) -> has many technical / management
implications. Same reliability & QoS requirements as POTS. 2) Skype.

Dan explained the need in some environments to have CDRs. Aiko said
that you do not need SNMP like agent capabilities in end systems, but
you can use Netflow to create CDRs. Henning and Amy said that Netflow
captures data at a too low level, you have IP addresses and not
end-user (SIP) information.

Dan went on discussing about performance management, like we have for
the POTS (route optimization, trunk provisioning etc). He said he is
not aware of much work in this area (within standardization).

Aiko asked what the management requirements would be if the the second
extreme (=Skype) would be successful. Dan did not know if / what they
did to manage a network like things. Henning said they are primarily
monitoring / gathering statistics, but not active interfering. Henning
said that we're not clearly defining what we mean with management. Amy
said Skype is doing a lot of management within the application, like
buffer management / auto gain control.

Question by Juergen: What are the challenging problems we have to work
on? Henning made an inventory of possible management activities:
provisioning, SLA management, CDRs (AAA, billing, fraud management,
dimensioning), QoS Statistics (Real time, utilization), security (SBC,
firewalls, IDS, patch management), real-time alarms. Juergen said we
should not list all possible management functions.

Challenges for research include: security impact in management (like
intercept of RTCP-XR), automation versus human planning, scalability
(data reduction, filtering), what are the SLA metrics (problems in
case of multiple providers), modeling. Henning concluded that these
are not very specific for VoIP. Is there something that is specific
for VoIP?

Discussion started on what level you should define measures. On a per
call basis (fine granularities), or on a monthly (for example) basis?
Aiko believed that metrics are not much different between VoIP and
other application (like video).

Michael argued that you need more modeling. Currently the device has a
big MIB, and you don't know which objects you need to modify if you
want to achieve a specific effect. For TDM networks, to name an
example, we have such models, but not for IP. In addition, metrics
equivalent to errored seconds in TDM systems would be useful to have
for VoIP networks. Dave said that this is a well-known problem, but
that within the IETF we have not been able to do so, since you need
for this the operators, and they do not know.  Meeting did not agree
on the feasibility / possible success of such modeling work. We agreed
that we should have a small number of "objects".

Henning put on the white board possible protocol / technology
needs. For discovery you have several approaches, like LLDP, Bonjour
and SLP. It was noted that LLDP is based on the ENTITY-MIB (RFC 2737).

Juergen said that we are making the same mistake as in the past. We
think in terms of the manager-agent model, and believe that we're done
if the agents are able to provide all kind of information. The actual
problem, however, is how the "magic" / intelligence performed within
the manager looks like. Aiko said that he wants to get rid of such
centralized components as much as possible, let's put more
intelligence within the devices, like the "do you see what I see"
approach, outlined by Henning the day before.

Michael would like to see work on SIP specific FCAPS. Dan proposed, as
example, to discuss a bit more performance. Latency was mentioned as
one of the interesting metrics. The question was if you have to
measure this specific for SIP, but for all kind of applications.

Juergen collected proposals for further work:

1) FCAPS SIP Data or Information Model (could be an informational RFC)

2) VoIP metric framework (at a higher level than MIBs) which explains
   which metrics and which access/distribution mechanisms are being
   defined.

3) Fault incident reports (readable by machines) which provide a
   structured high-level explanation of a fault experienced by VoIP
   entity.

4) Fault isolation / diagnostic procedures that can be used locally or
   called remotely to produce fault incident reports.

5) Management implications of VoIP business models (enterprises, 
   operators who own the network, pure VoIP operators which do not 
   have a network and non-standard compliant operators, like Skype)

How to proceed with respect to the various proposals:

1) Juergen does not yet sufficiently understand this idea, and asked
   Michael to create a document that explains this idea a bit better
   and provides examples. Initiative taken by Michael, Dan is also
   interested.

2) VoIP metric framework could be an article for IEEE Communications
   Magazine or IEEE Surveys and Tutorials). Initiative taken by Amy,
   Dan and others.

3) Fault incident report. Initiative taken by Henning.

4) Fault isolation / Diagnostic procedures. Initiative taken by Henning.

5) Management implications of VoIP business models. Initiative taken
   by Aiko.

Meeting closed Sunday at 12:05

--gKMricLos+KVdGMg--

