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Subject: SIP in higher education

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Re: [] Call Notes - 3/30

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  • From: (Dennis Baron)
  • To:
  • Subject: Re: [] Call Notes - 3/30
  • Date: Thu, 06 Apr 2006 09:00:31 -0400 Conference Call March 30, 2006


Dennis Baron, MIT
Alan Crosswell, Columbia
John Hess, UC Berkeley
Jerry Keith, UC Riverside
Jeff Kuure, Internet 2
Yul Pyun, University of Hawaii
Dave Roeper, Independent Consultant
John Todd, Tello
Ben Teitelbaum, Internet 2
Chris Trown, University of Oregon
Garrett Yoshimi, University of Hawaii
Dave Zimmerman, UC Berkeley


Today's call begins with Chris Trown asking about OpenSER
configuration. He has been unsuccessful in getting digest authentication
working. Dennis says that MIT has used it for a while, and he can forward
any specific questions to Mark Silis, who set up MIT's installation.

The majority of the call is devoted to John Hess discussing Berkeley's
RFP for a new campus VoIP solution. Berkeley is currently an AT&T Centrex
customer and does not own or operate their own traditional PBX. The
contract with AT&T runs out in two years, and they are interested in
taking over and maintaining their own voice service on campus. They hope
to award the contract in two weeks.

The RPF was fairly agnostic towards solutions, and they were willing to
consider a variety of options including TDM, SIP, or any combination. The
initial scope of the RFP is for 10% of the campus voice network which
currently consists of a total of 20,000 lines. Student residences are
covered by a separate Centrex contract and are not part of this RFP. Five
campus buildings which are either new construction or are undergoing
seismic retrofit will be included in this contract.

John mentions that his department is fairly conservative and has been slow
to adopt VoIP for endpoints. Many locations on campus still use coax in
the riser and AUI drops to desktops, which make it difficult to do VoIP.
Buildings that do have decent switched ethernet may not have electrical
power in closets or at distribution points. The RFP encouraged vendors
to come up with creative solutions, and never specifically mentioned SIP;
however, this is the implied protocol.

Berkeley has received responses from major vendors of traditional voice
networking companies, and providers of Nortel, Avaya, NEC, and Siemens
products. Proposals have ranged from traditional TDM from the central
servers to endpoints, to hybrids with a SIP core and IP trunking to
gateways between TDM or SIP endpoints. They have been pleased with the
spectrum of responses. The first network will be coming online in July,
so they have a short time to finish negotiations and order equipment.

Dennis asks if Berkeley is planning to cut over all Centrex users at
once. John says that the first part will cover the five buildings and
approximately 2500 lines. During this time they will be training their
staff to become familiar with the system so they can install and maintain
it themselves. They hope to move away from Centrex over the next few
years, depending on budget. The Centrex contract allows them to reduce
number of lines within a certain level without penalties.

John is asked how the dialing plan will be managed. The current plan
uses a 5-digit campus plan. The new system will integrate with the
existing Centrex system, so new users can use same dial plan and same
voicemail. Ideally, there will be no noticeable difference for users
between the two systems. The billing to customers won't change, but the
way they do billing will. They also expect to have long-distance calls
that are independent of the Centrex.

Integration with the central voicemail system is important. As of January
Berkeley converted from an old Digital Sound system to an Interactive
Intelligence system, which is more SIP oriented. The SIP based proposals
have said integration won't be a problem.

Dennis asks if the Interactive Intelligence Communite' product uses
the PRI for signaling with the Centrex, or if it also has an SMDI
interface. John says that as of now it only supports the message waiting
indicator using SMDI; it cannot use the D-channel or Q.931 signaling
to set or clear the MWI. For the current integration with the Centrex,
which is hosted on a DMS100, the incoming caller ID is signaled across
the D-channel; to light the MWI the Communite' uses a shared SMDI modem
to signal back to the DMS100 to light the lamp or set or clear the MWI.

Dennis asks if there is a method in Q.931 to set the message
indicator. John says that when they did another RFP two years ago for
a small PBX at a remote campus, they selected a Nortel Meridian Option
61. The original proposal was to use a separate SMDI link from the PBX
to their then-current voicemail system. They were doing a SIP pilot at
the time and realized that they should be able to signal through the
D-channel to set or clear the MWI. They were using a Lucent PSTN gateway
which was not up to current draft specs, but they were able to prove to
the Nortel and AT&T people that they could signal from the Nortel PBX to
the DMS 100, then across the SMDI links. It is unclear to John whether
the Q.931 signaling elements that they are using are Nortel proprietary,
or in the Q.931 spec.

Dennis believes that their Lucent PBX may not have some of the
capabilities that the Nortel ones have for Q.931 extensions or extra
features such as passing caller name over a PRI. MIT has given up on
passing calling name for the time being. On their PBX to SIP calls, they
use the calling number to do an LDAP lookup and reinsert the calling
name. Chris Trown says that Oregon's Avaya passes all this information;
Dennis is surprised that they can't, and believes that they haven't been
able to find the right settings on their Lucent switch.

John Todd asks if there were any particular codec requirements or encoding
systems in the RFP. This has been left open, as Berkeley didn't include
this level of detail.

Dennis asks about the ownership of infrastructure on campus. The CO
that serves their Centrex is immediately adjacent to campus; the dial
tone in the Centrex is served by copper from the CO which comes onto
campus at 9 locations around the perimeter. The intra-campus copper
plan is technically owned by AT&T. But if the conduit breaks or gets
cut, Berkeley has to pay for it. In the RFP they specifically required
solutions to not use the the copper and instead to use the campus
fiber. In the initial scope for the five buildings, there will be all
new voice-grade copper riser and horizontal. Doing TDM distribution to
endpoints is easier at first. For buildings beyond the initial scope,
changes will be on a building-by-building basis. Buildings may be done
with TDM depending on budget and existing voice riser, while new buildings
might get VoIP endpoints.

Dennis asks about the current mix of analog and digital station
equipment. Right now there it is slightly over 10 percent digital and the
vast majority are single-line analog sets. Costs for digital sets are in
the hundreds of dollars compared to tens for analog sets. Departments
aren't forced to buy digital sets, but if they want the features that
can only be delivered over digital sets they can purchase them.

Dennis also is interested in John's opinion on solutions that are VoIP
to the building and FXS to analog phones, or reusing proprietary digital
phones with some sort of conversion box. In looking at total cost of
ownership, Berkeley realized that a Nortel solution would probably allow
them to reuse all their existing digital sets, and this was a major
consideration in scoring for the proposals. However, the percentage if
digital sets on campus is low, so a non-Nortel solution would have to
be very expensive to affect total cost.

John is asked about future plans for the 90% of analog phones on
campus. He says that if it made monetary sense, and provided the same
call reliability and quality as well as E911 support, then they would
convert to VoIP.

Chris Trown asks about Berkeley's current switching environment. There
are currently 45,000 network connections on campus; over half of these are
switched using a combination of Cisco, Foundry, and Extreme. The routing
is all Cisco, except at the border where where they use Junipers. Chris
also asks about E911, and John says that this was in the RFP and all
proposals must be compatible.

Garrett asks if offers were allowed to propose soft phones or only
hardware phones. In the initial scope hard phones were required, but
optional equipment could include soft phones or expanded digital or
analog sets. The optional section had its own scoring. Garrett asks if the
options also included things like conferencing, multimedia, or presence.
Conference bridging and ACD functionality were core requirements in
the RFP. Conferencing is currently an added service but not part of the
Centrex contract. ACD is not part of the existing Centrex environment,
and call centers that need this offer it by other means.

John is asked about the price point where SIP phones would be considered
affordable. He says that it's more than the physical instrument, but
also the infrastructure to support it. Currently they charge customers to
install a phone cable, which is the same cost as installing a data cable,
in addition to the recurring service fee. At some point it will make
sense to only run one cable, but the digital sets are now so much more
expensive than analog phones. Berkeley has a campus wireless project,
and their customers would also like GSM and voice-over-WiFi service. In
the future, there may be fewer desk phones and more mobile devices that
render this discussion irrelevant.

Finally, Dennis asks if the RFP had any preference towards another
Centrex system, or a hosted IP-based proposal. John says that they had
a strong preference for a non-Centrex environment, and had no interest
in a hosted environment.

The next call will be on April 13th.

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