SIP in higher education

[(Dennis Baron)]

Call tomorrow, Thursday, 3/3/05 2:00-3:00 EST (UTC-5)

Agenda

- Minutes from 2/3/05 meeting - see below - any corrections?

- Deployment update - Harvard University - Candace Holman

  Candace is going to give us an update on the Harvard SIP.edu
  deployment.  And bring your updates and questions!

- Provisioning and configuring SIP phones - Art Gaylord

  The folks at WHOI have been working on how to configure SIP phones
  centrally.

- As always, let me know if you have any other items for this or
  future agendas!

                     Dennis

Thursday 2:00 PM US/Eastern (UTC-5)
To join the call, do one of the following:
  a. Dial +1-734-615-7474, or
          +1-800-392-6130, or 
          +1-866-411-0013 and enter access code 0153458, or
  b. Call 
sip:,
 or
  c. To have the system call you, click here:
          https://edial.internet2.edu/call/0153458

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SIP.edu Conference Call February 3, 2005

*Attendees*

Dennis Baron, MIT
Tim Bastian, Penn State
Steve Blair, University of Pennsylvania
Candace Holman, Harvard
Jeff Kuure, Internet2
Richard Machita, Alaska
Yul Pyun, University of Hawaii
Ben Teitelbaum, Internet2
Doug Walston, Cisco
Garrett Yoshimi, Hawaii



*Discussion*

Today's call begins with Dennis mentioning that Indiana University
is interested in getting involved with SIP.edu - they have completed
their implementation, which is based on the Interactive Intelligence
product. They have also submitted deployment notes for the cookbook,
and  will possibly be available for a future call.

The main topic for this call involves a previous discussion from this
fall about SIP peering with other providers, without necessarily
implementing E.164 dialing. Call participants are asked their thoughts
on allowing providers such as Free World Dialup or SIPphone to reach
SIP.edu users and vice versa. One possibility involves using access
codes, such as Freeworld's **-3-digit-number code, for peering with
others. FWD has reserved **EDU(338) for possible use with
SIP.edu. Calling in the other direction could be enabled by putting
the access code for FWD and SIPphone into SIP.edu proxies and PBXs,
allowing calls from PBX phones by dialing **FWD(393) or **SIP(747).

This leads into a discussion about the general state of peering with
external providers, what sort of services and options are becoming
commercially available, and the limitations of traditional 12-key
phones. Overall consensus is that peering will have to be investigated
more thoroughly in the future, but currently more technical knowledge is
required and specific needs will have to be examined. There is interest
about these issues, but no current momentum to investigate further.

A question is posed about the testing of E.164 dialing - overall, people
are interested in  such a thing but most call participants are currently
too busy or otherwise involved to devote a lot of time to it. Others are
unconvinced of the benefits in relation to the amount of configuration
work necessary to implement such a thing.

Following this is discussion about allowing current 12-key phones to
call SIP.edu URIs - there are workarounds involving  using a third-party
call control and a web based front end to place calls for legacy phone
users. At MIT, for example, you can authenticate users with a password
and the call control will set up the call back for users of 12-key
phones. This allows PBX users to contact SIP URIs without needing a SIP
enabled telephone. Client registration is an issue, and some participants
would prefer to upgrade phones rather than setting up compatibility for
legacy devices.

The call concludes with discussion of an upcoming workshop. Doing
something in conjunction with Joint Techs in the fall seems to be a
reasonable idea, depending on the date. The current plan is to have a
hands-on SIP workshop to build clients at the spring member meeting, all
day on Thursday and half a day on Friday. There will be nothing official
at the VON conference, except for a casual gathering of some sort.

Candace gives a brief summary of the Harvard issues - the LDAP plug in
created by Avaya is not in the call routing yet, and is not producing
any log messages. Candace is waiting for the developer to get back in
touch with her.  She also asks if anyone is interested in sharing a
T1 for purposes of connecting to SIP.edu. Dennis mentions that this
might be of interest to other Centrex users, like Harvard. Possibly
subscribing to a SIP termination service would be more cost effective,
depending on expected traffic.

Finally, Richard asks about working with a group to help configure
their gateway. Dennis asks Cisco people to assist in a configuration
demo, including PBX configuration. The gateway configuration would be
different based on what PBX is in use, but this could be figured out at
the workshop. Some configuration files will be posted to offer examples.

The next call will be on Thursday, February 17th.

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