SIP in higher education

[Dave Packham <>]

Just a question?  Why do all the SIP.edu meeting have to hang off the
Members meetings?  Would it be easier or better crowd for anyone
involved to hang off the Joint Techs meeting?  Or am I missing something

Dave Packham 

-----Original Message-----
From: 

 
[mailto:]
 On Behalf Of Dennis Baron
Sent: Wednesday, February 02, 2005 12:35 PM
To: 

Subject: Re: [sip.edu] Weekly SIP.edu Call Tomorrow


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

Agenda

- Minutes from last meeting - see below - any corrections?

- Deployment updates, issues, questions - all

  Maybe updates from Harvard and Indiana?  Others?

- SIP.edu Peering - By the Numbers - Dennis

  We didn't have time for this last week - so we'll try again...

  I'd like to revisit the topic we discussed in the fall -
  specifically the idea of peering with "open" SIP service providers.
  What are the possible service models?  Who would operate a central
  service?  Are there funding opportunities?  A copy of my slides from
  the last discussion are here:

         http://mit.edu/sip/sip.edu/presentations/np110.pdf

- Planning for SIP.edu Workshop at the Internet2 Spring Member Meeting

  Just an update from last week's call

- 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

--------

SIP.edu Conference Call January 20, 2005

*Attendees*

Dennis Baron, MIT
Gabriel Somlo, Colorado
Mike Van Norman, UCLA
Garrett Yoshimi, Hawaii
Ben Teitelbaum, Internet2
Tim Bastian, Penn State
Paul Heider, NOAA
Doug Walston, Cisco
Candace Holman, Harvard
Chris Celiberti, Pulver
Jeff Kuure, Internet2


*Discussion*

Today's call featured Mike Van Norman from UCLA discussing the
university's recent SIP explorations. The campus has had some VOIP
interest in the past, and implemented SIP approximately two months ago
using Cisco's AS 5300 as the gateway and iptel.org's SER as the SIP
proxy. It is still considered an experimental deployment and has not
been announced publicly - there is currently no expectation of
reliability or service availability, and some political issues still
need to be dealt with. SIP is being treated as a production service in
terms of deployment, but it must be more stable before it is announced
publicly.

Other campuses in the University of California system are interested in
SIP as well, particularly in terms of call routing via SIP between
campuses. SIP phone peering, however, is not being considered due to the
cost in terms of money and resources in relation to the limited
benefits. Having a generic gateway for use by anyone is considered to be
a better option.

User agent registration has not been discussed yet, but the system can
currently authenticate users. The main issue is finding software to
encrypt credentials. MIT uses MD5 currently, but TLS is being
investigated. UCLA currently uses UNIX crypt passwords and has not yet
gone to MD5.

When the service is officially announced, it will be framed as a way to
be reached by your email address from off campus, and will be initially
offered as more of a "plaything" than anything else. In the future,
integration with IM will be investigated.

Following this, Dennis discussed the recent survey of email list
participants which was conducted regarding the upcoming workshop.
Results were split between coinciding with the VON conference or the
Internet2 Member Meeting.  Most respondents expressed a desire for more
technology-based and hands-on type information, which can't be done at
VON. This will be done at the I2 Member Meeting in Washington, DC
instead. A meeting of some sort will still be done at VON, either a
published gathering or something less formal.

Following this, Dennis discusses a recent Cisco GSM codec problem. The
Cisco gateway accepts a SIP call the PSTN using GSM, but then fails
because the gateway doesn't actually support the GSM specified codec.
The workaround adds an inbound dial-peer to the outbound to reference a
codec list that doesn't include GSM.

The call ends with Candace asking about configuration questions about
testing Avaya CCS 2.1 with a Cisco 7960 phone. Currently server errors
occur when sending registration information. Follow up discussion will
occur on the email list.

The next call will be on Thursday, January 27th.