SIP in higher education

[John Todd <>]

> Though dual-mode cell and voip device convergence is nothing new, it 
> is a little hard to obtain, with many companies promising products 
> that are yet to be released or just for home consumer use, such as 
> Motorola's Residential Seamless Mobility Gateway for example [1]. 
> EQO has released a cell phone plugin "which runs on your computer, 
> connecting your Skype user account to your mobile phone" [2]  So a 
> call from your cell to a Skype buddy on your buddy list is delivered 
> over regular wireless phone services using SkypeOut (PSTN connected 
> Skype calls).  A bit too Rube Goldberg-esque [4] for my tastes, but 
> they got the job done.
>
> Interesting points from the CNET article [3]:
> * Instead of phone numbers, you use Skype buddy lists, downloaded to 
> your cell phone, to contact others
> * Your buddy list indicates user "presence" as a device capability, 
> ie whether you can call via Skype or cell
> * You can place privacy restrictions on your buddy list so you only 
> get calls that you really want
> * It's not supported on phones that don't run Java J2ME (like 
> Verizon wireless service, which uses BREW)
> * Eqo says that Skype is just the beginning, and they plan to 
> partner with LinkedIn, MySpace, and others
>
> So, are you ready to bite?
>
> Candace Holman
>
>
> [1] Roam where you want to!  New Motorola Technologies Break the 
> Barrier Between Home and Wireless Phone Services:
> http://www.motorola.com/mediacenter/news/detail/0,,6258_6213_23,00.html
>
> [2] Understanding EQO:
> http://www.eqo.com/download_how.html
>
> [3] Service lets Skype users chat on cell phones: 
> http://news.com.com/Service+lets+Skype+users+chat+on+cell+phones/2100-7352_3-6036663.html?tag=nefd.top
>
> [4] Rube Goldberg
> http://en.wikipedia.org/wiki/Rube_Goldberg


Apologies for mentioning another semi-vaporware item, but this seems 
interesting and I actually have it running (in pre-release mode.)

This is another Skype add-on, but it converts SIP to Skype.  It can 
work both ways, but I typically only use it as a one-way conversion 
in my PBX-like environments.  (SIP->Skype)  It allows me to assign a 
new SIP URI to users which actually is their Skype ID.

As an example:
  I have configured here at my workplace a "generic" Skype account 
that is named something like "tello-skype".  Then, I simply forward 
(via forking or sequential ringing) SIP calls to this extension, and 
they ring on the remote Skype account.  If I have configured my Skype 
Windows machine on 10.1.2.3, then to ring a Skype user called jhtodd 
I would dial sip: and the system would perform the 
remapping.  The Skype user "jhtodd" sees an inbound call from 
"tello-skype" every time I call that destination.  This is useful for 
treating Skype as a "trunk" to reach users who might have their Skype 
account running but are not using a SIP soft client.  An example of 
this in Asterisk dialplan with forking ringing might look like this:

; 2405 is John Todd's SIP phone, 2505 is his SIP softclient,
;  and "jhtodd" is his Skype username.
exten => 
2405,1,Dial(SIP/2405&SIP/2505&SIP/jhtodd@skype-gw1,40)

and in sip.conf:

[skype-gw1]
type=friend
host=dynamic
nat=no
secret=blahblahblah
canreinvite=no
disallow=all
allow=ulaw
allow=alaw


What you need:
  - a Windows machine
  - the Skype software for Windows
  - a "generic" account for Skype
  - the PSGW software with the "any Skype destination" patch(*)

Upside:
  - does not require dialing party to use Skype
  - does not require knowledge of Skype password information
  - does not require "opt-in" by Skype user population
     other than knowledge of Skype username for desired destination
  - works without modification to any SIP infrastructure systems

Downside:
  - client handles only one call at a time
  - does not relay presence information
  - audio-channel only
  - requires Windows
  - not a free product
  - audio quality is jittery sometimes (turn off "allow skype to 
modify audio settings" feature to minimize problems)

(*) This is not part of the release that is typically available on 
the website, but if you ask the author it may be available 
pre-release.  I've asked him when he's going to include this option 
in the general distribution, but he hasn't replied yet.  The cost of 
the software is $29.95 for a single user license.  The current 
shipping version of the software does static mappings, meaning that 
the SIP AOR is not converted into the Skype AOR, which is very 
restrictive and does not permit the SIP server to have the 
"intelligence" for routing.

Hints: The Windows firewall on the Skype host needs to be configured 
correctly for port 5060 to allow queries from your proxy but disallow 
from everywhere else.  Lots of codecs are supported, but I've only 
tried G.711 with any success or quality.  echo123 on Skype is a good 
test destination.  It could be possible to run multiple instances of 
Skype environments on a single CPU by implementing vmware machine and 
then cascade through them with some fairly simple Asterisk or SER 
dial rules.

Discussion: I am not a big fan of Skype, because of it's proprietary 
nature.  It's a great protocol, but it's closure to external 
connections makes it "unfriendly" to the goals that it seems are 
embraced by the SIP.edu community.  However, sometimes pragmatism 
overrides idealism, and we are forced to break out the tongue 
depressors, waffle irons, anvils, and duct tape and pay homage to Mr. 
Goldberg by making some contraption that connects two otherwise 
incompatible systems.

JT


--
John Todd
Networking, Tello Corp.

  +1-650-581-2405



--
John Todd
Networking, Tello Corp.

  +1-650-581-2405