[Serusers] Question on gateway routing through UA proxy

John Todd jtodd at loligo.com
Sun Jan 19 20:41:53 CET 2003


Goal: To learn about ser in a "non-critical" and very small PBX-like 
environment so as to be able to understand nuances of the system in a 
production environment at a later date at various firms whose owners 
who have expressed to me a high degree of interest in SIP call 
routing for larger enterprise and CLEC implementations.

Sub-Goal: to make all calls into and out of my house routed via IP to 
alternate destinations based on ser routing configuration.  I have 
subscribed to a long-distance plan via "iconnecthere.com", I have a 
PRI gateway configured at a remote location (for calls into two area 
codes only) and my plan is to have a Cisco 2610 with FXO card for 
terminating my "local" phone line.  My house phones are all Cisco 
ATA-186 devices.  Based on called number, my calls will be sent to 
the iconnecthere.com SIP service, the Cisco PRI gateway, or the Cisco 
2610 single-line gateway.  Calls will also be routed appropriately 
based on number called on an inbound basis from any of those three 
gateway systems (PRI, 2610, or iconnecthere.com)

Progress:  I have phone-to-phone calling working well, and I have 
phone-to-PRI gateway calling working well in both directions.  I have 
not yet received the 2610, so I do not have the single-line analog 
gateway running, but I don't expect any issues with that system, as I 
understand the Cisco implementation of inbound/outbound VOIP sessions.

Problem: My "iconnecthere.com" account is a username/password 
protected account which (of course) requires a UA at my side of the 
connection.  To forward calls from the various phones in the house, I 
would need to have something re-write my username/password requests 
on the fly when they are sent out to the iconnecthere.com SIP 
proxies/gateways.  My first assumption is that I'll need a (sigh) 
B2BUA to act as a gateway, running (for convenience) on the same UNIX 
platform as ser.   After thinking about it for a while, I am 
uncertain if that is required, but at this point I can't determine 
what I need to do.


I would appreciate hints on:

   a) Wether I need a B2BUA at all, and if not, what config options 
should I be looking at in ser?

   b) If I do need a B2BUA, what would you recommend?  I'm using 
OpenBSD as my platform, and (personally) I'd like to stay away from 
Java for the moment.



Continuing discussion:
   I could see this as a fairly useful toolkit trick for a small 
business who wants to replace their phone switch with SER or ser-like 
systems.  If you've got an office with 10 people, it may make 
economic sense to simply get "generic" accounts with a SIP long 
distance gateway provider like iconnecthere.com (there are others) 
and allocate each of those accounts to individuals in the 
organization.  This is not a solution for a large-scale operation for 
the various reasons outlined in the 
http://www.iptel.org/info/trends/#b2bua texts, but the number of 
small-scale shops out there is very large and a simply understood 
package (and simply billed) is what would be desired for many places 
who still use under ~10 analog lines for outbound dialing from their 
PBX.  Any outbound calls to certain prefixes would always be pushed 
through a specific account for LD calling.
    Additionally, inbound calling through a similar service would have 
to also come in via the same mechanism, with a REGISTER request being 
sent by the B2BUA and all subsequent calls being routed through the 
SER proxy.

PS: I'd appreciate any open-source hints on how to get an ATA-186 
(v2.15) running behind NAT with ser on the "outside" of the NAT, 
without statically configuring the "NAT" address on the ATA-186 every 
time the outside address changes.  Lots of keyword matches found on 
Google, but very few clues to be scraped from the resulting documents 
as to "how" do it from the server side.

JT



More information about the sr-users mailing list