Become a member today

register/login

DrVoip.com

Member Login

Not a member yet? Sign Up!

Get immediate access to hot downloads such as DrVoIP VoIP planning guide(pdf) and more.

Register

ShoreTel SIP Trunks a Snap with Ingate SIParator!

December 11th, 2011
siperator

I don’t think your average business professional wakes up in the morning and says “Gee, I need some SIP”!   They do however ask questions like how do we get an Area Code from New York to appear on our System in San Diego?  Do I really have to buy 23 channels of voice on a PRI when I only use about 12 channels max?  Can’t we just “burst” up to 23 or 50 channels when we need them?   Can we have our phone calls re-routed to our branch office, if something happens to our main location?   These are market drivers that are encouraging businesses of all sizes to consider implementing SIP trunks.

What is a SIP trunk?  If you mention a T1 to a telephone guy,  he will think you are looking for a channelized voice path to the telephone company.  Ask an IT guy the same question and he will think you are talking about a 1.5MB connection to the internet!   Now bring an integrated T1 into the conversation and things get a bit more interesting.  Ask about a SIP trunk and you may get both of them scratching there heads.  If you are not working SIP trunk integration today, you most certainly will be doing so in the very near future!

To over simplify for purposes of discussion, a SIP trunk brings “dial tone” to your phone system over an Internet connection.    In most PBX systems, the connection to the phone company has been through a traditional TDM interface.  In a ShoreTel PBX, for example, you would come from the phone company provided “smart jack” to an SGT1K.  You would then use the Shoreware Director to configure the parameters that make a vanilla T1 a Primary Rate interface  like defining  the CO Switch type  (e.g. NI2, ESS5, DMS100), framing and signaling.

In a traditional TDM deployment, the separation of the PBX system from the Telephone Company is clear and unambiguous.  In fact, you often trouble shoot analog telephone  lines by going to the 66 punch down block, known in the industry as the “D Mark”  and remove the bridging clips that connect the phone company with the customers phone equipment.    In this way you can easily determine which side of the block has the problem.   With SIP there si a new challenge:  how do you separate the customer premise equipment from the telephone company provided network connection?    Where does your network end and the telephone companies network begin?

As we add SIP trunks to our network, the border of our network can easily disappear.   For this reason, we need to introduce a “enterprise edge border controller”!  The Border Controller is an essential element in provisioning a SIP trunk and is generally a dedicated appliance that provides a range of functionality that includes  NAT traversal, Security, Port management, Normalization, Call routing  and most importantly it acts as a “D mark” for your network, setting up a B2BUA between your iPBX and  your border controller; and between your border controller and your ITSP.    In this way you can think of your Border Controller as logical equivalent to a a 66 Block!

Why not use a Firewall to manage SP?   Clearly, if you are connecting your phone system to the Internet, there needs to be a “firewall” function.   A SIP RTP media stream is basically your phone conversation and it will take place over a 1000 different firewall ports.  Clearly nobody is going to open 1000 ports in a firewall or you might as well not have a firewall!  So a key functional requirement  for a  SIP trunk implementation  is the ability to track legitimate requests to open a port and then to close it when the session is over.   Firewalls that are “SIP capable” have this ability and are the minimum requirement for establishing SIP trunks on a phone system.

There are other equally critical functions that you must have in place for SIP trunking that exceed the ability of a Firewall and are more appropriately handled by a Border Controller.   “Normalization” for example, enables the appliance to provide language translation.    Like English or any other language, SIP sessions have  “dialects” and ShoreTel SIP might be different than SIP from Level3, Net Solutions or Paetec.    The Border Controller can mediate these difference enabling interoperability between these systems.

Two of the most widely deployed Border Controllers in the market are the CISCO CUBE  and the Ingate SIParator.  Both are excellent solutions, offering the required functionality to securely enable SIP trunking, including NAT traversal, Normalization and Security.    If you are integrating SIP Trunks with a ShoreTel iPBX, you will be very pleased with the Ingate “SIPParator” solution.  The Ingate solution  “Start-up” tool that is designed to get your up and running as quickly and as painlessly as possible.   If you  know the basic configuration parameters of your  ShoreTel and ITSP, the start up tool is the shortest possible path to your first SIP phone call from a ShoreTel iPBX.  Using the start-up tool you can quickly configure the basics, sanity test the basic configuration, upload it and then use  the Ingate  Web based Administration portal can then be used to further your configuration, logging,  reporting, monitoring and trouble shooting.    The SIParator  has been tested with a long list of SIP carriers and has many of the ShoreTel required parameters per-configured.    The support team at Ingate is both knowledgeable, patient and committed to making your ShoreTel deployment a success.

It is time to start integrating SIP solutions

2 Responses to “ShoreTel SIP Trunks a Snap with Ingate SIParator!”

  1. Tobi Tufnell says:

    Great information :)

  2. I just read the whole article & found that it is very useful regarding SIP trunking.

Leave a Reply

Training Videos

 

DrVoIP VoIP Network Readiness Assessment Checklist

Download Free DrVoIP VoIP Planning Guide

Ads

Small Business Voip Business VoIP Support Hosted Voip VoiP Video Library Contact us
Cost saving Annual voip support contract Hosted pbx Free voip videos Get a quote
Disaster recovery Pay as you go support Small office solutions ShoreTel Training My account
Business voip features Pay per incident Call center solutions Cisco training Make a payment
Google+ Installation service VoIP Glossary Sitemap Privacy Policy
© Copyright DrVoIP.com 2014 - site by: Vivid Software Solutions