So, you are still using PRIs and individual analogue phone lines for connecting your business offices to the rest of the phones in the world (PSTN – Public Switched Telephone Network). And you hear more
and more that there is this new communication protocol called SIP (Session Initiation Protocol) that allows connecting IP phones and IP PBXs to the same PSTN.
Other than sounding cool, everything around us nowadays seems to run over IP and Internet, is there any benefit for YOU if you spend the time and money to understand how this works and make the switch?
The answer is YES. But, it depends on your current situation and the solution design chosen.
I will not go in to the details of how SIP trunking works, but I need to explain at high level what is the overall solution for SIP trunking connectivity, so you understand what are the components and the related cost.
Supposing your company has one head office, 3 remote sites equipped with IP phones registered to a centralized IP PBX (please see diagram). The voice traffic between the IP phones travels as follows:
- Between phones located in the same site, traffic stays on the local to the LAN
- Between the phones located in different sites, traffic goes over the WAN links (100Kbps per call, QoS recommended).
- Calls between the IP phones and the PSTN go to the local PSTN gateway that has the PSTN connectivity.
- The Head office has 1 PRI (23 concurrent calls) capacity, while the remote sites have each two analog lines for PSTN connectivity – inbound and outbound calls.
What does the transition to SIP trunking imply?
You order a SIP trunk with 23 concurrent channels only, supposing that this will be enough to support consolidated traffic PSTN for all 4 site offices. Your SIP provider will install the network connectivity between your head office and his PSTN SIP gateway. Then you transform your HQ PSTN gateway in a SIP compatible SESSION BORDER CONTROLLER (SBC), this will be your new centralized PSTN gateway. Then you remove the PSTN gateways from the remote sites and cancel the PSTN lines, since the call flow for inbound and outbound PSTN calls will go through the HQ SBC. There should be an increase of the WAN bandwidth QoS planned for these calls (see media flow in red for established PSTN calls to phones in the remote offices).
With this simple scenario, you get the idea that you can get economies of scale by centralizing all PSTN lines you have all over the place to a single “virtual pipe” connection to PSTN having IP networks support.
Here you a have a high-level list of possible migration and on-going costs.
- SIP trunk installation
- DID Number porting from PRI and analog lines to SIP-T
- SIP licenses for the SBC (or even new SBC acquisition and installation)
- IP PBX configuration for new routing patterns and PSTN gateway clean-up
- Cancel PRI and analog phone lines
- Cancel PSTN gateways support (exception is SBC)
- Add 23 concurrent SIP channels + DIDs
- Add QoS Bandwidth for voice traffic in HQ and remote sites for PSTN traffic
- Flexibility to scale UP or DOWN your PSTN channel capacity
- Flexibility and lower provisioning time in case you move your offices, need to take are of WAN connectivity only.
- Put all your eggs in one basket (if the HQ office goes down, you lose connectivity to PSTN). There are solutions to implement redundancy, IP networks are a wonderful support for these kind of solutions, but that will increase the price. A single SIP trunk can be terminated on two physical SBCs, with no cost per channel impact.
- Removing analog phone lines in the remote sites removes connectivity to the local 911 emergency service. If Enhanced 911 is required, additional solutions components need to be added on the SIP trunk and IP PBX, thus increasing the cost.
For a review of your particular case of SIP migration, please contact email@example.com