SIPTRUNK BLOG

Is Your Customer Ready for Cloud SIP Trunking?

Low cost and flexibility are the primary reasons that businesses are moving away from traditional PRI telephone lines to SIP trunking for their communications needs.  SIP trunking is an excellent choice for most of your customers, but some planning ahead is required and there are some infrastructure considerations that should be addressed.  Here are a few things to discuss with your clients to make sure they are ready to take full advantage of SIP.

How Many Concurrent Calls Do They Need to Support?

The number of expected concurrent calls is important in determining how many SIP channels your client will need, as well as the required Internet bandwidth.  Remember that the number of concurrent calls is usually far fewer than the number of employees or DID numbers. An average business has a ratio of about 3:1 or 4:1 employees to concurrent calls.  Meaning a 60 person company might have 20 concurrent calls.  Every business is different, so those with call centers, for example, or other high volume calling operations might have a 1.5:1 or a 2:1 ratio.  Each SIP channel supports one call, so you need to recommend enough SIP channels to ensure there is one available channel for each concurrent call during peak calling times.

Is the Client’s Internet Bandwidth Sufficient to Support Cloud SIP Trunking?

The amount of bandwidth necessary to support SIP trunking depends on the voice codec used by the SIP provider.  SIPTRUNK.com uses the G.711 voice codec which consumes 85kbps of bandwidth per call.  So we know, for example, that a DSL connection of 512kbps will support 6 simultaneous calls by dividing 512 by 85.  (Most broadband Internet connections these days are much faster and can support dozens and dozens of calls.)

You’ll want to base your calculations on the actual bandwidth at your client’s location, not on the capacity claims of the ISP, which can vary from actual results.  In order to measure Internet bandwidth, run a speed test from a computer that utilizes the Internet connection to be tested.  In order to get the most accurate measurement, it is best to run the test during off-peak hours when the Internet connection is not being utilized.

Is the PBX SIP-Enabled?

Your clients can utilize cloud SIP trunking whether or not their PBX is SIP enabled, but if it isn’t, they will need a small, inexpensive devise called an ATA.   Most new PBX systems are SIP enabled, so if your client has purchased theirs recently, it very likely is.  If it is older, there are several ways to check and see if it PBX is SIP-enabled.  First, if the PBX has a data jack or Ethernet jack on the back, it is probably SIP-capable.  Older PBX or key systems have only analog lines, so if the system does not have a data jack or Ethernet jack it is probably not SIP enabled.  If the PBX has a data jack and you are still unsure if it’s SIP-capable, check out the user manual.  If the system is SIP ready, there will be a section on ‘configuring a SIP Trunk’ or it will be mentioned in the specifications section, typically located at the end of the manual.  Look for words like SIP or SIP-enabled IP calling.

If the PBX is not SIP-enabled, your client can still take advantage of SIP with the use of an ATA (analog telephone adapter) that will convert SIP’s digital signal to analog.  The ATA acts as a gateway to the legacy PBX and allows the use of SIP trunks.  There are a variety of ATAs on the market and they come in several sizes, from single port all the way up to 24 analog ports.

Your client trusts you to make the transition to SIP trunking a smooth one. Addressing these important factors early in the conversion will help ensure that your customer is satisfied and your implementation goes off without a hitch.