Ironically, the topic I am talking with customers about more than any other over the last few months is the technology I’m somewhat reluctant to sell. It’s called SIP trunking, falls into the VoIP category, and it’s the phone line of the future.
SIP Trunking is the mechanism used to connect phone lines into an organization’s business telephone system, and is emerging as a viable alternative to legacy (TDM PRI and analog lines). A SIP trunk is essentially a telephone line, or series of telephone lines and DIDs brought into the office over a broadband (Internet) connection. The SIP trunk is a viable option, although the technology is relatively new and still not that well understood by the carriers.
Despite the fact that SIP is the telecommunications wave of the future, I quite honestly find selling SIP service to be a challenging exercise. While the LAN side of a VoIP deployment is easier to manage because we have control of the internal network, the WAN side consistently proves to be exceptionally difficult. The VoIP itself works great, but what causes us here at Digitcom so much grief is packet loss and latency issues, both of which need to be constantly and carefully managed.
For example, if an organization is connecting two offices together, QoS at the router’s edge is important, but the quality of data circuit connecting the two offices together is more important. There isn’t much point is prioritizing garbage, meaning it’s fine to prioritize packets on either side of the WAN, but if the data medium itself is of poor quality, then the voice will suffer.
So how do you guarantee voice quality?
You need a high quality data circuit connecting both or all sites, MPLS being the high water mark for data circuits.
In my book, Answering your Call, How to buy a phone system for your Business, I use the phrase “Garbage in = Garbage Out” to describe VoIP, and that holds true for running VoIP on the LAN, WAN, and SIP, which of course brings me to the point of this article: SIP, the phone line of the future.
SIP has a few amazing advantages: First, it is possible to have multiple area codes route to a single location. Let’s say, for example, that you want the area code 604, 514, and 613 all to route to your Toronto office phone system; with TDM (PRI and analog lines) this would be very difficult to deliver, while with SIP trunking it comes standard.
Second, and of equal importance, is SIP’s fail over features. That is, because the SIP phone line itself arrives on a data circuit there is much more flexibility with the routing of inbound calls, meaning that it’s very easy for SIP to fail over to an alternate location. Let’s say, for instance, that your Toronto phone system is down, the SIP provider will recognize the data circuit isn’t reaching its desired end point destination and it will automatically reroute the call path to an alternate data line.
The problem with SIP, and the issue I addressed at the beginning of this article, is that it is very difficult to deliver amazing sounding quality voice in a WAN environment, and there are very few carriers (I have come across one so far in Canada, two in the US) who actually know what they are doing with regards to delivering amazing quality sounding voice on a SIP trunk.
For our part, we here at Digitcom have tried selling both hosted VoIP and SIP trunking, most times resulting in frustration for everyone involved. This has led me to be increasingly sceptical when a customer informs me that they are going ahead with SIP and that the provider has promised them it will all work properly.
Truth be told, I get emails every week from SIP trunk suppliers or hosted VoIP providers, every one of them promising to be different from the last, but the fact remains that unfortunately both hosted VoIP and SIP trunking seem very difficult to do well. There are so many potential points of failure between the hosted VoIP provider’s backbone and the customer’s phone that any slight WAN or LAN degradation gets compounded, resulting in poor voice, and in turn, frustration.
The key to amazing voice on the WAN is that the provider needs to own both the data pipe (transport) and voice, and needs to host both at their own facilities. It’s not good enough if they provide that service on Bell’s T1 through a peering relationship at 151 Front Street. They need to own both.
At some point, and we’re getting there as the major carriers enter the SIP market, SIP will be the de facto standard. I suspect some time in the next two, maybe three years, Digitcom will be doing over 50% of our new installs using SIP. But that said, until the quality issue is sorted out and the providers realize they need to own both the transport and voice service, amazing voice on SIP will remain an elusive goal.