What are they and how do they differ?

Many phone systems today still use what are called Plain Old Telephone Service (POTS) or Primary Rate Interfaces (PRI). These are essentially cables specifically provided by your telephone service provider to run voice traffic from your organization to its destination. Below is a graphic illustrating this concept.

Session Initiation Protocol (SIP) is the modern alternative to PRI where the cables utilize a network connection to run voice traffic over the same lines that you use for anything else your organization does online. The graphic below provides an easy comparison between the two.

What benefits do SIP trunks offer over PRIs?

You’ll notice from the graphics that SIP eliminates the need for an entirely separate line from your organization out to your customer. By eliminating the extra hardware, organizations are able to save money on deployment and installation in addition to upkeep and maintenance. SIP is widely considered to be the more cost-effective option.

SIP Trunks also provide a level of flexibility that PRIs can’t compete with. With SIP you can completely customize your network to exactly fit your needs and do it at the click of a button. PRIs require on-site work to accomplish the same ends. Organizations that need to scale the amount of users or lines can especially benefit with a SIP solution for telephony.

SIP also allows for higher quality. PRI and POTS (Plain Old Telephone Service) lines both are limited by their bandwidth to what is referred to as narrowband calls. This term means their frequency range is limited from 300 Hz to about 3400 Hz. The human ear, as a comparison, can hear frequencies from about 20 Hz to 20,000 Hz. SIP, by comparison, allows for much greater frequency ranges and what is known as wideband calls, or HD voice. A modern conference phone from Polycom can cover a range from 160 Hz to 22,000 Hz, which covers and goes beyond what the average human ear can discern.

People utilizing HD voice for the first time often comment that it sounds like person on the other end of the phone is there in person with them – this is possible due to SIP’s drastically larger frequency range. Nearly all handset manufacturers now regularly include HD voice in even their cheapest models which makes it readily available to organizations that use SIP trunks for service. Do note that not all vendors or manufacturers may refer to wideband call voice quality as HD voice.

Here’s an example of how much HD voice can make a difference. First, listen to the following recording of a narrowband telephone call using a PRI with 3,000 Hz of bandwidth:

[sc_embed_player_template1 fileurl=”http://static.noctel.com/media/2011-Noctel-60-Ad1-3k.mp3″ volume=”100″]

When this makes your head hurt, try playing the next one, which is the same call, but with HD Voice at 22,000 Hz of bandwidth:

[sc_embed_player_template1 fileurl=”http://static.noctel.com/media/2011-Noctel-60-Ad1-22k.mp3″ volume=”100″]

Can you hear the difference? We thought so! The HD Voice call sounds much clearer and crisper than the old phone lines sound. Your customers will hear a big difference, too.

Are there drawbacks to SIP trunks over PRIs?

As with anything that is different, there are drawbacks to SIP when compared to PRIs. Since SIP utilize a network connection, it can be prone to connection quality issues associated with network interference, data packet loss, congestion, and flapping. Calls made over SIP trunks occur in real time, so when data transfer is slowed, bits and pieces of the conversation may get lost since the data frames that comprise the voice call are not all being transmitted or received at the expected rate.

Additionally, many organizations using SIP tie their service lines to their primary Internet connection. In this scenario, an organization may have peak points when Internet bandwidth is completely saturated with people using the web – many employees taking lunch at their desks for example. With network problems occurring, SIP trunks are liable to encounter the same problems but rather than high latency response or slowly loading web content, voice data through SIP would experience behaviors like echo or jitter, which can be very frustrating.

What can be done to mitigate potential issues with SIP?

SIP has been around for a number of years and has established guidelines to best practices which borrow strongly from good practices for network design. The simplest recommendation being to run multiple distinct ISPs into the organization to achieve service redundancy in the form of failover or load balancing.

Failover will prevent a total loss of network and voice operations if one of the provider ISPs experiences an unplanned outage. Load balancing will allow operations to between two or more ISPs to prevent over saturation and therefore end user performance degradation. Many organizations opt to implement both.

If additional ISPs are not an option, it is advisable to configure the network to give higher priority to voice traffic temporarily when quality degradation occurs. This generally will not cause noticeable impact for non-voice traffic as the voice data is very small by comparison. Presuming a bitrate of 100 kbps (kilobits per second) and a call duration of 5 minutes, the total amount of data for that call would be 3.66 MB, which on a fiber connection is near negligible. Alternatively, your SIP trunk service provider can configure inbound calls to route to cell phones temporarily. This results in no voice traffic on the network and slightly eases utilization under high load.