POTS Vs. SIP FAQs

“POTS” is the acronym for “Plain Old Telephone Service.” Traditionally it has been delivered to homes and businesses as a single pair of copper wires for each phone line. It’s a purely analog connection at the customer’s end, though nowadays POTS service is often delivered digitally to the subscriber’s neighborhood or to large office buildings. POTS may also be delivered digitally as “T-1” lines. T-1 service can be configured in several ways, but it’s often simply a digital replacement for POTS lines, and brings with it most of the same expense as POTS lines. POTS lines typically come from the Incumbent Local Exchange Carrier (ILEC) in your area. For many locations in the USA, this will be AT&T, Verizon or perhaps CenturyLink, Cincinnati Bell, Fairpoint, Sprint or Frontier. These ILECs have typically been slow to offer alternatives to their traditional, tariffed services. Moreover, ILECs’ traditional services often have myriad Federal, State and Local taxes tacked on. SIP/VoIP service is less regulated than POTS and may be taxed at lower rates. SIP/VoIP service is delivered by IP connections. These IP connections may be over the Public Internet, or may be delivered by a dedicated IP circuit (copper wires, cable, fiber, or even wireless microwave) to the customer’s premises.

From a cost perspective, SIP/VoIP services will nearly always be less expensive than POTS or T-1 service from ILECs. Depending on the exact circumstances and connections available, SIP/VoIP can typically save from 50 percent to 90 percent on monthly telecom expenses.

We tend to use “SIP” and “VoIP” interchangeably. In common parlance they both refer to communications over the Internet or other IP-based connection. SIP is “Session Initiation Protocol”, and is one of the protocols (the most popular one) used in VoIP communications.

The term VoIP stands for Voice Over Internet Protocol. It is a broad term that covers any phone calls made over the Internet, as opposed to traditional telephone lines, otherwise known as the PSTN (Public Switched Telephone Network). Other terms that are used interchangeably with VoIP include, IP telephony, Internet telephony, voice over broadband, broadband telephony, IP communications, and broadband phone service. These terms all describe the fact that the Internet is used to digitally transmit the voice signal to another telephone or endpoint.

Session Initiation Protocol (SIP) is a communications protocol that is widely used for managing multimedia communication sessions such as voice and video calls. SIP, therefore is one of the specific protocols that enables VoIP. It defines the messages that are sent between endpoints and it governs establishment, termination and other essential elements of a call. SIP can be used to transmit information between just two endpoints or many. In addition to voice, SIP can be used for video conferencing, instant messaging, media distribution and other applications. SIP has been developed and standardized under the auspices of the Internet Engineering Task Force (IETF).

One of the key differences in POTS and VoIP is management of the connections and the endpoints. With POTS, the entire end-to-end connection is managed by one or more Incumbent telephone companies (ILECs). With VoIP, the endpoints are typically managed by the customer via the customer’s Internet connection and Local Area Network (LAN). The ILECs operated “closed” systems for more than 100 years, and reliability was at or near the top of the priority list. “Ma Bell” managed everything from end-to-end, even owning the telephones themselves for most of that time.

VoIP service can be just as reliable, and in several ways better than POTS. However, the customer is typically responsible for the quality of the data connection that carries VoIP traffic. In earlier days of VoIP service there were growing pains and some technical issues, mostly due to improperly configured local networks. Nowadays there is much more expertise about real-time VoIP communications, plus today’s Internet routers and network switches can prioritize voice data and provide far more bandwidth than ever before. Moreover, VoIP service can be delivered to your business over a managed network connection. This implies every bit as much reliability as older POTS connections.

There are several different ways to acquire and use SIP/VoIP connections. Which one(s) will best serve in a particular scenario and at an attractive price? This question is best answered by those familiar with a variety of VoIP technologies as well as the broadcaster’s requirements and priorities.

It might be that a combination of dedicated SIP and cross-connecting to a business phone system will provide both monthly savings and high reliability. Other broadcasters may be fine using a “hosted” PBX and simply upgrading their business Internet service to bring the calls into the facility’s offices and studios.

The combinations of telecom service offerings and delivery methods, matrixed with differing on-air telecom requirements, reveals the necessity for a comprehensive analysis. Station engineering and IT personnel and outside telecom consultants should meet and share ideas for the best, cost-effective solution. Telos Support should be involved either from the beginning and certainly before any telecom or equipment orders are placed. Telos Support experts have worked with more than 400 broadcasters specifically on SIP/VoIP conversions.

We’ve produced a video which explains different methods to get SIP/VoIP telecom service to the broadcast facility. Here’s an outline of those methods::

SIP trunks via dedicated connection, often called “Managed SIP”. Provider is responsible for performance to the customer’s facility. (SIP trunks require an on-site SIP PBX, such as an Asterisk-based phone system.) Dedicated fiber installation. Dedicated SIP over DSL. Dedicated bandwidth over cable.

SIP trunks via a shared connection. Customer is responsible for providing an adequate and reliable Internet connection. (SIP trunks require an on-site SIP PBX, such as an Asterisk-based phone system.) Existing business Internet connection (cable, fiber, or DSL) Upgraded business Internet connection (subscribe to a higher bandwidth connection to accommodate SIP telephony) Additional, dedicated Internet connection (additional cable, fiber or DSL connection exclusively for SIP telephony)

SIP extensions from an outside, “cloud” SIP PBX. Customer pays for “extensions” from a remote SIP PBX. The provider manages the SIP PBX hardware. The customer may manage the configuration of the extensions. (No SIP PBX hardware is required on-site.) Existing business Internet connection (cable, fiber, or DSL) Upgraded business Internet connection (subscribe to a higher bandwidth connection to accommodate SIP telephony) Additional, dedicated Internet connection (additional cable, fiber, or DSL connection exclusively for SIP telephony)

A broadcaster already using an on-site SIP PBX for business purposes may arrange for additional SIP extensions from that hardware to serve a studio SIP talkshow system. Some considerations include the following: Will the business SIP PBX handle additional call volumes associated with on-air use? Can additional trunks be brought into the business SIP PBX for on-air use? Can such additional trunks be SIP, or would the have to be T1 or ISDN PRI service? Can additional SIP extensions be configured in the business SIP PBX without incurring additional SIP port licensing fees? (Some business PBX manufacturers charge one-time or annually for SIP port licensing.) Consider using two SIP PBXs for high reliability. Use a dedicated SIP PBX for on-air studios, but also connect some extensions from the business SIP PBX for backup or cross-functional use.

Use a SIP gateway as a temporary measure to convert between existing SIP and non-SIP telecom services. Telos does not recommend using a SIP gateway unless absolutely necessary. Some limitations inherent in converting technologies or formats usually result in a loss of audio quality or call reliability. Direct experience shows that going “all in” to a SIP/VoIP infrastructure is the best approach.

In practical terms the answers range from “Yes, a little better” to “OMG, that’s a phone call?” On the telecom side of the equation, VoIP phone calls sound the same as those digitally delivered by T1 or ISDN PRI. Any call traversing the Public Switched Telephone Network (PSTN) is typically using the G.711 codec, and this is true regardless of the digital delivery method.

However, Telos on-air phone systems have always squeezed the very best audio quality from phone lines and phone callers. The SIP-connected Telos VX (and Telos VX Prime) continue this tradition and even improve it further. Sophisticated audio processing - designed just for telephone calls - is built into every caller output. Telos VX users repeatedly tell us “Our calls have never sounded this good,” or “Callers are so clear that sometimes we do an ‘audio double-take’.”

There is another aspect to SIP/VoIP and better audio quality, however. The Telos VX and VX Prime systems are the first to allow SIP calls using G.722, the “HD Voice” audio codec. With G.722 the call’s audio range extends to 7 kHz. While not “high fidelity” by broadcast audio standards, the sound of “HD Voice” is astounding compared to regular G.711 phone calls. Because the PSTN doesn’t carry “HD Voice” calls yet, its practical use is limited to phones that are also connected to the same SIP PBX, or other SIP PBXs under the broadcaster’s control. This does allow for HD Voice remote broadcasts, news reports, traffic reports, and on-scene sports reporting, all from the convenience of a modern smartphone and a G.722 phone app. Station personnel can add a generic app to their smartphone, then use it to call-in from remote locations - directly into a SIP PBX and the Telos VX system - and go on-air with HD Voice audio quality.