SIP which is an acronym for Session Initiation Protocol is the ratified and widely accepted industry standard for VoIP communication today. SIP is now used in Dial Tone delivery by major carriers and is the base format for phones to pass voice communications over both local networks and the internet. SIP has many advantages that traditional dial tone sources do not provide. These include: Cost Savings, Added Redundancy, Fail Over Capability, and Connectivity options.


SIP is an abbreviation for Session Initiation Protocol and is the widely accepted industry standard for Voice over IP (VOIP) traffic. SIP was developed in the late 1990’s then became ratified as the standard VoIP protocol in the early 2000’s period. In basic terms SIP is a universal VoIP communication “language” that has been developed and agreed upon as the standard throughout the industry. This allows Carriers and others to deliver dial tone and for hardware Manufacturers to deliver equipment that can all communicate with each other.

Prior to SIP becoming the accepted standard, PBX and equipment manufacturers used proprietary communication protocols of several varieties. As examples: Cisco created its own SCCP or “skinny” while ShoreTel developed MGCP and 3Com created H3. These different and proprietary languages mean that only that specific manufactures software and equipment will work on their systems. You couldn’t get a Cisco using skinny to work on a ShoreTel device using MGCP because they didn’t communicate using the same language. Now that the SIP industry standard has been ratified, most telecom equipment vendors are either developing IP Telephony platforms using the SIP standard or they are re-designing their existing platforms to conform to the mainstream SIP standards. SIP based phones are generally less expensive as they can be used across many different platforms. Using the SIP VoIP standard also protects your telecom investment long term as your equipment can be used on any mainstream SIP standard solution and thus your equipment gear will have less chance of becoming obsolete.
Carriers are also now passing SIP phone lines and traffic or SIP Trunks directly to the end user consumer. Carriers have been passing traffic for years now to each other using SIP protocol, and in fact the dial tone in your business may get to the building as SIP even if your PBX uses T1 or Analog trunks. If you have a newer VoIP Phone System which is SIP compliant such as an SPX, you can plug the SIP/VoIP phone lines directly into your local area network.  If you need help finding the best source for SIP dial tone – try a company that offers multiple SIP options such as PhoneGuys.


Although most Voice over IP phone lines are all basically similar, the way these lines are delivered to your phone can vary greatly and will make a large impact on the quality of the call. There are two terms that we will review to fully understand the differences between various levels of service. Prioritization along with Quality of Service (QOS) are frequent terms used in discussing IP Telephony. These terms simply mean being able to control the call quality by marking voice data packets as being more important or as a higher priority on the network than other data packets. Typically you will find at least two methods of delivery.

Scenario 1 – Sending VoIP over your Broadband Internet Connection

This scenario uses the Public Internet to transmit voice traffic.  Carriers who transmit your VoIP phone lines over the internet from the PSTN (Public Switch Telephone Network) and then convert them to a digital signal at their locations.  Then when you request a phone line they send you a dialog box in which they send the voice call from their location over to yours via VoIP and then convert it back into an analog signal at the modem or conversion box which you then connect your traditional phone into. The problem develops as we look closer at the way voice traffic is sent through the internet.

The Internet is made up of numerous routers all over the world. There are millions or even billions of paths on which information flows from any one location to another. As you make a call using a VoIP phone line using Scenario 1 located in Salt Lake City it will be broken up into smaller pieces to be later reassembled at the destination. So a conversation like “Hey Bob, how was your work today?” will be broken down into little bits of information and then transmitted in digital packets over an IP network and then over the Internet. This all needs to occur in a matter of milliseconds in order for your call to sound clear and legible. If packets are lost, delayed for a few milliseconds or don’t arrive in a timely basis then your call will not be clear at all and will sound choppy, have static, be delayed or may even drop altogether.

Scenario 2 – Having the Carrier Provide Your SIP Connection

Carriers are able to provide VoIP phone lines across their own dedicated connections that allow them to reliably deliver Internet and Voice calls over a single connection – often a copper connection –  because they can control the QOS and Prioritization all the way across their network. The magic is in the Routers, switches and Firewalls located at each end of the path of the connection. Each piece of equipment that the phone calls passes through needs to support and manage QOS and Prioritization. Think of it like a controlled Toll Road. The voice traffic has a special “EZPass” which allow them to flow freely without stopping at the toll booth down the connection. The data traffic gets lower secondary rights to the road and the toll operator make sure the data traffic stops at the toll booth and does not get in the way of the voice traffic. This enables ability to guarantee closer to 100% voice clarity and quality.


There are several benefits for using VoIP phone lines over traditional lines.

Hard Cost Savings

The cost of a traditional POTS (Plain Old Telephone Service) dial tone lines is significantly higher than running a VoIP SIP based trunk. As your business expands and you need more phone lines, SIP allows you to ramp up additional services faster and much more cost effectively that traditional phone services.

Redundancy & Fail Over

With Analog, T1 or PRI connections, if the phone line is lost the best you can do is ask your carrier to forward all of your phone lines to a different number such as a cell phone. This can often take several hours for them accomplish this. With SIP trunks you can designate a primary route and a secondary call route.  Your primary route can be over an existing T1 provided by the carrier, but if that dial tone connection is cut or goes out of service for a time, your carrier can then automatically reroute your phone lines and incoming calls to another IP address or secondary broadband Internet connection – all without any noticeable interruption on either end! Many businesses bring in a second broadband connection, usually with a different circuit like fiber or cable, and then calls can be automatically rerouted over these secondary connections in the event of any type of outage. This is a very important feature in a business environment.

Not Limited to the Physical Location Where Lines Terminate

Traditional phone numbers and analog phone lines are set to terminate or end in rate centers. For example number such as 801-496-XXXX might be a located in a local rate center in Salt Lake City. If a business had this number as their main number and then moved its office location to a different city, they would have to pay on every call to forward calls from that phone number located in the Salt Lake rate center to a new number in their new office location. This is a known as a market expansion line and can be very costly to maintain. With SIP or VoIP lines it does not matter where your office is located, you can get local numbers with the prefix you want almost anywhere in the US made available to your business. So a company housed in Florida that would like to provide their customers with a local number in and area code in California can now provide this ability.  An example of having remote numbers is Elite Plumbing Supply whose corporate office is located in the Rockies.  They have numerous locations across the western half of the United States, all running off of one large phone system located in their main office corporate headquarters. This is done by bringing all the dial tone phone lines in to the main location and then providing remote phones to all their remote offices. And they can have local phone numbers for each area that they serve.  Economies of scale allow you to reduce the number of total phone lines when doing this centrally management which also saves money.   An SPX system is an example of a VoIP phone system that can do central management.

Reduced Amount of Equipment

The majority of new business phone systems on the market today are SIP based phone systems. Since when dealing with SIP dial tone coming into a SIP phone system there is no need for conversion boxes and modems to convert the PSTN SIP to Analog or T1/PRI prior to connecting to your PBX.   Therefore, there is less equipment and fewer points of failure.   Need help with all this – seek out an expert such as PhoneGuys to assist you.