Telephone systems for small & medium businesses

Summary

SIP or Session Initiation protocol is the ratified industry standard for VoIP communication. SIP is now used in Dial Tone delivery and is the base protocol for phones to pass voice communications over networks and the internet. SIP has many benefits that traditional dial tone does not provide. These include: Redundancy, Fail Over, Cost Savings, and Connectivity options.

What is SIP?

SIP stands for Session Initiation Protocol and is the industry standard for Voice over IP traffic. SIP was created in the late 1990’s then ratified as the standard VoIP language in the early 2000’s. In simple terms SIP is a universal VoIP “language” that has been established as the standard in the industry. This allows Carriers to deliver dial tone and Manufacturers to deliver equipment that can all talk to each other as it all speaks the same language.
SIP dial tone

Prior to SIP becoming the standard, PBX manufacturers used proprietary communication protocols. For example: Cisco created SCCP or “skinny” and ShoreTel used MGCP and 3Com used H3. These proprietary languages mean that only that specific manufactures equipment will work on their systems. You wouldn’t be able to get a Cisco using skinny to work on a ShoreTel using MGCP because they didn’t use the same language. Now that the SIP standard has been ratified, most equipment vendors are either creating IP Telephony platforms using the SIP language or they are re-designing existing platforms to conform to SIP standards. SIP based phones are typically less expensive as they can be used across many platforms in the market. Using the SIP VoIP standard also protects your investment long term as your equipment can be used on any SIP standard solution and thus your equipment will not become obsolete.
Carriers are also passing SIP phone lines or SIP Trunks directly to the consumer. For years, carriers have been passing traffic to each other using SIP protocol. Now they are offering SIP Trunks straight to the end user. If you have a VoIP Phone System which is SIP compliant you can plug the SIP/VoIP phone lines directly into your network.

What Is SIP/VoIP and is it Reliable?

Although Voice over IP phone lines are all basically the same, the way these lines are delivered can vary greatly and will make a huge impact on the quality of the call. There are two terms we must discuss to fully understand the different between levels of service. Prioritization and Quality of Service (QOS) are frequent terms used in IP Telephony. This simply means being able to control the quality of the call by marking voice data packets as more important or as a higher priority than other data packets. Typically you will find two methods of delivery.

Scenario 1 – Sending VoIP over your Internet Connection

Carriers who send your VoIP phone lines over the internet from the PSTN (Public Switch Telephone Network) and convert them to a digital signal at their locations. Then when you ask them for a phone line they send you a conversion box in which they send the voice call from their location to yours via VoIP and then convert it back to an analog signal at the conversion box which you then plug your traditional phone into. The problem comes as we look closer at the way traffic is sent through the internet.
SIP 2

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

Scenario 2 – Having the Carrier Provide Your Connection

Carriers are able to provide VoIP phone lines across dedicated connections that allow them to deliver Internet and Voice calls over a single connection because they can control the QOS and Prioritization across their network. The magic is in the Routers, Firewalls and Ethernet Switches located at each end of the connection. Each piece of equipment that the call passes through needs to support QOS and Prioritization. Think of it like a Toll Road. The voice traffic has a special pass which allow them to flow freely down the connection. The data traffic gets secondary rights to the road and the toll operator make sure the data traffic does not get in the way of the voice traffic. Giving us the ability to guarantee 100% voice clarity and quality.

Benefits of using SIP (VoIP) Phone Lines

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

Cost Savings

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

Redundancy

With Analog, PRI or T1 connections, if the phone line is cut the best you can do is ask your carrier to forward phone lines to a different number. This can often take several hours for them to do so. With SIP trunks you can have a primary route and a secondary route. For example: Your primary route can be over a T1 provided by the carrier, but if that connection is cut or goes out of service your carrier can automatically reroute your phone lines to another IP address or secondary Internet connection. Many clients bring in a second connection, usually with a different medium like wireless, or cable, and then calls can automatically reroute over these connections in the event of an outage. This is a powerful advantage.

Not Limited to Physical Location

Traditional phone numbers and phone lines are set to terminate in rate centers. For example an

801-486-XXXX might be a located in a rate center in Salt Lake City Utah. If a company had this number as a primary number and then moved its physical location to a different city, they would have to pay to forward calls from that phone number located in that rate center to another number in their new rate center. This is a called a market expansion line and is costly to maintain. With SIP or VoIP lines it does not matter where you are located, you can get local numbers to almost anywhere in the US to your business. So a company who is located in Florida that would like to provide their clients with a local number in California can now provide this service. An example of having remote numbers is Standard Plumbing Supply whose corporate office is located in Utah. They have 75+ locations across the western United States, all running off of one phone system located in their corporate headquarters. This is done by bringing all the phone lines in centrally to the main location and providing phones to all their remote offices. Quantities of scale allow you to reduce the number of total phone lines when centrally managed also saving money.

Reduction of Equipment

The majority of new systems on the market are SIP based phone systems. They would rather have IP or SIP dial tone brought directly into the network. If you choose to go with traditional dial tone like a PRI, a conversion box would need to take that PRI and change it into SIP. Thus needing additional equipment which cost money and more points of failure on your network.