What is VOIP?
VoIP stands for Voice over Internet Protocol. It is simply converting and sending voice signals originating and terminating on IP phones over the Internet. We recommend using “hosted” VoIP services — services delivered from a remote VOIP Phone system.
Benefits Of A Hosted IP Phone System Over An Analog System:
- Wide variety of phone options and features for a monthly fee per user that are not typically feasible in an “on-site” VOIP system.
- No “up-front” investment in equipment, annual licensing fees, or monthly maintenance fees
- Your software won’t become obsolete. Always experience the “latest and greatest” services.
- Ease of conversion to the system. Transparent and seamless transfer.
- System comes with a password-protected web portal, the click-to-change menus for phone, and call management options.
Why Choose VoIP?
- VoIP is driven by software programs manipulating digital signals rather than analog hardware equipment which has to regulate and modulate the analog signal.
- VoIP is more flexible. It can create unified messaging and call management features without lots of equipment or switches.
- VoIP gives the user the ability to call “anywhere from anywhere” because the phone has it own address on the Internet (“MAC” address). The analog phone is confined to a physical address where all the voice equipment resides.
- Many delivery options with unified messaging: calls can be directed to the particular IP phone, to another member of the company, to a voice mail system, or to the email client in the form of a “wav” audio file.
- Call management allows the administrator simple tools to track incoming and outgoing calls, who is making the calls and where they are calling. It also allows for simple report creation and call restrictions.
Calculating Bandwidth Consumption For VoIP
The bandwidth needed for VoIP transmission will depend on several factors: the compression technology, packet overhead, network protocol used, and whether silence suppression is used. There are two primary strategies for improving IP network performance for voice: 1) Allocate more VoIP bandwidth (reduce utilization) or 2) Implement QoS.
How much bandwidth to allocate depends on:
- Packet size for voice (10 to 320 bytes of digital voice)
- CODEC and compression technique (G.711, G.729, G.723.1, G.726)
- Header compression (RTP + UDP + IP), which is optional
- Layer 2 protocols, such as point-to-point protocol (PPP), Frame Relay and Ethernet
- Silence suppression/voice activity detection
How Many Phone Lines Can Be Compressed Through Each Codec?
The following is a standard being developed in the industry. Although some companies boast that they can deliver more phones than displayed below, call quality diminishes with every additional line. None of the established CODEC standards will cause packet loss, echoing, jitter, or dropped calls. This is usually casued by network strength and number of router “hops’ between server and customer.
CODEC | # OF LINES PER T-1 | RESULT IN QUALITY
G711 | 18 | Superior over TDM
G729 | 26 | Some degradation but superior over TDM
G726 | 32 | Quality of Cell Phone call
G723 | 40 | Quality of Cell Phone Call
SEDCOR