What can SIP do for your business?

Boost Network Traffic:

• Eliminates the need for protocol changes or conversions

• Provides seamless communications across multiple IP sites

• Squeezes more capacity out of T-1 circuits using SIP compression technology  (SIP does not increase bandwidth)

Simplicity

• Combines Voice and Data traffic on one single circuit and applies Dynamic Bandwidth Allocation (dynamically allows bandwidth for data to expand while bandwidth is not being used for voice. Voice has priority over data)

• Price based on bandwidth requirements/size, regardless of number of voice lines.

• Customizable call plans

Flexible

• Scalable Bandwidth (Access)allocation, usually from 1.5 Mbps to 45Mbps.

• Supports IP PBX functions and operations

• Customizable and optimizable through simple Web administration portal

Cost Effective

• Allows for "anywhere to anywhere" connectivity, eliminating the need for the expensive and equipment-heavy "Hub and Spoke" architecture.

• Transparent to a variety of different types of equipment, thus eliminating the cost of changing out of existing equipment.

• Network- based verses LAN Based VPN and management, thus optimizing the time and resources of the company's IT technician(s) and eliminating / reducing the need for customer supplied VPN equipemnt.

• Allows for the use of IP phone technology thus eliminating expensive TDM Voice PRI technology.

• Allows for VOIP voice compression which optimizes bandwidth usage and possible reduction of bandwidth requirements and cost.

•  Supports multiple application protocols, eliminating the need for additional compatibility compliance software.

What VoIP:

VoIP stands for "Voice over Internet Protocol.  It is simply converting and sending voice signals originating and terminating on IP phones (or devices which turn the analog phone signal into digital signals) over the Internet. 

Benefits of an IP phone system over an analog phone system:

• 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 and robust in creating 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, which is an Internet address (called "MAC" address). The analog phone is confined to a physical address where all the voice equipment resides.

• Because the IP Phone is Internet-based, it comes with a Web-portal (website) where the administrator can simply program their own phone by simply clicking on whatever features are desired.  Each Web Portal is dedicated to the individual administrator and is log-in / password protected.

• Unified messaging means that the call can be directed to the particular IP phone, to another member of the company, to a voice mail system and/or also to the email client in the form of a "wav" file File that can play the voice message over the computer.

• Call mamagement 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.

What is SIP or VOIP Compression:        

Calculating bandwidth consumption for VoIP
The bandwidth needed for VoIP transmission will depend on a few factors: the compression technology, packet overhead, network protocol used and whether silence suppression is used. This tip investigates the first three considerations. Silence suppression will be covered in a later tip.

There are two primary strategies for improving IP network performance for voice: Allocate more VoIP bandwidth (reduce utilization) or 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.722, proprietary)
• 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

Calculating the bandwidth for a VoIP call is not difficult once you know the method and the factors to include. The chart below, "Calculating one-way voice bandwidth," demonstrates the overhead calculation for 20 and 40 byte compressed voice (G.729) being transmitted over a Frame Relay WAN connection. Twenty bytes of G.729 compressed voice is equal to 20 ms of a word. Forty bytes of G.729 compressed voice is equal to 40 ms of a word.

The results of this method of calculation are contained in the next table, "Packet voice transmission requirements." The table demonstrates these points:

• Bandwidth requirements reduce with compression, G.711 vs. G.729.
• Bandwidth requirements reduce when longer packets are used, thereby reducing overhead.
• Even though the voice compression is an 8 to 1 ratio, the bandwidth reduction is about 3 or 4 to 1. The overhead negates some of the voice compression bandwidth savings.
• Compressing the RTP, UDP and IP headers (cRTP) is most valuable when the packet also carries compressed voice.

 

Packet voice transmission requirements (Bits per second per voice channel)
Codec	Voice bit rate	Sample time	Voice payload	Packets per second	Ethernet	PPP or Frame Relay RTP cRTP
G.711	64 Kbps	20 msec	160 bytes	50	87.2 Kbps	82.4 Kbps	68.0 Kbps
G.711	64 Kbps	30 msec	240 bytes	33.3	79.4 Kbps	76.2 Kbps	66.6 Kbps
G.711	64 Kbps	40 msec	320 bytes	25	75.6 Kbps	73.2 Kbps	66.0 Kbps
G.729A	8 Kbps	20 msec	20 bytes	50	31.2 Kbps	26.4 Kbps	12.0 Kbps
G.729A	8 Kbps	30 msec	30 bytes	33.3	23.4 Kbps	20.2 Kbps	10.7 Kbps
G.729A	8 Kbps	40 msec	40 bytes	25	19.6 Kbps	17.2 Kbps	10.0 Kbps
Note: RTP assumes 40-octets RTP/UDP/IP overhead per packet Compressed RTP (cRTP) assumes 4-octets RTP/UDP/IP overhead per packet Ethernet overhead adds 18-octets per packet PPP/Frame Relay overhead adds 6-octets per packet