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 is 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.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?

Here 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

Here are the "nuts and Bolts" of the COdec Calculations:

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

 

THE FOLLOWING IS BY COURTESY OF NUVOX COMMUNICATIONS:

Contact Bill Quaglia, 888-605-2630 for further information on, or to the get a quote from Nuvox.

 

What is SIP? Product Definition: SIP (Session Initiation Protocol) is a signaling service that allows for the set up of sessions (Calls and other communication methods). This signaling occurs over any type of traditional transport. (T-1, Ethernet, others) Premise based IPPBX systems have been using this technology for a number years on the station or customer premise side of the PBX in order to deliver enhanced voice, video and data applications such as private messaging, presence and conferencing tools.  With the introduction of SIP Trunks, Customers will be able to extend these types of applications to the Trunk or Public side of their networks. This will prove beneficial immediately in applications where customers have multiple locations and have a desire to manage remote sites and stations from the host site. SIP Trunks will also allow for a foundation for future enhancements as VoIP functionality continues to evolve.   NuVox’s Retail SIP Trunking is a product that offers enhanced features and functionality as delivered via NuVox’s Private, MPLS IP Network and as such, falls under the VoxIP product portfolio. Only customers in markets that have VoxIP will have access to the SIP Trunking product. NuVox’s Retail SIP Trunking Product is designed specifically for customers who have IPPBX systems. (Internet Protocol enabled or capable Premise based systems).  SIP Trunking provides these customers a way to combine their data (Internet access/WAN) and voice traffic into a single pipe and access the internet and PSTN through NuVox’s Secure, MPLS/QOS enabled Network.   SIP trunking allows the customer to terminate the trunk into an Ethernet port, often saving the customer hardware costs typically associated with PRI or T-1 cards.  Many IPPBX’s contain codec’s that allow for the compression of voice in order to improve efficiencies of transporting voice across PSTN and WAN networks.  NuVox will allow customers to subscribe to the compression feature for a flat rate designed to prevent cannibalization of transport revenue.   Product Attributes: Dynamic Bandwidth Allocation Silent Suppression Compression (optional, fee applies), supports G7.11,G7.23, G7.26, G7.29 QOS- Voice Prioritized over data T.38 for enhanced faxing capabilities Remote and Local DID 911 and Local Call routing based on DID physical address Caller ID   Network Diagram (Single-Site)

SIP AND MPLS ADVANTAGE THROUGH NUVOX:

SIP and MPLS - A Match Made in Heaven! VoxNet and SIP Trunking Application: Host/Remote site Deployments: In order to meet customer demands in a multi- site application, Nuvox will offer remote DID service in areas that are in NuVox Footprint. NuVox’s VoxNet Gold product should be purchased when connecting multiple sites. NuVox will improve processes to enable Local call routing and 911 routing down to the DID level. This service will only be available to SIP customers. This will allow customers to save money by not deploying multiple PBX’s at each remote site. This environment will allow the host site to manage their remote office/trunk side extensions in a manner that is equal to those that are station side at the main location. Depending on the particular PBX and its available features, this could include IP enabled communications solutions such as messaging, video, presence and other custom applications. Site to Site calling over VoxNet: Depending on the functionality of the Customer’s PBX and programming by the PBX Vendor customers may be able to make site to sit x digit extension dialing over the VoxNet Gold Connection. This is a function of the PBX, but not prohibited by NuVox VoxNet Product. Long Distance Long distance usage will be accumulated and billed at Host site. Customers have the option of subscribing to NuVox Long Distance Buckets at published rates under VoxIP tariff. Overage is billed at .053. Risk Mitigation The Host/remote site architecture has certain risks associated with deployment in the event that the customer and/or PBX vendor chooses to manipulate the out pulsed DID number to be different than the NuVox provided DID. In the event that NuVox does not recognize the DID, it will automatically default local and 911 routing to the physical trunk group on which the call was processed. Customers should be advised of these risks and take appropriate measures to ensure the safety of their employees. Mitigation language is included in the Tariff regarding these specifics. Remote Extension functionality: When deploying SIP trunks in a multi-site application, all features and access are provided by the IP PBX system at the host location. Sites are networked together via a VoxNet connection. Depending on the IPPBX type and architecture deployment, this will provide customers the ability to manage remote extensions in the same manner as those located at the host site. It also allows for the ability to implement IP enabled features across sites if purchased and enabled by the IPPBX. These features may include (but are not limited to) messaging, presence, and video conferencing. All of these features must be provided by the IPPBX. Redundancy: If the customer chooses to build a redundant network, a second IPPBX may be located in a data center. Co-location is an optional product which can be bundled with SIP as a solution for redundancy. Additional fees apply. This architecture is recommended for businesses that wish to assure minimum downtime for all locations.