Voice over IP (VOIP) Networking Solutions for Businesses in Silicon Valley and the entire Bay Area

Voice over IP is a protocol that allows for the transmission of voice signals through the Internet.  As part of a business’s communication infrastructure, VoIP telephony allows for many  value added enhancements not commonly available from the more traditional public switched telephone network (PSTN) PBX. Essentially, everything from accessing voicemail to registering a handset is done digitally through software management tools.  VoIP does for traditional plain old telephone system (POTS) as digital documents did for paper documents. 

• Monetary Savings
• One of the key drivers of combining voice and data networks is monetary savings. In North America, however, many large corporations pay 3 cents or less per minute for long-distance calls they make within the United States.
• Portability
• If you have an IP phone configured with DHCP, you can move the phone wherever you need and still keep the same phone number. This is similar to moving your laptop from office to office and still being able to log in to the same network server.
• Aggregated Information System (IS) department
• Only one Information Services (IS) department that supports both voice and data networks is necessary, as the networks are now one entity.
• Common Infrastructure tools
• VoIP enables voice mail systems to be put on standards-based platforms (such as PCs and UNIX machines). After a feature is on a standards-based platform, price gouging is much less likely to occur.
• Voicemails can be delivered through an email or a telephone

This process entails looking carefully at call quality, from one call to the maximum number of expected calls at peak network usage across a range of locations. If the call quality is okay and the other traffic is relatively unaffected, great – start VoIP deployment. If the call quality is not okay, determine what the problems are and where they’re located.

VOIP LAN Readiness Checklist:

• Is the Ethernet segment comprised of CAT5 or better cables?
• Are the Ethernet switches capable of VLANs?
• Will the Ethernet switches provide POE or is there ample space to add power injectors?
• Are there DNS servers accessible on this LAN segment?
• Does this segment have a DHCP server?
• Does the DHCP server have a large enough address pool to suppor the IP phone count?
• Is there documentation of the existing network (including VLAN info)?
• Is the power source for this segment backed up using a UPS, power generator, or both?
• Are switches centrally located and have ample backup power?
• Do switches and routers that touch this LAN segment support IP precedence and/or ToS?
• Has layer 3 routing policy been documented?
• Is the nominal packet loss on the Ethernet LAN currently below 1%?

VOIP Server devices/services Checklist:

• SoftPBX server
• Voicemail server (if separate)
• PSTN or TDM gateway (if separate)
• Analog endpoint gateway
• TFTP server (if used)
• Call accounting database server or syslog server
• Directory server (LDAP, Active Directory, etc.)
• DNS server
• DHCP Server
• Firewall, NAT, or Internet Gateway Router
• Default router

VOIP WAN readiness Checklist:

• Does the WAN backbone have less than 1% nominal packet loss?
• Is the WAN backbone capable of ToS and IP precedence?
• If this network's utilization will exceed 30% voice traffic, do all the WAN routers support RSVP?
• If this network's utilization will not exceed 30% voice traffic, do all the WAN routers support DiffServ?
• Do all possible call paths across the WAN have a round-trip latency of 150ms or less?
• Are all remote office links at least 128kbps
• Do all WAN connect points have backup power?
• If using frame-relay, is the typical maximum jitter on all PVCs less than 30ms?
• If VPN is to be used to connect the telecommuters to the VOIP network, do all telecommuters have broadband Internet access?
• If your WAN uses MPLS, are all the shims/zones documented?
• If using point-to-point T1s with legacy voice signaling, and planning to add VoIP to them on some of the DS0 channels, are the encoding and framing methods suitable for IP networking?

What influences poor call quality:

• Delay/latency
• Jitter
• Digital sampling
• Voice compression
• Echo
• Packet loss
• Voice activity detection
• Digital-to-analog conversion
• Tandem encoding
• Transport protocols
• Dial-plan design

Can a simple change in the VoIP configuration options, such as the choice of codec, improve the call quality sufficiently? What are the costs of making the required network improvements.

Choices include:

• Adding more bandwidth
• Upgrading or replacing your existing equipment
• Laying out your network architecture in an improved manner
• Reconfiguring or tuning the network for QoS
• OR a combination of the above

• Technology is built on internet protocol (IP) standards, making it mature, reliable, ubiquitous, and secure.
• Value added applications allow for features such as voicemails being emailed to as attachments (Voice-emails, Roaming, Software based calls, Archiving, Digital Recording)
• Generally, costs are significantly reduced due to lower infrastructure costs, phone call costs, and management overhead costs.
• Software updates can enable new capabilities and features in the future.

NetCal is experienced in setting up a complete VOIP infrastructure.  Our engineers can also troubleshoot VoIP related issues with audio quality involving:

• Network delay
• Jitter
• Packet loss
• Bursts
• Gaps
• Codec Selection

With the Cisco Smart Business Communications System, you can meet the growing communications needs of your small- and medium-sized business customers by offering them:

• An affordable, complete portfolio of Cisco Unifed Communications products that interoperate seamlessly to provide secure voice, video, mobility, and data networking.
• Access to the right mix of key communications, productivity, and business operations applications

This versatile solution supports desktop-mounted deployments of up to 16 voice users (as shown here) or rack-mounted deployments of up to 64 voice users. In addition, the system can support wired and wireless access for up to 250 users.

Here are a few of the most impressive components and features of the Cisco UC500 appliance:

• Cisco Unifed Communications 500 Series for Small Business (Cisco Unifed Communications 500 Series)

IP telephony solution that provides an integrated voice-messaging system and automated attendant, eight Power-over-Ethernet (PoE) ports for wired Cisco Unifed IP Phone and PC connections, and wireless connectivity through an integrated wireless access point option. Supports public switched telephone network (PSTN) trunks, IP trunks, and local analog devices. Enables data, security, and wireless services for up to ten teleworker sites (up to fve users per teleworker site).

• Cisco Catalyst Express 520 Switch

Enables the number of voice and data users to be increased by providing additional PoE ports for wired and wireless Cisco phones, Wireless Express Access Points, and PCs. The desktop model provides 8 additional PoE ports while the rack-mount models provide up to 24 additional PoE ports.

• Cisco Secure Router 500 Series

Provides advanced security features that include secure Virtual Private Network
(VPN) access and comprehensive threat defense with Cisco IOS Firewall, Intrusion Prevention Solution (IPS), and URL fltering. Also provides dynamic routing and advanced quality of service (QoS) features.

• Cisco Mobility Express Solution

Cisco 500 Series Wireless Express Access Points—Wireless 802.11g access points that are available in two modes of operation: standalone-mode and controller-mode Cisco 521 Wireless Access Points (AP521s). Controller-mode AP521s work only with the Cisco 526 Wireless Express Mobility Controller (WLC526).

Cisco 500 Series Wireless Express Mobility Controller—Controls and automatically confgures the controller-mode AP521s. Provides a platform for secure guest access, voice over WLAN, and optimized radio coverage.

The Digium AA50 appliance is a standalone Asterisk-based PBX targeted for small businesses (2-20 users), remote branch offices of larger organizations (2-20 users per site), and managed service providers for on-premise CPE-based solutions with SIP or IAX trunking. The AA50 also offers a hybrid solution alternative (a combination of VoIP applications using legacy telecom equipment) for customers who are not yet ready to migrate to a complete VoIP solution.

Here are a few of the most impressive features of the Digium Asterisk appliance:

• Complete Asterisk Server with AsteriskGUI
• Embedded Asterisk Business Edition with commercial license
• Built-in Router Ideal for Small Offices
• Up to Eight Analog Ports
• Support for a Combination of FXO and FXS Modules
• 1GB Compact Flash® Card
• Hardware-based Echo Cancellation
• 8 MB Onboard Flash
• 64 MB Onboard RAM
• 5 Ethernet Ports (4 LAN, 1 WAN)

NetCal can also provide you with larger PBX implementations using other PBX vendors, such as: Avaya, Nortel, Alcatel-Lucent, ShoreTel, 3com, and Talkswitch. Please contact us to find out the right solution for your phone system needs.