You’ve made an investment in IP (internet protocol) voice infrastructure that uses you data network. You are already enjoying lower maintenance costs, quick add/removal of voice nodes and lower infrastructure costs, it’s now time to start thinking outside the box.Since the same infrastructure now carries voice that used to carry data only, you can start thinking of how you can deploy voice devices in areas that were cost prohibitive to reach in the past. You can use WiFi or optical bridges to spread your reach, you can use the same infrastructure not only for voice but also for data and security.So once you have a location “lit” up with IP connectivity, you can use the same pipe for data, voice and devices like cameras for security and sensors for temperature/humidity, etc. The possibilities are endless; deploy paging speakers and horns that connect via WiFi, deploy IP cams or remote door locks for security or employee clock-in devices and enjoy the benefits and simplicity of putting other services over IP. It’s not just for data and voice anymore!

There are a lot of options available to small business as far as a VoIP phone system is concerned. A list of these options is the content for another blog, Cisco is working on penetrating this market with its Unified Communications for Small Business strategy.

Cisco’s UC500 series for small business is an all-in-one solution. This modular system offers a phone system, switch, wireless access point and firewall in an easy to use and manage package. The feature set as it related to VoIP includes:

• Eight IP phone station support
• Four trunks
• Optional T1/E1 voice interface (PRI and CAS)
• Integrated voicemail
• Automated attendant
• Basic call center capability
• Music on hold
• Optional wireless access
• System management

This system can be expanded as the business grows to 32 or 48 IP phones, additional IP ports can be added via Cisco Catalyst Express 520. Here are the specifications of the system:

 

Modular Support

WLAN Specifications

Voice Configuration Options

The Cisco Unified Communications 500 Series for Small Business is now available for order. If you need help with ordering or doing an initial assessment, contact a service provider like PC.Solutions.Net at http://www.pcsn.net.

To top it all off, Cisco SMARTnet can be added for enhanced support and technical assistance.

If you are tired of paying for high-end Cisco or Polycom phones, and I know some of you are, here’s a list of low cost commodity SIP phones and adapters. So, if you’re on a budget these sub $100 phones can help.

Aastra

9112i SIP Phone $89.99

Aastra 9112i Features

• Enhanced Call Management — Large storage for personal directory, callers log, and redial list
• Tight Integration — Support for multiple IP telephony systems including BroadWorks®, Nortel, Sylantro, and Asterisk SIP
• Remarkable Audio — Quality speakerphone with excellent voice delivery
• Protect Your Investments — Firmware upgrades can be downloaded and installed in the field as standards develop and protocols evolve

Aastra 9112i Specifications

Physical

• • 20.2 cm W x 19.2 cm D x 8.9 cm H (8.0”W x 7.6”D x 3.5”H)
  • 812 g (28.6 oz)
• Power
  • AC wall adapter included
• Handset / Headset
  • Modular RJ9 headset connector, compatible with amplified business headsets
  • Hearing aid compatible handset
  • Quality speakerphone
• Display
  • 3 line backlit display
• Feature Keys
  • 4 navigational keys
  • 2 programmable keys
  • 11 predefined hard keys including Callers log, Conference, Call Transfer, Redial, Options, Directory, Save, Delete, Speaker/Headset, Mute
• Networking
  • 10/100 Mbps Ethernet port
  • Manual or Dynamic Host Configuration Protocol (DHCP) IP address setup
  • Time and date synchronization using SNTP
  • Built-in HTTP server for web administration and maintenance
  • Server provisioned user configuration files
• Protocols / Codecs
  • IETF SIP (RFC3261)
  • G.711 ?-law / A-law
  • G.729
• Feature Highlights
  • Personal directory
  • Call forward
  • Call transfer
  • Call waiting
  • Caller and calling line information
  • Callers log
  • Conference
  • Redial list
  • Live dial pad or pre-dial support

D-Link

DVG-2001S SIP Adapter MSRP $59.99

D-Link DVG-2100S Features:

• (1) FXS Port
• (1) RJ-45 Network Port
• Enable VoIP Instantly
• Secured Remote Web-Based Access For Configuration
• Integrated QoS To Prevent Dropped Calls

Download D-Link DVG-2001S Product Datasheet The D-Link DVG-2001S VoIP Phone Adapter comes with one FXS port to connect to the existing analog telephone and one Fast Ethernet port to connect to the broadband router.

With a built-in secured provisioning feature, VoIP service providers can configure service settings such as a server address, CODEC and STUN settings via HTTPS/TFTP directly to the DVG-2001S.

DVG-2100S Specifications:

• Standards
  • TCP
  • UDP
  • ARP
  • HTTP
• Connection Port
  • RJ-11
  • FXS Port
  • RJ-45 Ethernet Port
• Ethernet Port
  • IEEE 802.3 for 10M Ethernet
  • IEEE 802.3u for 100M Ethernet
• Telephony Support
  • SIP Call Control Protocol
  • Supports Audio CODEC• G.711 (A-law and U-law)
  • G.723.1
  • G.726
  • G.729A
  • G.168 (Echo Cancellation)
  • DTMF Relay• G.711 (In Band)
  • RFC2833
• Device Management
  • TFTP Client
  • HTTP Web Interface
• Configuration/Management
  • DHCP (Dynamic Host Configuration Protocol) RFC2131
  • Embedded Web Server HTTP1.0 (RFC1945)
  • Auto-Provisioning Via Automated Centralized Configuration File
  • Configuration Restore/Backup
  • TELNET
  • TFTP Client
  • Performance Monitor DSP/Ethernet Statistics
• Quality of Service (QoS)
  • TOS-Type of Service Supports 3 Levels:
  • Normal
  • Signaling
  • RTP Packets
• Security
  • SIP Authentication with Password Encryption
  • HTTP Digest Authentication
  • Configuration Download Using HTTPS and SSL/TLS Clients Certification Encryption and Authentication
  • Encryption of Configuration File
  • VoIP NAT Traversal (SIP/STUN)
• Fax Support
  • FAX Relay
  • PCM (G.711)
• LEDs
  • Power ON/OFF
  • LAN Link & Activity
  • Phone ON/OFF Hook & Ringing
• Power
  • External AC Power Adapter
  • Output: 12V AC, 1.2A
• Temperature
  • Operating:0°C to 40°C
  • Storing: -10°C to 55°C
• Humidity
  • 5%-95% Non-Condensing
• Certifications
  • EMC: FCC Class B, VCCI Class B, CE Class B
  • UL/CUL
• Dimensions
  • 90mm x 82.46mm x 31mm (WxDxH)
• Warranty
  • 1 Year Limited Warranty

Grandstream

Budgetone GS-101 $44.99

Grandstream GS-101 Features:

• Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
• Support NAT traversal via STUN & symmetric RTP
• Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
• Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
• Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
• Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length
• Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
• Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
• Support DIGEST authentication and encryption using MD5 and MD5-sess.
• Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
• NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
• Support for fail-over SIP server and DNS server (pending)

The Grandstream GS-101 has one Ethernet port.

Bugdetone GS-102 $49.99

Grandstream GS-102 Features:

• Support SARP/RARP, ICMP, DNS, DHCP, NTP, TFTP protocols
• Support NAT traversal via STUN & symmetric RTP
• Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
• Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
• Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
• Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728. Dynamic negotiation of codec and voice payload length
• Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
• Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
• Support DIGEST authentication and encryption using MD5 and MD5-sess.
• Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
• NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
• Support for fail-over SIP server and DNS server (pending)

The Grandstream GS-102 has two Ethernet ports.

Budgetone GS-200 $64.95

Grandstream GS-200 Features:

• Support SIP, TCP/UDP/IP, RTP/RTCP, HTTP, ICMP, DNS, DHCP, NTP, TFTP protocols
• Support NAT traversal via STUN & symmetric RTP
• Interoperable with various 3rd party SIP end user device, Proxy/Registrar/Server, and gateway products
• Advanced Digital Signal Processing (DSP) to ensure superb hi-fidelity audio quality
• Advanced and patent pending adaptive jitter buffer control, packet delay & loss concealment technology
• Support popular vocoders including G.723.1 (5.3K/6.3K), G.729A/B, G.711 (a-law and u-law), G.726, G.728, and wide-band G.722 (Model 102D). Dynamic negotiation of codec and voice payload length
• Support standard voice features such as Caller ID Display or Block, Call Waiting, Hold, TransfForward, FLASH, in-band and out-of-band DTMF (RFC2833), Dial Plans, off-hook auto dial, configurable emergency dialing (e.g., 911), early dial, click-to-dial
• Support 3-way conferencing (Model 102D), full duplex hands-fredomain acoustic echo cancellation (pending), redial, call log, volume control, voice mail with indicator, downloadable ring tone (pending)
• Support Silence Suppression, VAD (Voice Activity Detection), CNG (Comfort Noise Generation), Line Echo Cancellation (G.168), and AGC (Automatic Gain Control)
• Support DIGEST authentication and encryption using MD5 and MD5-sess.
• Provide easy configuration thru manual operation (phone keypad and Web interface) or personalized automated provisioning via central configuration file for mass deployment.
• Support for Layer 2 (802.1Q VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS)
• NAT-friendly remote software upgrade capability (via tftp) even from behind firewalls/NATs.
• Support for fail-over SIP server and DNS server (pending)

Linksys

Sipura SPA-2002 SIP Adapter $59.99

Linksys SPA2002 Features:

• Terminating Impedance Agnostic - 8 Settings
• Call Waiting, Cancel Call Waiting
• Caller ID with Name / Number
• Caller ID Blocking
• Call Waiting Caller ID with Name / Number
• Call Forwarding: No Answer / Busy / All
• Do Not Disturb
• Call Transfer
• Three-Way Conference Calling with Local Mixing
• Message Waiting Indication - Visual and Tone Based
• Call Return
• Call Back on Busy
• Call Blocking with Toll Restriction
• Delayed Disconnect
• Distinctive Ringing
• Off-Hook Warning Tone
• Selective / Anonymous Call Rejection
• Hot Line and Warm Line Calling
• Speed Dialing of 8 Numbers / Addresses
• Music On Hold

SPA 901 SIP Phone $79.95

Linksys SPA901 Telephony Features

Zoom Telephonics

5801 SIP Adapter $59.99

Zoom 5801 Specifications:

• (1) FXS Loop-start interface with RJ-11
• (1)FXO analog interface with Teleport and RJ-11
• Up to 5 REN (Ringer Equivalence Number), supports more than 5 typical telephones
• Programmable ring patterns
• Call progress tones supported; Initial dial tone, Secondary dial tone, Stuttered dial tone, Message waiting dial tone, Call forward dial tone, Pre-Ringback dial tone, Ring back tone, Call waiting tone, Call holding tone, Call disconnect tone, Call conference tone, Busy tone, reorder tone, Off hook warning
• Power fail over
• Auto switch to PSTN for emergency calling using 911 and other programmed 3 digit codes
• FXS to FXO call bridging
• Supports SIPv2

Can be configured remotely using a TFTP or HTTP download from the service provider and updates to the firmware can be automatically delivered. Local configuration is done with a browser based interface.

An important aspect of Voice over Wi-Fi (VoWFi) is cell phone seamless connectivity over wi-fi networks. T-Mobile USA (part of Deutsche Telekom) has something to say as well: mobile phones don’t have to break the bank, when you can make calls over both cell towers and Wi-Fi routers. T-Mobile claims calls made via cell phone won’t be dropped as you switch between cellular and Wi-Fi networks. Homes with poor cell coverage can still make calls if Wi-Fi access is made available, I say about time!

This is the culmination of the company’s Hotspot @Home initiative, which is going nationwide after several months of trials in the Seattle area. With this service users can make unlimited calls by adding $10 a month, or $20 a month for up to five lines on a family plan, to an existing T-Mobile phone account. The “unlimited” part only includes the VoIP calls made via Wi-Fi, of course – except if the call started on the Wi-Fi side. If the call then roams to the cell network, it remains unmetered (maybe this aspect will change soon).

This may yet be the most important “break-through” since introduction of cellular phones.

I’m sure, in the beginning there will be a small number of phone manufacturers offering dual mode phones, but as this service catches on, I’m sure other providers will jump on the dual mode band wagon.

Users are not limited to Wi-Fi calls on their home networks. The phones will work with any open Wi-Fi connection. T-Mobile will also be selling home wireless routers from Linksys and D-Link that it says are optimized for the service by supporting the WMM standard, part of 802.11e with a proprietary method for setting up a secure link. The routers will be free after a mail-in rebate. T-Mobile apparently won’t guarantee call quality with other routers, nor at non-T-Mobile hotspots. T-Mobile hotspots all have full T-1 lines for backhaul to the Internet, while typical public hotspots with open access are likely only to have DSL or cable connections with no service level agreements (SLA).

The Hotspot @Home network is powered by Unlicensed Mobile Access technology, which enables seamless hand-off from Wi-Fi to cellular and back. This technology has been used by networks overseas by BT in the UK and Orange in France.

The iPhone being a hot product it is, while offering Wi-Fi, will not support this kind of hand-off, as its exclusive carrier, AT&T, doesn’t offer this service. Wi-Fi on the iPhone will likely be used for data only, unless Apple introduces a VoIP application in the future, or people use a third party soft phone software but then I don’t see how seamless switch over will work.

Think of Femtocell as a mini base station for cell phones. It can connect via Wi-Fi or Ethernet to service provider’s network and handle a few cell phones.

Unlike a repeater or booster, that some people use to boost their cell phone signal, a Femtocell provides an alternate route to connect to service provider’s network in areas where you might have internet but no cell phone coverage.

There are obvious small business and end user benefits, another use maybe in hospitals where only specific handsets are allowed to connect to serve as a link between staff while general public has no signal.

As Femtocell’s capture the market, it will be as simple as deploying a Wi-Fi access point to extend cell phone coverage in areas where there is none. As you have probably already deduced, a Femtocell deployment requires your service provider to support it. We’ll keep an eye on this technology as it develops.

After Voice over Internet Protocol (VoIP) the natural next step is Video over Internet Protocol. Video over Internet Protocol is not that big of a problem; unless you want to light up 20 thousand houses in a community then you need a huge internet pipe, but the problem with video streaming poses some challenges when pushed over a Wi-Fi connection.

The bottom line is, just as voice has requirements of pushing a lot of packets in sequence, video has the same requirement but at a larger scale and there are more packets and bigger packets. For a single dwelling, I’ve tested 1080i HD video streaming over 802.11g with success, but for a similar solution for an entire community over WiMAX, I’m not sure how well that will work.

AT&T’s U-Verse service broadcasts video over IP but there is no wireless component. I don’t know of anyone who is happily using the U-Verse service, but if you are, please chime in and give us an update.

Here is an article of interest from FAQ’s section on Wimax.com about hi definition video over WiMAX

If you’ve been a little confused about all the different WiFi standards, here’s a link that will sort it all out. The major standards you need to know about are as follows along with their salient features:

Source: http://en.wikipedia.org/wiki/IEEE_802.11 

Wateen Telecom has officially launched its WiMAX / HFC services in Pakistan and it is a major breakthrough in Pakistan’s broadband market. It is a joint venture of Wateen Telecom and Motorola, where Motorola played a major role in the deployment of first commercial 802.16 (d/e) network. Right now WiMAX service is available in 20 cities, which includes R.Y.Khan. It will provide cost effective alternative to existing broandband users. WiMAX is capable of supporting Triple Play (Internet, Voice and MultiMedia services) using single CPE (Consumer permises Equipment). For details visit their website 

VoFi stands for Voice over Wi-Fi. Whenever you use your VoIP connection over a wireless connection, you are using VoFI. Do you have people connecting to the network via Wi-Fi connection and then using a soft phone to make a call? Then you are using VoFi.

For a few people, a regular data Wi-Fi connection works fine, but when you are planning on rolling out VoFI for a large number of people, a shared data Wi-Fi connection will not work.

To successfully use a Wi-Fi connection you need to make sure your corporate wireless connection has high bandwidth to host voice connections, low lag times and fewer hops. You may need a higher number of access points if you plan on using voice over the connection compared to plain data.

You can get away with using VLANs on same access points, some companies go a step further and setup a totally separate network for voice. Normally if a company is using 802.11b/g for data, they might want to consider using 802.11a for voice since they use different frequencies and this minimizes any chance of interference.

The best thing you can do is to perform an extensive and comprehensive wireless survey before rolling out VoFI. Take care of all the problem zones and if at all possible have a separate Wi-Fi infrastructure for voice side. Using a 802.11a connection would be better for voice and/or video since this standard offers more bandwidth and is less interference prone compared to the 802.11b/g standard.

I’ve talked about potential pros of VoIP as it pertains to direct cost savings to the enterprise. There are some hidden benefits of VoIP as well, these don’t get the same attention or get measured like the other well known features.

Since VoIP works over the same infrastructure as your data traffic, this makes it very easy for VoIP systems to be integrated with your data systems. This is becoming more apparent as software like Microsoft’s Exchange 2007 gets released with direct unified communication hooks.

What this means to you is, you can easily integrate your voice and data to have one point of contact for your users. All email, faxes and voicemails come into the same universal inbox, this inbox gets backed up and is accessible via proprietary client (Outlook), web (OWA) or handheld.

Imagine the cost savings, your field and office personnel all use the same method of getting access to all forms of communication. Learning curves get flattened quickly, user productivity rises and users are happier while their frustration levels drop.

Most VoIP systems will offer such integration out of the box with little configuration. A big player, as I mentioned in an earlier article, is Microsoft. Microsoft’s strategy is to leave your phone system as it is and still enable VoIP and Unified Communications via software; Moto – VOIP as you are. Microsoft has already signed up vendor partners like Nortel, D-Link and a few others