There are many ways to enable unified communications across your organization, there are hardware based solutions from companies like Cisco to software based solutions from the likes of Microsoft.

Office Communications Server 2007 or OCS 2007 is a venerable platform for providing communications connectivity, be it voice or video, keep in mind Exchange takes care of mailbox side of things for both voice, email and fax.

Currently, there are a limited number of devices available for direct connection to OCS, there’s even less number of gateways. So can OCS replace your phone system? Well it’s really not designed to do that, but with a gateway, it can and if you are already using softphones with your current phone system, you may be pleasantly surprised at the quality of the codec OCS uses, even on low bandwidth connections.

The beauty of OCS is the ability to embed presence information in items like documents, emails and spreadsheets. Imagine, you receiving a Spreadsheet of a colleague and immediately you can see in the Spreadsheet that your colleague is available via IM or voice, you can quickly and easily connect with the colleague and update the Spreadsheet. This resolves the issue of phone tag and allows for quick resolution and connectivity. Now, imagine you sent the Spreadsheet to a client and the client now has the same capability. I can see customer satisfaction soaring and everyone being able to get more done. Only catch is, the client will have to have OCS deployed on their side as well.

For an enterprise, an OCS deployment can entail 2, 3 or even more servers and gateways. For small deployments, all roles for OCS can be deployed on the same hardware and you can integrate it with systems like 3CX to provide the gateway capabilities. There is currently no documentation from 3CX about how to accomplish this, but if you understand the workings of OCS, you can integrate it and make it work. One company that is using OCS 2007 is Gibson Guitar.

Gibson uses Polycom CX700 IP phones, providing users with access to Exchange and Active Directory contact lists right from the phone! Very nice.

Lets face it, most small businesses won’t be implementing an elaborate voice system comprising of various components like Cisco Unified Communications Manager, Cisco Unified Communications Manager Express, and all the related hardware and additional software.

To ease the implementation for small business and lower the cost while providing more value, Cisco has come out with the UC500 appliance which does not require a Windows Server (it’s stand alone) and can be bundled with a POE Switch, WiFi Access Point and Voice in a nice, tightly integrated package.

It is inevitable, companies will be looking to integrate UC500 with their Exchange 2007 infrastructure. The good news is, UC500 can work with Exchange 2007 UM with some configuration. You have to make sure the UC500 can see the Exchange 2007 server, you have to allow h323 to h323, h323 to sip, sip to h323 and sip to sip. You have to setup dial-peer and define the IP of the Exchange 2007 server. Remember the codec is g711ulaw, and you should be good to go.

The actual configuration can require a little effort but is easliy doable, remember to apply Exchange 2007 SP1 before even trying!

Phone systems based on the venerable Asterisk are prolific in the market. Most companies boast about their phone system being based on Open Source software, but that’s as open as these companies are. When investing in a new phone system, you have to do due diligence if the phone company charges you a license per each device that connects to the system or not.

Most companies are clear and tell prospective customers exactly what they’ll have to pay to add additional nodes to their network, some companies, however, try to hide this fact and call their licenses a maintenance license or support contract.

Call this fee what ever you want, it’s still a license fee per device! Trying to hide this fact will just give you a bunch of angry customers. I’ve encountered many a pissed off customer due to this practice by a well known phone system provider, so when I talk to prospects I lay it out straight, call this fee whatever but in the end, you have to per for each device you want to connect to the phone system and you can’t just get an off the shelf SIP phone and get it connected.

There is a venerable PBX system available in Asterisk and its variants for *nix OS, this has even been ported to Windows and is called AsteriskWin32 that runs under CYGWIN.

If you are looking for a pure Windows native PBX system, that comes in freeware version is 3CX. There are plenty of Windows based high priced PBX software available on the market, not to mention, Cisco’s Call Manager. The freeware version and the cost of the paid version itself puts 3CX in a whole different category.

The 3CX system is in active development, supports almost any SIP device and will even connect to VOIP service providers along with connecting to PSTN via VOIP gateways. 3CX is not yet SIPConnect certified, for those of you who desire to connect it to PSTN without any interconnect but that functionality is being worked out as per their user forum.

The beauty of 3CX system is it comes with the PBX software and also a softphone that is easy to use and SIP based, call quality is just like a POTS line, no echo issues and configuration is simple.

The best feature of all…The management console is totally web based! Simply connect to the management console via any web browser and change any aspect of the 3CX configuration.

If you want to test an Asterisk based PBX, then trixbox is the system to get. trixbox now owned by Fonality, is a full fledged SIP based VoIP phone system. You can download a pre-configured VMWare image of trixbox 1.1 here http://www.vmware.com/appliances/directory/49.

trixbox Community Edition (CE) began in 2004 as the massively popular Open Source IP-PBX project named Asterisk@Home. Since then, it has grown into the world’s most popular distribution of Asterisk with over 65,000 downloads per month. trixbox CE is known for its flexibility to satisfy the needs of custom deployments and will continue to be FREE (as in beer and freedom).

Please don’t use these tools to break-in, hack, or otherwise disrupt your VoIP network. Here are some tools that may come in handy to load test, secure and verify your VoIP roll-out.

Asteroid

Asteroid is a SIP denial of service attack and can be found here http://www.infiltrated.net/asteroid/.

Callflow Sequence Diagram Generator

This collection of scripts uses your packet capture data and creates a callflow sequence diagram. http://callflow.sourceforge.net/.

PJSIP Open Source SIP Stack

PJSIP lets you create large number of calls to load test your system. http://www.pjsip.org/.

Nastysip

Nastysip generates bogus sip messages and sends them to any peer. Good tool for SIP load testing. http://phoenix.labri.fr/documentation/sip/Documentation/Material/Clients/Tools/
Test/NastySIP/SX%20Design.htm.

Distributed SIP Analyzer

Web based web analyzer that can collect/store information from multiple subnets and show results via browser. http://ant.comm.ccu.edu.tw/sip/.

SIP Proxy

SIP Proxy allows you to eavesdrop on and manipulate traffic in real-time. http://sourceforge.net/projects/sipproxy.

SIP Scenario Generator

Creates SIP call flows or scenario diagrams in HTML format from packet capture files. http://www.iptel.org/~sipsc/.

SIPP

SIP traffic generator. http://sipp.sourceforge.net/.

SIP Swiss Army knife

Allows admins to run tests on SIP devices. http://sipsak.org/.

VoIPong

Sniffs traffic and creates .wav files based on traffic. http://www.enderunix.org/voipong/.

Web Interface for SIP Trace

WISP captures traffic from remote hosts and shows live SIP messages to allow for easy debugging. http://www.devel-it.org/index.php?modulo=projetos&id=2.

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.

If the Voice over IP standards abbreviations and acronyms seem like the alphabet soup here’s a listing of protocols/standards and a short description that might help:

Signaling

H.323 H.323Megaco H.248  Gateway Control ProtocolMGCP  Media Gateway Control ProtocolRVP over IP Remote Voice Protocol Over IP SpecificationSAPv2 Session Announcement ProtocolSGCP Simple Gateway Control ProtocolSIP Session Initiation ProtocolSkinny Skinny Client Control Protocol (Cisco)

Media

DVB Digital Video BroadcastingH.261 Video stream for transport using the real-time transportH.263 Bitstream in the Real-time Transport ProtocolRTCP RTP Control protocolRTP Real-Time Transport

H.323 Protocols Suite

H.225 Covers narrow-band visual telephone servicesH.225 Annex GH.225EH.235 Security and authenticationH.323SETH.245 Negotiates channel usage and capabilitiesH.450.1 Series defines Supplementary Services for H.323H.450.2 Call Transfer supplementary service for H.323H.450.3 Call diversion supplementary service for H.323H.450.4 Call Hold supplementary serviceH.450.5 Call Park supplementary serviceH.450.6 Call Waiting supplementary serviceH.450.7 Message Waiting Indication supplementary serviceH.450.8 Calling Party Name Presentation supplementary serviceH.450.9 Completion of Calls to Busy Subscribers supplementary serviceH.450.10 Call Offer supplementary serviceH.450.11 Call Intrusion supplementary serviceH.450.12 ANF-CMN supplementary serviceRAS Manages registration, admission, statusT.38 IP-based fax service mapsT.125 Multipoint Communication Service Protocol (MCS)

SIP Protocols

MIMESDP Session Description Protocol

SIP Session Initiation Protocol

If you are not comfortable with Asterisk based IP PBX and you love working with Windows, there is a solution for you. 3CX phone system is a SIP based IP PBX for Windows with lots of promising features like voicemail via email, Unified Messaging (MS outlook integration) and call parking.

They have four different versions (Free Edition, Small Business Edition, Professional Edition and Enterprise Edition) to suit your needs, all of them support unlimited extensions (internal and external). The major differences between their free and commercial editions are that commercial editions support 8-32 numbers of Simultaneous Calls (depending on commercial editions) and other features like BLF status, call recording, show callers in queue and remote configuration of extensions. Commercial Editions also support G729 codecs to save bandwidth. 

3CX Phone System includes a compact VOIP Client which can be used in combination with a headset, as a fully functioning SIP software phone, or in combination with an SIP hardware phone. It works with all popular SIP soft and hard phones, SIP trunking providers and SIP VOIP Gateways like Grandstream, Patton (can be automatically configured within 3CX Phone System), Cisco, Linksys and others.

3CX Phone System has a built in VPN solution, which eliminates firewall reconfiguration and it also has an integrated Fax server that works with T.38 protocol .

I have been using their free version from last two years along with Trixbox CE and Pro at home with three SIP software phones (two internal extensions and one external extension) without any problem. 

3CX Phone System is well developed, easy to use and complete IP PBX application which challenges Asterisk based PBX systems.  For details visit their website