Asterisk got its start when Mark Spencer needed a phone system for his company and the cheapest system available on the market was way too expensive. Already skilled in programming and application development, Mark created his own phone system that has now become the standard open source PBX that is being used at places like the city of Madera, California.

Cost savings are significant as there are no licensing fees with Asterisk, the city of Madera budgeted $400,000 for the phone system and ended spending only $140,000.

Asterisk runs on a variety of OS’s including, Linux, Solaris, NetBSD, FreeBSD and OpenBSD. Protocols supported are SIP, Inter-Asterisk Exchange (for authentication), Media Gateway Control and H.323.

If supporting open source software is not a risk you are willing to take, there are many vendors providing custom flavors of Asterisk with extensive support options like Aspect Software, CyberData, Escaux, Fonality, LumenVox and SimpleSignal.

The list of hardware supported on Asterisk keeps growing and there is no shortage of low cost FXS, FXO, PRI and other such ports, along with being successfully used with SIPConnect interface with a variety of dialtone providers.

You can manage only what you can measure. Managing a VoIP network requires some rigorous data collection and special analysis techniques.In addition to measuring latency, jitter and hop counts, it’s essential to calculate an overall voice quality score such as a Mean Opinion Score or R-factor score. 

VoIP monitoring tools calculate such scores by using a formula known as the E-Model. This is calculated based on statistics collected from the network and then taking into consideration factors such as codec compression, jitter, delay, packet loss and hop count. These factors all effect the user’s perception of call quality.

Since you have to take into account many different variables from networking side as well as traditional telecom industry, individually, such tools are not well suited for VoIP management and you need specialized VoIP tools such as the ones available from Agilent Technologies, Brix Networks, Empririx, Radcom and Qovia. These tools not only use MOS technology but also let you systematically gather data and in most cases relate to other network events and statistic to help diagnose problems and create SLA (service level agreement) reports.

Network load sharing and load balancing in most cases are both good things to have on a data network, but how does this effect your voice newtork?

Before rolling out VoIP on your highly redundant data network, you need to make sure your load sharing does not allow voice packets to take different routes to target and cause packets to arrive out of order. These out of order packets are discarded by VoIP systems and this sequence of events can cause high level of jitter and constant call quality issues. So if you have constant call quality issues, capture some voice packets to see if some device is trying to be too “intelligent” and trying to load balance/share to accomodate high bandwidth usage.

You’ve probably read several posts on here about Spam over IP Telephony or SPIT, here are some ways companies are putting into place to filter SPIT out:

1. VoIP Provider Filtering: Some VoIP providers, such as Vonage and Skype, can actually help protect you from SPIT. Calls through those providers travel, at least in part, through proprietary closed systems. These systems have existing defenses in place, which can help filter out a lot of the more obvious SPIT before it ever reaches your phone. Provider based security isn’t foolproof however, as hackers can and have invaded VoIP provider systems. Still, provider based filters can be a good first line of defense to SPIT and other threats to your VoIP.
2. Strong Authentication: This is probably the most important first step to filter out SPIT. By forcing users to be authenticated before allowing calls through, ideally, very little spam would ever reach you. Authentication programs work by using a “circle of trust.” If you make a call from provider A to provider B, in order for provider B to accept that call Provider A would have to authenticate that the call actually came from Provider A. The networks “trust” each other to validate users. These kinds of systems are hard to hack into - a hacker would have to steal a user’s identity or create a fake network identity to be able to override this security. If you are using VoIP for your business, these kind of authentication systems can and should be set up, not only to protect you from SPIT but from other threats as well.
3. Reputation Based Systems: A reputation based system works by assigning a score to users. The score is based on the history of the caller. For example, if you are being targeted by SPIT calls, the source of these calls can be flagged as bad and, going forth, calls from this source will be assigned a reputation based on this label which can be distributed across the entire network. While this system might be affective in some situations, it isn’t without its drawbacks. Generally, those wishing to distribute SPIT will use a number of different identities making keeping track of the “bad” calls difficult. Also, it’s possible that some calls you want to receive might get mislabeled as bad. A reputation based system can help you filter out some SPIT calls, but chances are some will still get through.
4. Central Black Lists: Another way to help filter out SPIT is to use a system similar to that used by your email spam blockers. This is a very simple system, but it can be effective. A list is created of all known SPIT numbers and those numbers are blocked by the system, just like you can block certain email addresses from going into your inbox. SPIT users will continue to change their numbers, but the list will grow and evolve along with those changes, becoming increasingly effective. The drawback to this, of course, is that calls not coming from blacklisted numbers will still get through, but when used in conjunction with another SPIT filtering method, blacklists can be very effective at limiting the number of SPIT calls that will get through.
5. VoIP SEAL: VoIP SEAL is a relatively new release from Japanese manufacturer NEC. VoIP SEAL is an automated system that is designed to protect your VoIP from any unsolicited calls. It works by employing a number of tests, which the system analyzes to give each call a “score” to measure the risk or danger. What is great about VoIP SEAL is that it works in a two-step process. Incoming calls that pass initial tests that determine they are not a threat are allowed to go through. Calls that still have a degree of suspiciousness to them are routed to specialized answering machine where additional tests can be performed. This two-part system can help reduce the amount of real callers that could erroneously be identified as SPIT. A system such as VOIP SEAL could provide a simple solution to filter your incoming calls.
6. Automated Challenge: One way to make sure that your calls are coming from actual humans instead of automated recordings is to set up a system that requires all calls coming from an unknown source to answer a simple question. Instead of routing the unknown call to you, the call will be answered by a recording asking the caller to enter a series of numbers or something along those lines. This may sound like an irritant to valid callers, but it ideally only needs to be done once. After the number has been judged as a human then the system would remember the caller and allow it go through. This can be an extremely effective way to manage incoming calls (aside from the minor irritation) but might be limited by the amount of database power required to know whether the caller is a new caller or is in the existing database.
7. VoIP Firewall: A firewall for your VoIP can be a great security investment. A VoIP firewall is an application driven by a security policy defining whether to allow or to deny certain calls. It manages and protects the traffic, flow and quality of VoIP and other SIP-related communications. Borderware has launched an SIP firewall called SIPassure to help mediate the threats that could potentially take down your VoIP system. One of the benefits of the firewall is that it filters an controls any SPIT that might be coming through to your phone. Since calls go through a system of authentication, it’s unlikely that much SPIT, or any other VoIP would get through.
8. Voice Recognition: Though it might sound like a strange way to determine if a call is SPIT or not, there is technology out there which uses the voice of the caller to determine whether or not the call is someone you want to speak with. V-Priorities from Microsoft can analyze the characteristics of a caller’s voice and their word usage to determine whether the person is a friend, family member, colleague or stranger; and route the call appropriately (the system was 90% accurate in tests). This is neat technology, but I can’t help but wonder what happens to those other ten percent of calls. There is potential to route an important business call to a junk call voicemail, which could be an annoyance. The technology will most likely improve over time, however, and can provide an innovative way to monitor your calls.
9. Calling Rate Limit: Another SPIT filtering technology you can employ involves calling rate limits. Eyeball has released AntiSPIT, a program that uses calling rate limits to keep out unwanted SPIT. The AntiSPIT engine employs a calling rate limit that is dynamically adjusted. Malicious calling behavior is identified and blocked but it does not interfere with legitimate calls as it uses an algorithm based on the caller-recipient history among other factors. This information is used to create a calling limit unique to each number. Once the calling limit is exceeded, further calls can be blocked, challenged, or forwarded to the recipient. The dynamic calling rate limit allows the server to add a rating tag to a call signal that indicates whether the call is good, suspicious or bad using green, yellow or red colors (or differing ring tones).
10. Secure Your VoIP: While programs aiming to target SPIT specifically are great, the simplest thing you can do to protect yourself is to make sure your VoIP is secure against more than SPIT. Employ a program to encrypt your VoIP conversations. Additionally, you should protect your servers and networking hardware with an IDS (intrusion detection system). Your VoIP provider can also affect the security of your calls, so go with a provider with the capability to handle most, if not all, Internet telephony security issues. Originally posted at voipnow.org

With a VoIP roll out, you can expect to provide your users with a seamless and unified inbox of all their messages, be they voice, email or fax. Phone systems on the market can be integrated with CRM systems relatively easily to enhance customer experience and increase call center productivity.

Now that voice and video has come into the data side of things, you have to consider increasing and better managing your storage systems.

Email discovery rules and regulations demand that you make sure you are archiving your messaging systems, not only for litigation but for the simple task of being able to play back a user’s order for confirmation. All this digitizing of analog communications and now video takes up huge amounts of space.

Consider modularizing your storage system so you can scale up quickly to meet demand. Invest in some good storage management software (a lot of times this may come bundled with your hardware), you might have to consider virtualizing storage and now, the basic building block of storage, the hard drive is evolving as well!

Solid state drives (SSD) are out now, and offer huge MTBF (mean time before failure) numbers along with using a fraction of the energy as traditional drives. Not too far into the future, you will probably see mostly solid state drives providing you with very fast access to data and at the same time using far less power and generating almost no heat. You’ve probably already seen laptops and ultra portables with these solid state drives boasting out of this world data through put rates and quick boot up times, the major benefit to such drives will definitely be in the data center, providing quick access to your voice, video, emails, files and databases while using up much less power and requiring a fraction of energy used for cooling.

Microsoft’s Exchange 2007 has been out for some time now and recently MS released Service Pack 1 for the venerable messaging system.

As you probably already know, Exchange 2007 requires Windows 2003 64BIT and will not run on 32BIT OS. You’ve probably also heard about Exchange’s ability to provide Unified Communications out of the box.

Exchange 2007 really takes UC to the next level. Not only can you get all your voicemails, faxes and emails in the same mailbox but you can also get your emails over a phone and Exchange 2007 can read you your emails!

Such a system, especially in the enterprise space, requires robust and tested hardware to accompany the software. Here’s a list of hardware that has been tested with Exchange 2007 system:

VOIP Gateways 

Integrating Exchange 2007 UM with TDM-based PBXs requires the use of VoIP gateway(s) to translate the media and signaling data between circuit-switched protocol formats (understood by TDM-based PBXs) and IP-based, packet-switched formats (understood by Exchange 2007 UM). Currently, there are two vendors and several models of VoIP gateways that have been tested and are supported for Exchange 2007 UM, including:

PBX Systems

The following PBXs are supported through AudioCodes gateways (MediaPack-114 FXO, MediaPack-118 FXO and Mediant 2000)

The following PBXs are supported through the low density Dialogic® Media Gateway (DMG1000). When an analog DMG1000 is used, supplemental signaling (RS232 SMDI, MD110 or MCI protocols, or Inband DTMF signaling) is required.

The following PBXs are supported on the T1/E1 Dialogic® Media Gateway (DMG2000). The gateway, which comes in single span (DMG2030DTIQ), dual span (DMG2060DTIQ) or quad span (DMG2120DTIQ) densities, supports the following protocols:

If CAS signaling is used, supplemental signaling (RS232 SMDI, MD110 or MCI protocols, or Inband DTMF signaling) is required. If QSIG signaling is used, the PBX must support the supplemental services associated with Call Party Information and the Call Transfer capabilities required by Exchange 2007 UM.

IP PBX

The following PBXs are supported through direct SIP connection with Exchange Server 2007 UM.

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