How Not to Mess Up an IP PBX Installation.
Assuming a business quality broadband connection, the Internet will not be the problem. The phones won't likely be the culprit either. It won't even be the PBX appliance or gateway if we may say so. The majority of call quality problems will be created by your customer's local area network.
Properly engineered, VoIP is equal to or better than TDM voice. In fact, if bandwidth and system costs weren't an issue, everybody would be enjoying phone calls with CD quality audio. Rather, the best you can do is deliver a voice-quality pipe and advise your customer on the minimum LAN requirements to ensure consistent sound and negligible disruption.
The good news is that network analysis and preparedness is a billable service. To save yourself heartburn later, you should insist your customer evaluate and upgrade where needed before initiating VoIP. Network analysis and optimization tools get better every day. Whether you have the expertise in-house or subcontract, this pre-evaluation should be mandatory.
The LANs and Internet connections (WANs) used by most small to medium size businesses are simply not ready to handle VoIP. The basic firewalls commonly used for security and virus protection can and do break up VoIP calls.
The low cost routers from the local computer store don’t typically have the horsepower to drive quality VoIP calls. LANs can also become congested, especially when users are transferring large files across the internal network.
Even the best designed network will, from time to time, experience glitches and outages. The goal should be to minimize such problems and, to the extent that the project's budget allows, provide a solid and reliable VoIP experience. IP voice killers are latency, jitter and packet loss.
Latency is the length of time it takes for the speaker's words to be received by a listener at the other end of a phone connection, typically milliseconds. According to a white paper from Brooktrout Technology, latency starts to affect phone conversations when it exceeds 150 milliseconds each way.
It's unacceptable when it exceeds 450 milliseconds (nearly half a second). The company recommends engineering a VoIP system so that latency is always below 200 milliseconds.
When different packets reach the receiver at different times, the out-of-order packets reproduce as gobbledygook called jitter. If your customer experiences jitter, consider the network big picture. If there is a correlation between jitter and bandwidth usage (say during a full scale network backup or large data transfer), the problem may well be overall network usage. If there is no direct correlation, then jitter may be coming from congestion on the Internet, outside your LAN.
Packet loss happens when some packets never reach their destination for whatever reason. The problem can cause completely dropped calls. Related conditions include dropped packets (too much data arrives at the receiving server too quickly) and packet delay (data takes the long way around the Internet).
A network engineer can beat packet loss by separating voice and data traffic (using a virtual LAN or vLAN, a technique that creates independent logical networks within a physical network). Full duplex, non-blocking switches can help to avoid collision and packet loss as well.
Intelligent routing protocols, like MPLS (multi-protocol label switching) and OSPF (open shortest path first), prioritize network traffic. These protocols intelligently optimize the routing of network traffic in accordance with predetermined parameters.
In addition, know that there are network analysis tools that can identify and eliminate congestion points on the network. According to a VoIP network best practices list provided by Network Instruments ®, metrics to assess VoIP call quality include jitter, MOS, R-Factor, gap density, burst density, quality of service prioritization, and compression techniques.
Voice bandwidth is actually very thin, with the most common encoding schemes requiring about 100kbs per conversation at most. Compared to conservatively speaking, the 10Mbs connections used for desktop applications, voice represents 0.1 of a percent of a LAN's bandwidth.
The trick is to not allow too many conversations at once. The problem controls itself in traditional voice. There are a finite number of lines off the network. When you dial for an outside line and all of them are in use, you get a busy signal.
VoIP's limited bandwidth segments could be engineered to do the same thing. If your customer requires 60% of the available pipe for data, when voice reaches 40% of bandwidth, the system would refuse the call attempt until traffic fell below the threshold.
Most people don’t understand that just having a broadband connection is not enough. They actually need a high quality connection to deliver the call fineness needed to run a business.
A lot of smaller businesses connect to the Internet via consumer DSL or cable, most often with inexpensive modems. While such connections work fine for web browsing and email, they were never designed to handle VoIP transmission. Much less the combination of voice and data.
Likewise, many of the WANs now in use by ISPs were built before the advent of IP telephony. They weren’t originally designed to meet the demanding requirements of error-free, reliable VoIP transmission.
If your business is Internet service provisioning, and you haven't already done so, you'll need to make certain that you aren't oversubscribed and can give right-of-way to voice traffic. You may need to deploy additional QoS technology to ensure voice packet recognition and priority.
Since the VoIP gateway lies at the heart of VoIP QoS, it's important to employ gateways that support QoS applications. Low-end VoIP gateways generally provide only a limited number of QoS functions. Epygi's Quadro gateways support such traffic shaping algorithms as priority queuing, class-based queuing (CBQ) and token bucket with priority queuing. You may also want to add an adaptive dynamic jitter buffer to further optimize the performance of voice and fax traffic.
Steve Bieniek of the UK's VoIP Unlimited offers several tips for resellers deploying VoIP trunking services in the UK with some global application.
The ITSP or VSP (VoIP Service Provider) has to be solid as well. Network strength, proximity to the PSTN backbone and the ability to provide sufficient customer service are important matters.
It would be good policy to offer a service level agreement that protected customer and vendor. An SLA would guarantee basic performance benchmarks in exchange for the customer's commitment to spend a fixed amount with the provider over a time period.
VoIP service SLAs tend to be judged on their Mean Opinion Scores (MOS). They can include such factors as call completion rates and the length of time required for a user to hear a dial tone or to connect to the dialed party.
Various measurement techniques are used in association with SLAs, including active network tests made at regular time intervals as well as passive measurements that are based on actual calls placed across the network.
If money is no object (the dream part) and your customer must have the absolute best voice quality, you'll recommend he invest in two entirely separate LANs at his place of business and two entirely separate WANs (connections to the Internet).
The first WAN+LAN is dedicated to data and the second WAN+LAN to voice, ensuring no physical possibility that data packets can “stomp” voice packets.
Since cost is no obstacle when planning the voice WAN, opt for a T1 (North America and Japan) or E1 (everyone else) connection instead of a lower bandwidth DSL/cable connection. And for the two LANs, your customer will need two separate routers, each with their own physical wiring, which will terminate as two separate Ethernet jacks for every employee. They'll plug their computer into one jack and their IP phone into another.
Finally, when you have built the dedicated WAN+LAN combos, you'll hook them up with a primo ITSP connected to a private backbone. It's only money.
So your customer balks at the two LANs. She can still get exceptional quality by upgrading to an ISDN or E1/T1 WAN and insisting on a QoS Ethernet switch for her LAN. A QoS-capable switch (with IEEE 802.1P support, for example) will ensure that voice packets and data packets are prioritized properly.
When large files are moving across the LAN, the switch will make them pause momentarily to let voice traffic go first. Data users won’t even notice the pause, but voice quality will be significantly improved. Keep in mind the broadband provider must have QoS capabilities and the ITSP must use the same type of QoS employed on the LAN.
At the absolute minimum, your customer's broadband connection must ensure sufficient bandwidth for voice traffic from the business' premises to the Internet.
A typical VoIP call will use about 64KB of upstream and downstream. You should factor in at least 90KB for your first call due to what's called "IP overhead." Assuming peak concurrent line usage, the math looks like this: If your peak usage will be 5 concurrent calls, then you add 90KB + (4 x 64KB) = 346KB.
Be sure to remember that this is 346KB up and down. The standard broadband connection typically provides much more downstream than upstream. Be sure your customer has at a minimum "business-grade" DSL. Anything less and his disappointment is your grief.
First, ensure that the ITSP has a local public switched telephone network (PSTN) media gateway in your customer's area. This will shorten the path VoIP calls have to take over the Internet before they are converted to travel across the PSTN.
Second, the same broadband provider should serve the main office and all remote offices. VoIP performance improves when calls travel over a single provider’s backbone vs. having to "hop" across multiple backbones.