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VoIP Deployment 

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PCQ Bureau
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Deploying new applications and technologies on an enterprise network is always a challenge for the concerned managers, right from the technology officers to network administrators. They have to balance productivity with costs, and integrate it all with the existing infrastructure. Other issues that deserve priority are-troubleshooting, teething and compatibility problems during the initial stages, and performance tuning and maintenance later.

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Talking about all the new application and technology deployments in one article is not possible. So we have taken the example of deploying VoIP on a network. A lot of organizations, today, are considering converging voice and data on the same network, and, therefore, need to ensure that their existing networks can take on this additional load. For this, you'd have to analyze your existing network for congestion, and in case you're planning to use VoIP over your WAN links, then you'll have to work out the bandwidth requirements for it. This analysis will also vary depending upon the level of implementation you're planning. 

How much bandwidth?

This is the most commonly asked question on a VoIP network. That's because voice is much more sensitive to traffic congestion, on the network, than data. When implementing VoIP over a LAN, you obviously have much more bandwidth at your disposal. Therefore, you can configure VoIP devices to use a more bandwidth-intensive codec, such as the G.711, which consumes up to 87.2 kbps of NEB (Nominal Ethernet Bandwidth) in one direction. This will ensure better voice quality. In addition to this, you would also need to analyze the existing traffic patterns on your network. If it's already congested with a lot of broadcast traffic or other bandwidth-intensive applications, then the response time would be higher. This in turn will affect the voice quality of VoIP calls. You might face breaks or choppy voice due to this.

You can calculate precisely how much bandwidth a VoIP call can consume from various sites on the Net 

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While deploying VoIP on LANs, experts recommend that you create a separate VLAN (Virtual LANs) on your network for IP telephony. This will keep the voice and data networks separate, and anything happening on one will not affect the other. For this, you'll also have to ensure that you're using manageable switches on your network, which support VLANs. Having a separate VLAN will also ensure that your VoIP network remains unaffected from security threats that might occur on the data network. A DoS attack for instance, would affect a VoIP network much more adversely than a data network. 

The key issue in deploying VoIP over WAN links is that you have limited bandwidth. So you have to start by determining how many simultaneous voice calls you'd want to hold over your WAN links. Then determine how much bandwidth would be consumed by each voice call. For this, you have to take into account the compression technique to use, the payload size of voice packets, and the type of link used for VoIP communication. There are a lot of different codecs that can be used for VoIP communication, supporting bit rates that range from 5.3 Kbps to 64 Kbps. Using the codec data along with the payload size and type of link, it's possible to calculate the amount of bandwidth required per call. You'll also find a lot of bandwidth calculators available on the Net to help you with this job. One good link for this is www.voip-calculator.com. There are also several good software tools that can help you. Check out www.ixiacom.com, which has some specific tools to help you with VoIP deployment. Plus, they even have other tools to help you calculate the response time and throughput on your network. 

Ensuring Quality of Service 

Once you have calculated the bandwidth required for VoIP, you have to ensure you're your voice calls get that much bandwidth. This is when you have to enforce the QoS policies, else, you will get poor quality reception, delays, jitters, missed speech or even dropped calls. Typically toll quality calls require at least 16-20 kbps of bandwidth. QoS is normally controlled at the router level. Therefore, you'll need a router that let's you configure QoS for VoIP packets on your WAN links. 

You can also deploy switches that support QoS for VoIP on your LAN. There are also some
bandwidth-management solutions available that support QoS policies. 

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NetIQ's Qcheck is
a good tool to determine the response time and throughput between
two endpoints on your network. This can help determine how congested
your network is

Handling power problems

This one may come as a surprise, but yes, you will have to do some power planning while deploying VoIP. For instance, unlike normal phones, most IP phones require their own power supplies. So, in case of a power cut, they would stop functioning unless you've provided power backup for them. Calculate how much additional power your VoIP equipment requires, and check whether your existing backup equipment can take on the additional load. Many IP phones also support PoE, or Power over Ethernet, in which case, additional power backup may not be required. 

Security issues

While a regular PBX is not under security threats that normally occur on a data network. However, since VoIP deployment occurs on the same data network, it's also vulnerable to similar security threats. For instance, while a circuit-switched voice network requires a PBX switch, VoIP deployment requires servers that will act as the gatekeepers or voice gateways. These would run some standard software applications, which could be vulnerable to virus attacks, or other security threats.

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Therefore, these servers require the same type of care, as you'll give to other servers on the network. Subject them to the same routine maintenance, take regular backups, do security checks and install anti-virus software, etc. 

You would also need to implement VoIP performance analyzers such as 'Appmanger 6.0' from NetIQ to check for smooth functioning of
VoIP.

Technical issues in deploying VoIP

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Pre-requisites

Gunneswara Rao, Director -VoIP, D-Link India

The following are to be observed/recommended for good and smooth VoIP implementation at each branch/main office of any organization: 

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  • At each site LAN Network architecture should be such that at least 70% throughput is achievable within less than 10
    msec. 
  • Round the trip delay of each IP packet on LAN should be less than 10 msec.
  • Round the trip delay of each IP packet in LAN and WAN together (that is from any to location to any location with in the organization) should be recommended to be less than 150
    msec.
  • If the firewalls are present, engineering help needs to be planned for negotiating the VoIP-based packets through these firewalls. 

Integration with existing PBX systems

Existing PBXs may be integrated with the new generation VoIP functionality using available 2-port/4-port VoIP Station Gateways to divert voice traffic from PBX on the existing LAN/WAN instead of the present PSTN. For those organizations wanting to replace existing/old PBX to add new functionality and expansion may do away with any PBX to bring in the IP PBX. These are based on open standards like SIP or H.323.

VoIP Equipment and standards

For VoIP implementation, one may have to plan for the following equipment based on specific site needs:

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  • VoIP Station Gateways

  • LAN-based and Dial-up based IP phones.
  • VoIP Trunk Gateways to converge on to the existing PSTN networks.
  • Multimedia equipment/sound cards etc., on to PCs only if one wishes to extensively use the PC-based communication,
    may be in call centers.
  • IP PBX/SIP Server/Gatekeeper (H.323) or communication servers based on H.323+SIP protocols. For investment protection one needs to ensure that these are based on open standards like H.323, SIP, H.363/H.364 conferencing so that all future developments and benefits may be taken care with out redeploying such devices. 

Caution: Any propriety protocol based device deployment in this VoIP region shall cripple the organization during their future expansions.

Difference between VoIP and Data

Any voice communication has 60% silence + 40% useful data. VoIP technology understands this and proper measures have been included in open protocols like SIP or H.323. Plus voice communication is of burst nature while data communication is of study nature. 

Security

There are proven standards and methods available in VoIP technologies, which handle NAT traversal, firewall negotiation, STUN functionality, etc.

Gunneswara Rao, Director -VoIP, D-Link India

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