6 Tips Before Putting Voice on MPLS

June 19, 2008 (Vol. 29, No. 13)

Don’t let the name fool you. The “real-time voice” class of service on your MPLS network isn’t necessarily the best way to transport your IP voice traffic. Consider sending it over the highest data class of service instead.

It’s a neat workaround that could save you from some engineering and sizing exercises, says David Lee, senior consultant at TechCaliber Consulting, in Washington, D.C.

The voice class of service will start dropping packets once the traffic exceeds the allotted bandwidth to prevent jitter (when packets arrive at different intervals). But the highest data class of service is “burstable,” so any traffic that exceeds the bandwidth allotted to that class won’t get dropped, Lee explains.

Several large enterprises “have had extremely good results” running voice on the highest data class, he reports.

Many businesses have moved to newer IP-based networks, running the gamut from MPLS to business Internet services with MPLS features (like AT&T’s Managed Internet Service with the additional MPLS Private Network Transport option). Some move because they need more bandwidth. Others are part of a forced migration away from legacy services like frame relay onto services like MPLS, for which discounts can be too good to pass up [VR 8/30/07, VR 10/2/06].

Don’t assume that just because you have a network with class of service you’re ready to start pumping through voice traffic. On the latest podcast episode of Telecom Junkies.  Lee and fellow TC2 senior consultant David Rohde offer five more tips to those looking to move TDM voice traffic to IP networks.

First, Send Voice as Packets

This one’s pretty obvious: You’ll need to “packetize” your voice traffic before it hits an MPLS port. You know this can be accomplished by an IP PBX, but it’s also possible with a legacy PBX, Lee instructs.

You need new WAN access routers to accommodate the additional demands of the CPU and the digitization of voice packets.

Choose Right VoIP Calling Package

News flash: While class-of-service capability is a prerequisite for running voice over an IP network, you’ll also need to buy a VoIP calling package from your service provider, Rohde says.

One such package is AT&T’s “Business VoIP” (BVoIP).

Your network service provider likely will offer different VoIP packages depending on the number of concurrent calls you expect your end users to make, Rohde says.

Fewer than 12 concurrent calls on AT&T’s BVoIP cost $15 per concurrent call, per site, per month, according to the carrier’s service guide. More than 12 concurrent calls costs $10 per concurrent call, per site, per month.

In a throwback to TDM, some packages include local and LD calls, while others include only LD or only local. Some contracts actually mention intraLATA toll rates, Rohde says.

Choose the wrong package and you could end up paying extra for on-net-to-off-net calls, even if you’re calling across the street, he warns. The cheapest plan might allow you to run calls only on net, for example. Off-net calls could incur fees steeper than running calls on the PSTN.

Don’t Let Traffic Engineering Skills Get Rusty

That’s why you’ll need to do some good old-fashioned traffic engineering.

Your contract might even stipulate that you measure your traffic in terms of concurrent calls – or how much capacity you’ll need in the busiest hour – the same analysis used in TDM environments, Rohde says.

You’ll need to know the number of trunks (based on the number of concurrent calls) to assign per site in order to achieve your ideal call-blockage rate. [See traffic engineering article] This calculation will determine how much bandwidth you need.

Expect your contract to specify that you’ll need at least a T-1, maybe a T-3, depending on the capability you’re running, Rohde says.

Compress Voice to Save Bandwidth

The 56-kilobit-per-second call that you’ve gotten used to in the TDM voice world will take up even more bandwidth in an IP environment, Lee says.

Expect an IP call to use up 80 kbps because of packet headers, says IP telephony expert Gary Audin, president of Delphi Inc. These headers carry the “shipping and handling” information for a packet, he explains.

To conserve bandwidth, WAN access routers compress voice using CODECs, Lee explains. “G.729” is the standard used by IP telephony equipment providers for inter-site communication, both on- and off-net.

This standard compresses the voice portion of a packet so an IP call only takes up 24 to 26 kbps, Audin says.

Expect the Usual Contractual Headaches

Your purchase of a VoIP calling package will mean adding an amendment to your service contract. Depending on the service, you could incur additional charges for setup or ongoing charges for running voice, Rohde says.