October 1999

How Do I Add  IP Telephony To An Existing Network?

BY LARRY BREAKWELL

If you’re a network or telephony manager anywhere in the world, likely you’ve started to feel pressure from sales people to get on the IP telephony bandwagon. Hey, everything’s going IP, so hook up some phones, pass them through a shared packet-based network, eliminate the circuit-switched leased lines to save money, and drop the call off at the final destination to bypass long distance fees. Or maybe you want to add new applications like H.323 gatekeepers and H.323 PC-clients, and voice enable your Web site. Cool. But wait. The actual implementation and choice of technology is fraught with pitfalls.

STEP 1:
Why would you want IP telephony?
The customer's network and application goals should determine what technology is used rather than the technology driving the customer. The availability of network infrastructures varies widely around the world from readily available low-cost high bandwidth ATM to just 19.2 Kbps dial-up links. In recognizing these differences, I believe in supporting the new ITU H.323 standards ("Multimedia over a packet network without guaranteed quality of service") and proven voice over frame relay or ATM, when it makes sense for a particular customer.

In general, a converged network offers a wealth of opportunities, such as:

• Enabling new applications such as voice-enabled Web pages.
• Better cost allocation by charging back long-distance and data charges to individual departments.
• Maximizing network efficiencies — reducing or eliminating the voice-only network.
• Cutting costs from voice and fax long-distance charges.
• Maximizing the reach of existing equipment and services (e.g., extending PBX functions from a central PBX to remote sites without requiring a separate PBX/key system for the remote site).
• Allowing new service providers to establish themselves at a fraction of the infrastructure cost of traditional Class 4/5 switches or PBXs.

STEP 2:
What is the existing network?
Have you conducted a Converged Voice and Data Network (CVDN) audit? A CVDN audit examines your existing network to determine what your network (voice network, routers, protocols, network capacity, available WAN infrastructure) can handle today (spare bandwidth, interfaces, signaling requirements, etc.), as well as what is required to handle any new applications. Is your spare bandwidth still available during the typical corporate peak period of 10 a.m. – 2:00 p.m? How many calls will your users be making and where will they be making them to? How long will those calls be?

STEP 3: DESIGN ANALYSIS
What do you need?
Armed with the results of your CVDN audit and your goals for implementing a VoIP solution, now it’s time to get down to the details of the new network design. This is where the abilities of your router/gateway vendor will become either enablers or inhibitors.

There are two basic architectures for adding voice to an existing network: a voice gateway/router feeds voice into the existing data router network, or a voice-enabled router performs both the gateway and router role.

Each architecture will have its own unique problems, and the severity of the impact depends on your existing network configuration. For example, if there is too much data feeding through the Ethernet link to the data/voice router, it might block the voice packets — increasing the delay and reducing the quality — even though that Ethernet link may be 10 or 100 Mbps.

Do you plan to completely replace all of your PSTN connections? If so, what happens in an emergency such as a power blackout, or if a 911 call must be made. Will you need to provide battery backup or does the router support failover to PSTN? If not, perhaps you should leave at least one line available per site to permit emergency calls.

What voice encoder will you use? 64Kbps for voice is an unnecessary price to pay in the packet voice world. New voice encoders provide excellent voice quality at a fraction of the bandwidth. This allows network designs to save bandwidth and squish more voice channels over the same bandwidth. Today, there are two generally accepted voice encoder standards: G.723.1 and G.729A.

Look for a vendor that offers several voice encoders so that you retain maximum design flexibility. Perhaps G.723.1 with 5.3Kbps sounds excellent in parts of your network, but you need G.729A in others because of congestion or existing equipment. Regardless, it is critical to test the voice quality in your actual network — not just a lab test — before making your final decision.

DTMF signals, like fax signals, don’t like to be compressed by low-bandwidth voice encoders. In the H.323 standards, DTMF tones are recognized at the gateway, demodulated, and sent as special signaling packets across the network. At the final edge gateway/router, these packets are converted, and true DTMF tones are played out. This allows for interoperability with voice mail servers or for capturing PIN codes for calling card applications.

Unfortunately, there is no fax standard called for in the H.323 standards. For now, fax requires both edge routers to be from the same vendor, although the core routers can be from any vendor. Over the next year we should see a consensus build in the industry as to how to support packet fax (i.e., T.37 or T.38 or some other) and how to integrate control and billing of fax into an H.323 gatekeeper-controlled VoIP network.

Delay and jitter (variations in delay) can “make or break” a voice over data network implementation. One-way delays from sender to receiver greater than 250ms will turn a voice conversation into something like a “walkie talkie” session. Therefore, delay from fixed sources (voice encoding, propagation, serialization) and variable sources (mixing voice and data streams, network congestion, queuing delays, route processing) must be minimized.

Serialization delays come about be-cause of the time it takes to transmit bits onto the local wire. Therefore, the vendor must provide a data segmentation scheme to allow interleaving of voice packets. A good rule of thumb is to set the maximum data size at twice the link speed.

Once the signal is on the wire or fiber, it will experience propagation delays, which are an important consideration for international networks. A signal traveling at the speed of light just across North America will experience about 25ms delay.

Quality of service (QoS) is not just about packet prioritization. QoS is everything that goes into ensuring a packet is transmitted and received across a network in an acceptable manner. Ideally, your vendor should be able to support both architecture modes and provide multiple QoS mechanisms to manage the mixing of multiple voice, data, and fax streams (see sidebar).

Some additional questions to consider in network design are:

• How many gateway/router voice interfaces are needed? What types? What kinds of signaling are needed? How many concurrent calls should my H.323 gatekeeper control?
• Are you connecting locations within a campus environment or connecting to the PSTN (this requires different interfaces and signaling, and certifications)?
• What WAN links will you use? A channelized T1 provides 24 concurrent call channels, but if you use a frame relay link, you can run 3-4 times that many voice channels (provided you have enough voice encoder digital signal processors).
• Will you use the same equipment in your network around the world (homologation issues, remote monitoring/control, software maintenance, learning curves)?

The answers to these questions delve into the realm of traffic engineering studies and are very application specific. With the average telephone call lasting approximately three minutes, 30 ports on one router may easily support hundreds of workers at peak busy periods. But those same ports feeding from a voice-enabled Web page into your call center may only support 30-60 call center stations. Don’t forget to count any H.323 PC-clients you’ve added to the network because they are going to consume extra bandwidth, especially if video conferencing is permitted. Again, your CVDN audit and engineering analysis is critical.

STEP 4: COST ANALYSIS
Are you getting what you want?
You must look at your alternatives to VoIP. Maybe the best design requires you to converge only a portion of your voice and data networks. Your network design must compensate for regional differences in link availability and quality — you may not be able to obtain the same high speed links everywhere in the world. Perhaps a voice over frame relay or ATM network makes better sense because of local carrier tariffs and your application needs.

Many times, salespeople will quote that you will get pennies-a-minute long distance if you converge your voice calling onto your data network. Unfortunately, this assumes that all your calls are on-net (i.e., between your own sites). Your CVDN must include a calling pattern analysis. For example, suppose you already receive 10 cents a minute or less in long-distance fees between North American sites, and this represents 50 percent of your traffic but only 20 percent of your cost. If this is the case, then maybe you should concentrate on reducing your international calling costs and look at ways of converting the other 80 percent of your costs into on-net calls.

You must also compare the existing and incremental new network costs. Don’t gloss over hidden costs. If you originally had a 56K dial-up (at say $50 per month) and you’ve replaced it with a fractional T1 or 256K FR link (at say $250 per month), your incremental cost is $200. But did you have to add a new router or interfaces or upgrade everyone’s PCs to enable H.323 PC-clients, or add ports on your PBX? Did you pay a penalty to break an existing contract from a service provider or add bandwidth to ensure QoS for your voice calls? What about support costs? Hardware and software can represent only a small fraction of the total cost of ownership.

CONCLUSION
IP telephony opens a world of opportunities, but you should walk, not run. Carefully examine what you want to achieve. If not properly thought out and implemented, hidden costs and other pitfalls can result in a waste of corporate dollars and in unhappy customers. Vendors should have many years of experience conducting CVDN audits and designs and should have the depth of equipment in their product line to support critical edge-of-the-network QoS mechanisms. And, most important, look for vendors who share your vision of the future and who are willing to work with you to achieve success.

Larry Breakwell is Vanguard product line manager, Voice and IP Applications, for the Motorola Internet and Networking Group. He can be reached for comment at [email protected]. The Motorola Internet and Networking Group (ING) delivers Smart Access Technology to connect people and organizations in a world of converging communications. For more information, please visit their Web site at www.motorola.com/ing.

Items in italics are critical for your IP telephony gateway vendor to support.

• Congestion control via bandwidth on demand.
• Dial-on-demand.
• Alternate link backup of primary link.
• Time-of-week routing for cost control.
• Automatic link restoration with data protection during transition.
• Data compression.
• A “voice aware” network that ensures voice over data prioritization.
• Traffic prioritization of selected data over other data.
• Multicast traffic.
• Voice jitter smoother buffers.
• Voice activity detection/ silence suppression.
• Queuing algorithms.
• IP RTP header compression.
• Echo cancellation.
• Voice compression (G.723, G.729A versus G.711).
• Fragmentation/segmentation schemes (ex. FRF111/12).
• MLPPP.