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IP Telephony Across The Public Cloud

By Tony Rybczynski


Meeting IP telephony QoS, security, and reliability requirements across public packet networks requires special attention. While leased lines are always an option to interconnect sites, virtual private lines using frame relay, ATM, and increasingly IP VPNs (including MPLS or Multi Protocol Label Switching) and optical Ethernet offer better price/performance. What are your challenges in handling IP telephony across the WAN?

Engineering The Bandwidth
Unlike LANs, MAN and WAN bandwidth carries a monthly recurring cost. QoS allows the enterprise to use expensive WAN bandwidth most cost-effectively. Traditional voice engineering methods can be used to determine the number of voice and fax calls that need to be handled over the WAN link, factoring in communities of interest, the number of busy hour call attempts, and the average call holding times. The bandwidth required for voice and fax calls can then be calculated, dependent on the codecs used. Over highly-utilized high-speed links, up to 85 percent of the available bandwidth can be used for voice traffic. For low-bandwidth under (<1 Mbps) connections, no more than 50 percent of the available bandwidth should be used for voice traffic. This minimizes the maximum queuing delay that the VoIP traffic experiences.

In packet-based services such as frame relay, ATM, and optical Ethernet, tariffs are based on the access link speed and some form of committed rate: committed information rate (CIR) in frame relay, peak cell rate (PCR) in ATM, and committed access rate (CAR) in optical Ethernet. Adding IP telephony traffic results in the need to increase the committed rate of the link.

Flexible QoS Mapping At The WAN Edge
QoS is required to ensure that IP telephony traffic receives priority handling. Running IP telephony over leased lines leaves QoS and traffic management totally under the control of the enterprise. Support for QoS mapping when working into carrier packet services is another matter.

Optical Ethernet, generally available in metropolitan areas, provides native Ethernet connectivity with support for IEEE802.1p/Q Ethernet QoS. The high-speed, low-latency attributes of this service make it ideal for connectivity among metro sites. The CAR may need to be specified such that it supports the maximum number of simultaneous voice channels plus any data traffic.

Frame relay is a highly popular service available from 56Kbps to T3 rates, and even higher with ATM interworking. While frame relay QoS standards and products exist, service providers have not generally offered QoS-based services, though some publish statistical bounds on frame relay latency. Separate virtual circuits (VCs) with appropriate CIR should be established for IP telephony, to minimize interaction between voice and data traffic. ATM, on the other hand, is designed for multi-service transport, though it is relatively bandwidth-inefficient in supporting IP telephony; a voice stream coded in G.729 (8 Kbps coding) could take up over 80 Kbps across ATM. Voice (and optionally data) should be carried over appropriately sized VCs, either using constant bit rate [CBR] or real-time variable bit rate [rt-VBR] VCs.

QoS-enabled MPLS-based services offer a new option, which are starting to become generally available. These are well suited to IP telephony, and are generally premium priced over frame relay in hub and spoke configurations. In many cases, for reach or to work into the installed base, these will use leased line, frame relay, ATM, or Ethernet to reach your site, thus resulting in some complexities as already discussed.

With the increased availability of low-cost broadband access, running voice back to your site using IP VPNs over the Internet is very attractive for remote access and for connectivity to remote offices. Theres no need to map QoS at the WAN edge since there isnt any QoS in the Internet. The good news is that for the vast majority of the time, this will work very well, though you may want to have a contingency to handle critical calls when the Internet performance goes downhill, by using cell phones or the PSTN.

Reducing Delay Through Packet Fragmentation
Data packet fragmentation is another tool that should be used to minimize voice delay and jitter over bandwidth-limited (<1 Mbps) connections. For frame relay connections, the provider can use the FRF.12 standard; once fragmented, recombination only takes place at the remote site. ATM natively provides fragmentation, since all packets are fragmented into 53-byte ATM cells. Over leased lines, Point-to-Point Protocol (PPP) fragmentation allows higher-priority VoIP packets to interrupt and transmit ahead of the remainder of larger, lower-priority data packets that have already been queued, this process being done on a link-by-link basis. The fragmentation size is adjusted to achieve a maximum delay of 20 ms over the different connection speeds. The recommended fragmentation size is N times 128 bytes for a link speed of N times 64 Kbps (e.g., 512 bytes at 256 Kbps).

The Right Balance Between Price And Performance
When you run voice over the WAN, you need to pay careful attention to bandwidth engineering, QoS handling at the WAN edge, fragmentation over slow speed links and of course, price/performance. While leased lines, Optical Ethernet and QoS-enabled MPLS VPNs offer the simplest QoS handling at the WAN edge, tariffs and service availability may move you to leveraging frame relay, ATM and even the Internet, with their implications on complexity and performance. IP telephony systems have addressed these realities by including proactive voice quality management capabilities which help IT better manage IP telephony quality of experience running over the WAN technology you have chosen. IT

Tony Rybczynski is Director of Strategic Enterprise Technologies at Nortel. He has over 30 years experience in the application of packet network technology. For more information, please visit

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