VoIP: is your network ready? There is plenty to consider, including the cabling plant, when switching to IP telephony

VoIP: is your network ready? There is plenty to consider, including the cabling plant, when switching to IP telephony – Voice Networks

Carrie Higbie

Matching carrier-grade voice quality is a unique challenge to the voice over IP (VoIP) services being introduced in today’s market. With everything moving toward IP, can this medium be used to actually create a carrier-grade service?

A true carrier-grade service includes reliability, availability and scalability; in a VoIP environment, add in security, manageability and interoperability. The additional factors are necessary as the voice transmissions are carried over network links in conjunction with data services. Voice-grade services must have the continuous available capacity from the beginning to the end of the conversation. Latency, loss and jitter can all affect the quality of the voice service.

In a typical IP packet carrying data only, transmission through the network is fairly simple. The data packet is formed, sent and received. Small delays or retransmissions are accepted. Voice traffic, however, cannot tolerate errors. Voice packets need a mechanism to move at a higher priority so that their transmission is, in effect, guaranteed. Today, this is accomplished by setting the quality-of-service (QoS) bit in the IP header.

All IP headers have a type-of-service (ToS) byte, more recently defined as the diffServe code point field. QoS refers to a set of parameters for both connection-mode (TCP) and connectionless-mode (IP) transmissions, which provide for performance in terms of transmission quality and availability of service. It encompasses maximum delay, throughput and priority of the packets being transmitted.

In IP transmissions, there is no actual circuit-the packets are routed over a network to the receiving workstation. If the call is to be placed outside of the network, the conversation will still be routed to the point that the PBX or gateway puts the call on the packet-switched telephone network (PSTN).


To build QoS into a voice system, first address construction of the framework. Applications and appliances must be able to set and understand the QoS bit, or recognize the other transfer mechanisms if using another standard.

Applications called traffic shapers work much like a traffic cop. They will slow “less important” traffic and allow priority movement of datagrams that have a higher priority. The problem with these applications, however, is that they offer a single point of failure in a network. They also may have issues with scalability for networks over a certain number of devices.

Routers and Layer 2 switches that understand this bit will provide both redundancy and, in the near future, multicast abilities. In Layer 2-type switches, IEEE has developed two standards (802.1p and 802.1q) to address QoS.

A Layer 2 802.1p-compliant switch has the ability within the MAC layer to group LAN packets according to their traffic class. Of the eight classes that network managers map to their specific applications, priority seven is generally used for router-to-router communication and path information within the network. VoIP, videoconferencing and other delay-sensitive applications will use values five or six. These switches also must understand multicast traffic.

LAN administrators must determine the settings of these bits in their applications. With new Layer 3 switch and routing capabilities, this task can be done on an address level within the router tables, reducing or eliminating the need for traffic shapers and complicated mapping tables. The 802.1 q standard allows network administrators to break up larger LANs into smaller segments, or virtual local area networks.

Another means of identifying QoS is through setting the diffServe bit. The first bits of the ToS byte or diffServe code point field in the IP header are set using one of three per-hop behavior mechanisms. A hop is defined as packet movement from one forwarding point to another (router to router, router to switch).

The advantage of this bit is that all Layer 3 devices throughout the Internet, including routers and Layer 3 switches, understand it. It will allow traffic to travel via various paths until it reaches its destination. All routers and switches within the given paths must adhere to this setting and forward the packets accordingly.

Session initiation protocol (SIP) is an application-layer protocol to overcome the limitations of H.323 QoS and diffServe. H.323 operates in a connectionless mode, where no end-to-end session or “circuit” is created for the duration of the conversation. SIP handles setting up the sessions and encompasses user location, availability, redirection and multiparty abilities in Layer 7 of the OSI protocol stack. Further, it allows VoIP gateways, PBXs and other communication systems devices to interoperate, making it far more scalable.

This protocol has less overhead, as it reuses the same header information from HTTP. Also, as it is connection-oriented multicast, unicast and other connection-dependent conversations are more reliable than they would be with dependency on QoS only. Name mapping and redirection are built into the protocol, making having one URL possible.

The Megaco/H.248 standard, which is expected to replace the media gateway control protocol, specifies specific protocols for all devices and divides the gateway functions into subcomponents. This standard also allows for devices to function with “clocked” type devices found in the PSTN to provide more affordable call-switching techniques.

To make your VoIP network ready, first determine your requirements, the health of your network and infrastructure, the vendor with whom you will be working, and the priorities of your applications on your network. There may be a need to replace older electronics, due to any lack of intelligence and antiquated technology.

Other key factors to consider are whether your systems are open enough to allow a variety of phones for later connections to the network. A call to the phone service provider is advised to find out what VoIP services they are planning to offer in the near future. The latest trend on the carrier side is to become an Ethernet local exchange carrier, which would eliminate your need to supply the gateway to the PSTN. If they plan to offer this service, find out what standard they will be supporting prior to making a decision.

In order to realize cost savings, what steps should you take? Terminations, cabling runs and labeling should be compliant with all of the appropriate standards. A protocol analyzer or network analyzer can point out potential problems. Also make sure that your cabling meets today’s standards.

Next, examine your electronics and the general health of your traffic patterns. Make sure that your electronics will understand QoS, diffServe or whatever mechanisms used by your VoIP system. Get rid of unnecessary protocols that are using up your bandwidth. Finally, and most importantly, monitor your service after it is in place.

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Carrie Higbie is network applications market manager for The Siemon Company. Watertown, Conn. She has more than 20 years’ experience in the computer industry, holds several certifications, and has taught related topics at the collegiate level and within the industry.

COPYRIGHT 2004 Nelson Publishing

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