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VoIP analysis

Best-effort delivery of packets is not good enough for VoIP, so the network must be stable and predictable to provide quality comparable to circuit-switched telephone networks.

Terry Slattery

VoIP is the new big thing in networking. Best-effort delivery of packets is not good enough for VoIP, so the network must be stable and predictable to provide quality comparable to circuit-switched telephone networks. The challenge is to provide the reliability and consistency that's needed.

Call quality factors
VoIP call quality is primarily affected by three parameters: delay, jitter, and packet loss. Combinations of these factors affect the resulting call quality.

Increasing delay affects the interaction of the parties participating in the call. One-way delay of up to 150 ms is generally considered to be acceptable. Of course, lower delay increases the level of interaction.

Jitter is the variation of the amount of delay. Let's say that a VoIP call is operating over a path that has a one-way delay of 30ms. Most packets will arrive at the destination about 30 ms after leaving the sender. But because other data packets are queued on the same interfaces as the voice packets, an occasional voice packet arrives after a delay of 35 ms. The jitter, or variation in delay, is 5 ms. Voice calls deteriorate as jitter approaches 30 ms. At some point, high jitter results in a packet arriving after the audio time slot in which it should have been played back. A burst of silence sounds a lot like a pop unless a whole stream of packets are significantly delayed, in which case it sounds like interrupted speech.

Dropped packets or damaged packets are just like very high jitter. The output audio stream is silent for the time slots of the lost packets. More than 1% of lost or damaged packets is unacceptable. Any regular volume of dropped or damaged packets should be examined and the cause determined. The most likely cause of dropped packets is due to output drops on an oversubscribed link. Damaged packets may occur on switch ports where a duplex mismatch has occurred.

Finally, network stability is a major factor in whether a network will support VoIP. Routing changes will typically result in jitter. Spanning tree topology changes will result in outages of 30 to 60 seconds if the default spanning tree timers are used. A network where QoS is not consistently implemented may exhibit intermittent symptoms of delay, jitter, and lost packets as data packets are queued in front of VoIP packets.

Identifying the source of each factor that affects voice call quality is important to a high-quality and smoothly running VoIP implementation. Performing end-to-end testing will tell you the characteristics of the path taken at the time of the testing. A network that is lightly loaded and not properly configured for VoIP may pass the tests and yet will fail after a period of time as data traffic builds and begins to compete with voice traffic. The dynamics of the interaction between voice and data require that the network infrastructure be checked for proper configuration as well as validating the basic characteristics of voice traffic.

Analyzing data
Delay, jitter, and packet loss can be measured by conducting active tests or by collecting call detail records. A VoIP analysis tool can perform active tests without adding software or probes to the network. Often, it is a good idea to install a dedicated router to act as a centralized responder for the tests.

Another source of delay, jitter, and packet loss information is the call detail records (CDRs) of each call placed on the network. Cisco's Call Manager can be configured to collect these statistics from VoIP phones. An analysis tool sorts the values for delay, jitter, and lost packets to identify the worst calls. You can then isolate the paths of the calls and determine whether the poor characteristics are intermittent or constant. Once the call path is known, further analysis can be performed on the routers, switches, and subsystems in the path.

About the author:
As founder and CEO of Netcordia, Terry Slattery is responsible for leading product development and implementation. Terry has been a successful technology innovator in networking during the past 20 years. He did development work on the Cisco IOS interface that started with release 9.21, and was instrumental in the development of CiscoSecure version 1.0. Terry co-invented and patented the v-LAB system, a router and switch pod integrated with customized virtual training courses for internetworking engineers. Terry co-authored the successful McGraw-Hill text "Advanced IP Routing in Cisco Networks," sits on Cisco Systems' Networkers Technical Advisory Board, is the second CCIE (1026) awarded, and is a sought-after industry speaker and advisor. Terry started Chesapeake Computer Consultants in 1990 which trained over 35,000 network engineers and grew profitably to become an Inc 500 company.
This was last published in December 2004

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