Variable Delays Associated with VoIP Traffic


The bullet points below each type of variable delay encountered with Voip traffic, and how these issues can be compensated for where possible.

• Queuing Delay. When a WAN interface is congested, traffic must be queued using any of the various methods looked at in this chapter. Although a method like LLQ can prioritize voice traffic, another issue exists. Consider a situation where a router interface currently has no priority traffic waiting to be sent, and begins to forward a large frame containing FTP data. If a voice packet then arrived, it could not be sent until the FTP packet (which is already being serialized) is completed. As such, the voice packet is subject to wait, incurring a delay that might not be reasonable. For example, if the voice packet is stuck behind a 1500 byte frame being sent over a 64 kbps link, it would be subject to a delay of approximate 185 ms, which (in conjunction with other delay factors) would make it well beyond acceptable. To account for queuing delays, a technique called Link Fragmentation with Interleaving is used on links with speeds below 768 kbps. When implemented, a router will fragment larger packets (like FTP in this example) into smaller sizes, and then “interleave” the voice packets onto the link. As such, the voice packets would not need to wait for the entire single FTP packet to be sent. When choosing a fragment size, it should be one that aims for approximately a 10 ms delay but does not fragment voice packets.

• Dejitter Delay. As mentioned earlier, jitter occurs when packets do not arrive when expected. When dejitter buffers are configured at the receiving end of a voice network, packets that arrive with timing variations are buffered, and then converted to a constant delay. The use of dejitter buffers does add some delay to voice network, so the buffers should generally be kept small. Other QoS techniques looked at earlier in this section help to reduce the overall exposure of voice traffic to jitter issues, but dejitter buffers are a specific solution to help minimize jitter on the receiving end of a VoIP network.

It’s important to keep the concept of “end-to-end” in mind when calculating delay. Don’t forget that if a packet needs to pass through 3 routers, and each router adds a 10 ms delay to the forwarding of the packet based on queuing considerations, that adds an additional 30 ms of delay to the packet between the source and destination. Calculations of end-to-end delay will be looked at in more detail shortly in the planning section.

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