Managing delay / latency critical to successful VoIP

by Jeff Wiener on July 7, 2009

Last week I posted an article on how packet loss can impact voice.  Another issue which is critical to a successful VoIP implementation is DELAY.

Another issue which is critical to a successful VoIP implementation is DELAY. An IP packet gets transmitted normally in a very timely fashion. Therefore, the recipient will receive a continuous stream of packets every…let’s say 40 milliseconds. In this case, there is a 40-millisecond delay between when the sender sends the packet to when the recipient receives the packet. If the continuous IP packet delay is 40 milliseconds on both sides for the entire flow of conversation, then that would make for a clean VoIP conversation. However, it never happens exactly as planned. You see, in a data world, especially when packets are getting sent over the Internet, packets can get delayed. If the delay is too long, then it might appear as though the packet is actually lost. Let’s say for example, that we expect a 70-millisecond delay from sender to recipient. Therefore, every word that is said on the sending side is played out 70 milliseconds later on the recipient side. Packets are then coming out as follows: 70, 70, 60, 50, 40, 70, 70, 130, 70…and so on. The packet that arrived 130 milliseconds later will appear as a dead zone in the conversation, or essentially, be not that much different from a lost packet (as described in packet loss earlier). Delay and packet loss are therefore two of the more critical issues that need to be addressed in the VoIP world.

I will continue the discussion in future BLOG postings about how to mitigate against packet loss and what you might consider doing about it.

P.S.  It doesn’t matter what kind of phone system you are running – Avaya IP Office, Nortel BCM, Cisco Unified Communication Manager – all systems are equally affected by poor VoIP.

Jeff – 250 Rimrock Rd, Toronto, Canada

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