Can you reduce signaling delay?

Can you reduce signaling delay?

It seems any time VoIP is discussed, the terms "delay" and "jitter" are invariably mentioned. It is widely known that the transmission time of voice packets from one end of a network to another is a major factor in experiencing an acceptable quality of sound. However, delay also affects the signaling in your IP Telephony network.

The term signaling refers to the messages between your IP phones, gateways and other components, and it's responsible for setting up and tearing down calls, as well as transmitting dialed digits, tones, and negotiating codecs, etc. The important thing to know is that several signaling messages have to go back and forth before two people can hear each other's voices through the telephones.

In the case of two IP phones using the H.323 protocol, as many as 11 messages are required. Thus, if your delay is, for example, 125 ms (which is considerably less than the maximum recommendation of 150 ms), then almost a second and a half will elapse between the time a user picks up a ringing phone and the media path is established. Now, a second and a half might not seem like a lot, but if the user answers the phone "hello", which usually takes about a half a second (unless you're from the South, like I am, where it takes about 2 seconds) the calling party may miss the greeting entirely, which can cause some confusion and unhappy users.

Since much of the delay is a function of the speed of light, there is often little you can do to

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shorten it, but there are two things you can investigate which may help you reduce this time. The first is to use MGCP instead of H.323. MGCP requires far fewer messages, and thus, sets up calls faster. The second thing you maybe able to use is H.323's Fast Connect.

Thomas Alexander Lancaster IV is a consultant and author with over ten years experience in the networking industry, focused on Internet infrastructure.


This was first published in April 2003

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