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	Convoq ASAP

Audio Conferencing - VoIP

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Convoq ASAP integrates full multi-party VoIP audio conferencing.

The VoIP component of Convoq ASAP is based on the standard Flash Communication Server abilities, optimized and adjusted to give the best performance without taxing too much participant resources.

On average, the amount of bandwidth required by a ASAP to send your voice as VoIP is about 15 to 30 Kbps. Therefore for a two people conversation you need at least a 56K line working well.

Convoq ASAP offers an extended set of controls to optimize both audio and video bandwidth to the individual needs of each participant.

These functions are accessible only to ASAP licensed users and provide the ability to set the connection bandwidth available as well as the audio sampling rate, echo cancellation and to control the level of voice at which your mike will switch itself on. These last two functions in particular are of great help in reducing the amount of background noise and loopback that may easily take place in this live sessions.

In my testing the echo cancellation seemed not to have any effect on my sessions, while I greatly enjoyed being able to properly set my microphone minimum treshold level for activation through the handy and intuitive facility created by Convoq.

The audio channel is always open on all sides and automatically activated by each participant's voice.

Audio quality is good, though the audio sampling rate can be changed from 5 Khz to 44 Khz. Default is 8Khz. There are few drop outs and one can enjoy a good uninterrupted conversation without any problem.

That means everyone is working in full-duplex and there is no need to press and hold any button to talk to one another.

Cons:

Audio delay can be from 500 milliseconds (half a second) up to several seconds even on very speedy Internet connection lines. This seems to be an inherent Flash Communication Server limitation which will be gradually overcome in time.

Due to the above delay multi-party conversations may suffer a bit unless they are properly moderated. It is not easy to have a fluent and live discussion if voice interventions have delays greater than a few hundred milliseconds.

It is absolutely essential to use headsets and mikes, versus open speakers, as the delay causes an annoying an distracting echo, which can be mostly avoided in this way.

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Posted on April 02, 2004 at 07:19 AM

Updated on April 02, 2004 at 09:13 AM

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Audio Conferencing - VoIP