Synchronization signal on ISDN when absolute silence

Hi Guys,
I know this is probably an OT and more for a developers groups but I’m taking my chances.

On some of our outbound calls we were having audio quality issues like breaking of voice and noise and we were told that this could be result of synchronization signal.
Does Asterisk recognize and remove synchronization signal/noise ( injected by the international voice network equipment) on ISDN according to A-law or U-law spec?
Are there any settings in DAHDI (or in codecs.conf) which we can tweak to see if that makes any difference?


I assume you are talking about robbed bit signalling. How can anything down stream of that know what the bit would have been if it was sent as media. information.

Thanks David,
To be frank, I had no idea about this before. I’m yet to go through ITU-T standards for A-law and u-law for synchronization signal. What I know so far is end terminal should have the ability to remove those synchronization signals and I’m trying to figure out how Asterisk does this (if it supports).

As I suspected, ITU G.711 says absolutely nothing abut synchronisation signals, and doesn’t even seem to mention robbed bit signalling. It’s a basically a fairly simple audio codec specification covering nothing but the audio samples and not providing for any other information to be sent on the channel.

Note that some international circuits may well transcode to other than G.711, and/or not transmit silence at all (TASI). Any distortion resulting from that will not be removable, and there will be no end to end indication of gaps introduced by TASI.

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