[asterisk-users] Initial cut off audio

[jg]

It seems that initial audio for SIP channels does not get transmitted 
for a period of varying length, typically about 1 second. This also 
applies to bridged SIP calls as well to one-legged calls where only 
Playback() gets called.

The Definitive Asterisk Guide uses constructs like "silence/1" or 
"Wait()" extensively and the explanation given in the text is "to 
establish audio", if I remember this correctly. Normally, this delay 
does not seem to be a problem, but I have two installations 
(restaurants---because every syllable seems to be important when they 
shout at each other) that are problematic and where I got complaints 
about the initially cut off audio.

Does somebody know whether the delay in establishing the audio signal is 
a typical Asterisk problem or are all VoIP solutions are affected? I am 
also not sure about the real cause. Is it really Asterisk that needs 
some time for the RTP streams or are the SIP phones responsible?

jg