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?