Hi
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in many
calls, Caller can hear the voice of the receiving side
but the receiver doesnt get any voice for
about 5 to 10 seconds, conversation will become two
way after 5 - 10 seconds.
Everything use to be fine if Asterisk isnt there.
Everything use to be fine if I use SJPhone or XLite as
SIP UA.
Does anybody ever had this problem ?
Any suggestions will be higly apreciated
Thanx in Advance
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