Hi,
is there any way to increse the buffer or something to make SIP
connections sound better? When I make the calls with Asterisk as a SIP
client (through sip.voipbuster.com) the sound quality is poor -
constantly breaking (there are few occasional seconds when the sound is
OK)- but with any other SIP client on the same network and through
sip.voipbuster.com sound is allways OK.
Matic
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