I have tested SIP channels, there have bad voice quality.
Have installed Asterisk 1.2 on Redhat 9.0 behind NAT firewall.
the tested 2 SIP clients X-lite behind another NAT firewall.
The one x-lite input the MP3 player output… the other x-lite mute micphone, the output of other x-lite have bad voice quality…
and the audio codec all is GSM.
the sound is jitter…very bad,
do asterisk 1.2 support sip jitter buffer…
i know iax support jitter buffer… oh323 support jetter buffer ?
how about sip>…
or there have some idea for get better voice quality…?