Do SIP support jitter buffer?

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…?

It doesn’t appear to.

Use a hardphone.

what about

make sure your x-lite’s are Transmitting Silence.

Does anyone want to work with me in getting sip jitter buffer working. Call me.