One way audio problem with one provider

I need help. I got one way audio with one specific provider and found something in tcpdump log.

SIP client s.s.s.s to Asterisk a.a.a.a - Provider p.p.p.p - PSTN

After connection has been created, one package received from provider to tell asterisk another RTP ip to connect, so the connection changed to

SIP client s.s.s.s to Asterisk a.a.a.a - Provider o.o.o.o, then I see connection to be,
s.s.s.s:s => a.a.a.a:a
a.a.a.a:a => o.o.o.o:o
p.p.p.p:p => a.a.a.a:a
( No connection o.o.o.o:o => a.a.a.a:a or a.a.a.a:a => p.p.p.p:p have been found after rtp ip change)

result is I (SIP client) can hear other party, but other party can’t hear me. Is there anything I should setup in asterisk for rtp ip change in sip. I use 1.6 and 1.8, same issue. Please help.

One way audio problems are related to SDP. It is not clear which fields in the SIP messages your are referring to, but I don’t think they are the SDP.