[asterisk-users] TCP & UDP path not the same

[Rilawich Ango]

HI,
  I got a one way audio when an ip phone dial to another ip phone in
the same network.  What I find is TCP & UDP run different legs.  Below
is my configuration.

asterisk (192.168.1.10)
ipphone-A (192.168.1.111)
ipphone-B (192.168.1.101)
router (192.168.1.1) external IP (116.48.138.83)

When A makes call to B, signal from A to router goes in the internal
network.  Then B pickup the call and I find that B will use external
IP to reach the router.  The signal from B finally can't reach to A.
Below is a flow and you can see it involves using external IP.  Is it
related to the setting?  Where and how to set it to make it work?

U 192.168.1.10:5060 -> 192.168.1.101:5060
INVITE sip:101 at 192.168.1.101:5060 SIP/2.0.
Via: SIP/2.0/UDP 116.48.138.83:5060;branch=z9hG4bK612c1103;rport.
From: "111" <sip:101 at 116.48.138.83>;tag=as0a0b2a95.
To: <sip:101 at 192.168.1.101:5060>.
Contact: <sip:101 at 116.48.138.83>.
Call-ID: 5ab3539a064164ed6bf80c255c545a47 at 116.48.138.83.
CSeq: 102 INVITE.
User-Agent: PBX.
Max-Forwards: 70.
Date: Fri, 13 Jun 2008 17:20:14 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Content-Type: application/sdp.
Content-Length: 265.
.
v=0.
o=root 21200 21200 IN IP4 116.48.138.83.
s=session.
c=IN IP4 116.48.138.83.
t=0 0.
m=audio 19770 RTP/AVP 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.
a=sendrecv.