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I believe the NAT device in your scenario is not performing fixup. NAT
involves rewriting not only the IP header but also fixing up the
message body to rewrite embedded IPs and ports. This may involve
dynamic port allocation as well. It is generally referred to as
'protocol fixup' on the NAT device.<br>
<br>
/Wes<br>
<br>
On Tuesday, April 06, 2010 12:15:15 PM, David Eco
<a class="moz-txt-link-rfc2396E" href="mailto:david.eco@msn.com"><david.eco@msn.com></a> wrote:<br>
<blockquote cite="mid:COL114-W184FB1C029EEED102D0CA9FC180@phx.gbl"
type="cite">
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Verdana
}
--></style>Hello,<br>
I tried to send the SIP call to AS5350 from a IP PBX but got one way
audio.<br>
The scenario is SIP phone ->NAT->IP PBX -> AS5350 ->T1
circuit.<br>
When the call reached AS5350, it came with private IP address and sent
the RTP back to it instead of public IP's.<br>
Is there any way to set the media address as the IP's in FROM which
means sending the media back to the origination, IP PBX? Thank you.<br>
<p style="padding: 0px; min-height: 8pt; height: 8pt;"> </p>
======Debug output=======<br>
(x.y.z.a=AS5350 x.y.z.b=IP PBX)<br>
<p sizset="75" sizcache="5">Apr 6 14:15:23.866:
//-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:8411647@x.y.z.a">sip:8411647@x.y.z.a</a> SIP/2.0<br>
Call-ID: <a moz-do-not-send="true" class="jive-link-external-small"
href="mailto:6a34022c7c7a0d2c6234438c553eceb2@172.16.1.179">6a34022c7c7a0d2c6234438c553eceb2@172.16.1.179</a><br>
CSeq: 2 INVITE<br>
From: <a class="moz-txt-link-rfc2396E" href="mailto:sip:test01@x.y.z.b"><sip:test01@x.y.z.b></a>;tag=e002bb19cef9ffdee832<br>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:8411647@x.y.z.a"><sip:8411647@x.y.z.a></a><br>
Via: SIP/2.0/UDP x.y.z.b:5061;branch=z9hG4bk-e002bb19cef9ffdee832<br>
Max-Forwards: 70<br>
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS,INFO,MESSAGE,UPDATE<br>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:ippbx@x.y.z.b:5061"><sip:ippbx@x.y.z.b:5061></a><br>
Supported: replaces<br>
Content-Type: application/sdp<br>
Content-Length: 339</p>
v=0<br>
o=CMI-SIPUA 63439 0 IN IP4 172.16.1.179<br>
s=SIP CALL<br>
<font color="#ff0000">c=IN IP4 172.16.1.179</font><br>
t=0 0<br>
m=audio 60000 RTP/AVP 0 8 4 18 23 22 2 21 101<br>
a=rtpmap:23 G726-16/8000<br>
a=rtpmap:22 G726-24/8000<br>
a=rtpmap:2 G726-32/8000<br>
a=rtpmap:21 G726-40/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=fmtp:18 annexb=no<br>
a=rtcp:60001<br>
a=sendrecv<br>
<p style="padding: 0px; min-height: 8pt; height: 8pt;"> </p>
Stream type : voice-only<br>
Media line : 1<br>
State : STREAM_ADDING (2)<br>
Callid : -1<br>
Negotiated Codec : g711ulaw, bytes :160<br>
Negotiated DTMF relay : inband-voice<br>
Negotiated NTE payload : 0<br>
Negotiated CN payload : 0<br>
Media Srce Addr/Port : x.y.z.a<br>
<font color="#ff0000">Media Dest Addr/Port :
172.16.1.179:44042</font><br>
<br>
<br>
<br>
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target="_new">MSN.ca Video.</a>
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