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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Some other firewalls call it ‘ALG’, application layer gateway.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Wes Sisk<br>
<b>Sent:</b> Tuesday, April 06, 2010 1:55 PM<br>
<b>To:</b> David Eco<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] AS5300 Media address<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>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
href="mailto:david.eco@msn.com"><david.eco@msn.com></a> wrote:<br>
<br>
<o:p></o:p></p>
<p class=MsoNormal>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>
<br>
<o:p></o:p></p>
<p> <o:p></o:p></p>
<p class=MsoNormal>======Debug output=======<br>
(x.y.z.a=AS5350 x.y.z.b=IP PBX)<br>
<br>
<o:p></o:p></p>
<p>Apr 6 14:15:23.866: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
INVITE <a href="mailto:sip:8411647@x.y.z.a">sip:8411647@x.y.z.a</a> SIP/2.0<br>
Call-ID: <a href="mailto:6a34022c7c7a0d2c6234438c553eceb2@172.16.1.179">6a34022c7c7a0d2c6234438c553eceb2@172.16.1.179</a><br>
CSeq: 2 INVITE<br>
From: <a href="mailto:sip:test01@x.y.z.b"><sip:test01@x.y.z.b></a>;tag=e002bb19cef9ffdee832<br>
To: <a 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 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<o:p></o:p></p>
<p class=MsoNormal>v=0<br>
o=CMI-SIPUA 63439 0 IN IP4 172.16.1.179<br>
s=SIP CALL<br>
<span style='color:red'>c=IN IP4 172.16.1.179</span><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>
<br>
<o:p></o:p></p>
<p> <o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>
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>
<span style='color:red'>Media
Dest Addr/Port : 172.16.1.179:44042</span><br>
<br>
<o:p></o:p></p>
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<p class=MsoNormal>Videos that have everyone talking! Now also in HD! <a
href="http://go.microsoft.com/?linkid=9724460" target="_new">MSN.ca Video.</a> <o:p></o:p></p>
<pre><o:p> </o:p></pre><pre style='text-align:center'>
<hr size=4 width="90%" align=center>
</pre><pre><o:p> </o:p></pre><pre>_______________________________________________<o:p></o:p></pre><pre>cisco-voip mailing list<o:p></o:p></pre><pre><a
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><o:p></o:p></pre><pre><a
href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><o:p></o:p></pre><pre> <o:p></o:p></pre>
<p class=MsoNormal><o:p> </o:p></p>
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