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<div class=Section1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Ross,<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>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>OK – RTP to asterisk is general to the same address as the
SIP signalling, because technically Asterisk is a SIP B2BUA and when Asterisk Detects
NAT is will always put its self in the media path. You can also affect this by
setting in the SIP.CONF file nat=yes in the peer settings.<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>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>From You description the RTP coming back towards your NAT
firewall is either incorrectly addressed or being blocked by the NAT
configuration.<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>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>It might be worth – since Asterisk is in the path to relax
the NAT fix-up of Kamailio . If the purpose of having Kamailio there is as a
NAT traversal solution, then you have a couple of choices to either Deploy the
RTPProxy partner to Kamailio or leave Asterisk as the termination point of all
media – whether it is PSTN based or not. It is the latter case I guess
your currently heading towards.<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>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Try adding fix_nated_contact() to your INVITE processing:<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>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>This is some excerpts from my INVITE processing<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
if (method == "INVITE") {<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
log (1, "Inside INVITE Processing\n");<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
# Basically challenge all end-points that make calls<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
# before we ship the call to gateway<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
if (!proxy_authorize("domain.com","subscriber")) {<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
proxy_challenge("domain.com", "0");<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
break;<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
};<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
# Fixed NATed SDP and contact
headers
<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
if (nat_uac_test("3")) {<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
log(1, "Inside NATed INVITE Processing\n");<o:p></o:p></span></p>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
fix_nated_contact();<o:p></o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
fix_nated_sdp("3");<o:p></o:p></span></b></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
t_on_reply("1");<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
}<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
else {<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'> log
(1, "In else of nated UAC test\n");<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
t_on_reply("1");<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
};<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
};<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
log (1, "Sending call to PSTN gateway\n");<o:p></o:p></span></p>
<p class=MsoNormal style='margin-left:36.0pt'><span style='font-size:11.0pt;
font-family:"Courier New";color:#1F497D'> rewritehostport("w.x.y.z:5060");<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'> forward(uri:host,
uri:port);<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'> break;<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
};<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>And my Reply Route logic.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>onreply_route[1] {<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>log (1, "Inside on reply route processing\n");<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
if (status =~ "(180)|(183)|2[0-9][0-9]|4[0-9][0-9]") {<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
log(1, "Inside 180|183|200|4XX Processing\n");<o:p></o:p></span></p>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
fix_nated_contact();<o:p></o:p></span></b></p>
<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
fix_nated_sdp("3");<o:p></o:p></span></b></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
break;<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>
};<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Courier New";
color:#1F497D'>}<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>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Neill...;o)<o:p></o:p></span></p>
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<p class=MsoNormal style='mso-line-height-alt:9.0pt'><b><span
style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Neill
Wilkinson</span></b><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'><br>
Principal Consultant<br>
<o:p></o:p></span></p>
<p class=MsoNormal style='mso-line-height-alt:9.0pt'><span
style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Aeonvista
Ltd - opening up new ideas<o:p></o:p></span></p>
<p class=MsoNormal style='mso-line-height-alt:9.0pt'><span
style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'> <o:p></o:p></span></p>
</td>
</tr>
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</td>
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<td style='border:none;border-top:solid #CCCCCC 1.0pt;padding:1.2pt 0cm 0cm 0cm'>
<p class=MsoNormal><span style='font-size:6.0pt;font-family:"Arial","sans-serif";
color:#1F497D'> <o:p></o:p></span></p>
</td>
<td style='border:none;border-top:solid #CCCCCC 1.0pt;padding:1.2pt 0cm 0cm 0cm'>
<p class=MsoNormal align=right style='text-align:right'><span
style='font-size:6.0pt;font-family:"Arial","sans-serif";color:#1F497D'> <o:p></o:p></span></p>
</td>
</tr>
</table>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><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>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Ross Beer [mailto:ross_beer@hotmail.com] <br>
<b>Sent:</b> 31 July 2008 14:25<br>
<b>To:</b> neill.wilkinson@btinternet.com; users@lists.kamailio.org<br>
<b>Subject:</b> RE: [Kamailio-Users] NAT Problems<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif";color:black'>Hi Neill,</span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><br>
<br>
<span style='color:black'>The outgoing audio works fine. The audio that is not
passed is from PSTN -> Phone. </span><br>
<br>
<span style='color:black'>The Snom does have </span></span><span
style='font-family:"Calibri","sans-serif";color:black'>symmetrical RTP set. I
did have this working about a week ago then it stopped working!</span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><br>
<br>
</span><span style='font-family:"Calibri","sans-serif";color:black'>How does
asterisk deal with NAT, as if I connect the phone directly to Asterisk
everything works perfectly! </span><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'><br>
<br>
</span><span style='font-family:"Calibri","sans-serif";color:black'>Both
Asterisk & Kamailio are on publically accessible IP addresses.</span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><br>
<br>
</span><span style='font-family:"Calibri","sans-serif";color:black'>Thank you
for your help so far.</span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><br>
<br>
</span><span style='font-family:"Calibri","sans-serif";color:black'>Ross</span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><br>
<br>
<br>
<br>
<br>
<o:p></o:p></span></p>
<div class=MsoNormal align=center style='text-align:center'><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
<hr size=2 width="100%" align=center id="EC_stopSpelling">
</span></div>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'><br>
From: neill.wilkinson@btinternet.com<br>
To: users@lists.kamailio.org<br>
Date: Thu, 31 Jul 2008 13:32:05 +0100<br>
Subject: Re: [Kamailio-Users] NAT Problems<br>
<br>
<o:p></o:p></span></p>
<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Which direction is the one way audio in – from the Phone
to the PSTN or Vice-Versa – I assume audio outbound is OK and it’s
the audio returning to the phone from the PSTN you’re missing?</span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Do you have symmetrical RTP set in the Line settings
(under NAT) on the SNOM?</span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Is Kamailio inside or outside the NAT?</span><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'> </span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Neill...;o)</span><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'><o:p></o:p></span></p>
<div>
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<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
color:#1F497D'>Neill Wilkinson</span></b><span style='font-size:10.0pt;
font-family:"Verdana","sans-serif";color:#1F497D'><br>
Principal Consultant<br>
</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> users-bounces@lists.kamailio.org
[mailto:users-bounces@lists.kamailio.org] <b>On Behalf Of </b>Ross Beer<br>
<b>Sent:</b> 31 July 2008 12:50<br>
<b>To:</b> users@lists.kamailio.org<br>
<b>Subject:</b> [Kamailio-Users] NAT Problems</span><span style='font-size:
10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <o:p></o:p></span></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>I am having problems with a NAT device. I
have an asterisk server that is hosted on a public IP and when I connect
my Snom phones to it directly audio passes correctly both ways. When I
introduce Kamailio 1.3 into the mix I get one way audio.<br>
<br>
I have fixed the SDP and contacts etc and everything appears to be ok with the
sip packet. I would be very grateful for any ideas what the problem could be.<br>
<br>
Outgoing calls from Kamailio -> asterisk -> PSTN work fine.<br>
<br>
Thank you for your help in advance, it is much appreciated.<br>
<br>
<br>
=================== config =======================<br>
<br>
</span><span lang=EN style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>if
(!mf_process_maxfwd_header("10")) {<br>
sl_send_reply("483","Too Many Hops");<br>
exit;<br>
};<br>
if (msg:len >= 2048 ) {<br>
sl_send_reply("513", "Message too big");<br>
exit;<br>
};<br>
# NAT detection<br>
force_rport();<br>
if(nat_uac_test("3"))<br>
{<br>
<br>
fix_nated_contact();<br>
}<br>
if (!method=="REGISTER")<br>
{<br>
record_route();<br>
}<br>
if (loose_route()) {<br>
append_hf("P-hint: rr-enforced\r\n"); <br>
route(1);<br>
};<br>
if (!uri==myself) {<br>
append_hf("P-hint: outbound\r\n"); <br>
route(1);<br>
};<br>
if (uri==myself) {<br>
if (method=="REGISTER") <br>
{<br>
if (!www_authorize("", "subscriber")) {<br>
www_challenge("", "0");<br>
exit;<br>
};<br>
<br>
<br>
if (isflagset(5)) <br>
{<br>
# set branch flag -- when someone will call this user<br>
# the INVITE will have branch flag 6 set after lookup("location")<br>
setbflag(6);<br>
<br>
};<br>
<br>
fix_nated_contact();<br>
fix_nated_register();<br>
consume_credentials();<br>
save("location");<br>
exit;<br>
};<br>
if (!lookup("location")) {<br>
sl_send_reply("404", "Not Found");<br>
exit;<br>
};<br>
append_hf("P-hint: usrloc applied\r\n"); <br>
};<br>
<br>
if(method=="MESSAGE")<br>
{<br>
if (!lookup("location"))<br>
{<br>
sl_send_reply("404", "User Offline");<br>
exit;<br>
}<br>
route(4);<br>
}<br>
<br>
if (method=="REGISTER")<br>
{<br>
save("location");<br>
}<br>
<br>
route(1);<br>
}<br>
<br>
route[1] {<br>
if (subst_uri('/(sip:.*);nat=yes/\1/')){<br>
setbflag(6);<br>
};<br>
if (isflagset(5)||isbflagset(6)) {<br>
route(3);<br>
}<br>
t_on_reply("1");<br>
if (!t_relay()) {<br>
sl_reply_error();<br>
};<br>
exit;<br>
}<br>
<br>
route[3] {<br>
if (is_method("BYE")) <br>
{<br>
unforce_rtp_proxy();<br>
} <br>
else if (is_method("INVITE"))<br>
{<br>
fix_nated_sdp("3");<br>
<br>
};<br>
if (isflagset(5))<br>
{<br>
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');<br>
}<br>
t_on_reply("1");<br>
}<br>
route[4]<br>
{<br>
if (!t_relay()) <br>
{<br>
sl_reply_error();<br>
};<br>
exit;<br>
}<br>
<br>
<br>
failure_route[2] {<br>
if (isbflagset(6) || isflagset(5)) <br>
{<br>
unforce_rtp_proxy();<br>
}<br>
}<br>
onreply_route[1] <br>
{<br>
fix_nated_contact();<br>
if (((isflagset(5) || isbflagset(6)) &&
status=~"(183)|(2[0-9][0-9])") || is_method("INVITE")) <br>
{<br>
fix_nated_sdp("2");<br>
force_rport();<br>
<br>
}<br>
search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes');<br>
}<br>
<br>
<br>
<br>
==================== SIP PACKETS ===========================<br>
INVITE sip:10001*202@**** IP ****;line=e25l7qyi SIP/2.0<br>
Record-Route: <sip:*** IP ***;lr=on;ftag=as0a2e0c2e><br>
Via: SIP/2.0/UDP *** IP ***;branch=z9hG4bKb0a1.cf08f082.0<br>
v: SIP/2.0/UDP *** IP ***:5060;branch=z9hG4bK2bc03835;rport=5060<br>
f: "Name" <sip:*** DOMAIN ***>;tag=as0a2e0c2e<br>
t: <sip:10001*202@** DOMAIN **><br>
m: <sip:number@**IP**><br>
i: 74391cce38d7c7f7549c863a651b8f81@*** DOMAIN ****<br>
CSeq: 102 INVITE<br>
User-Agent: asterisk<br>
Max-Forwards: 69<br>
Date: Thu, 31 Jul 2008 11:38:39 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
k: replaces<br>
c: application/sdp<br>
l: 422<br>
P-hint: usrloc applied<br>
<br>
v=0<br>
o=root 5390 5390 IN IP4 194.xxx.xxx.xxx<br>
s=session<br>
c=IN IP4 194.xxx.xxx.xxx<br>
b=CT:384<br>
t=0 0<br>
m=audio 12558 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
m=video 17918 RTP/AVP 34 103 99<br>
a=rtpmap:34 H263/90000<br>
a=rtpmap:103 h263-1998/90000<br>
a=rtpmap:99 H264/90000<br>
a=sendrecv<br>
<br>
<br>
////////////// PHONE REPLY ////////////////////<br>
SIP/2.0 200 Ok<br>
Via: SIP/2.0/UDP **** IP ****:5060;branch=z9hG4bK19d7ec13;rport=5060<br>
Record-Route: <sip:*** IP ***;lr=on;ftag=as486f79a5><br>
From: "*** NUMBER *****" <sip:*** DOMAIN
*****>;tag=as486f79a5<br>
To: <sip:10001*202@*** DOMAIN *****>;tag=jgdy33ee6n<br>
Call-ID: *** ID ***<br>
CSeq: 102 INVITE<br>
Contact: <sip:*** DOMAIN ****:54686;line=e25l7qyi;nat=yes>;reg-id=1<br>
User-Agent: snom370/7.3.7<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO<br>
Allow-Events: talk, hold, refer, call-info<br>
Supported: timer, replaces, from-change<br>
Content-Type: application/sdp<br>
Content-Length: 508<br>
v=0<br>
o=root 1569142945 1569142946 IN IP4 192.168.1.20<br>
s=call<br>
c=IN IP4 ** EXTERNAL IP ***<br>
t=0 0<br>
m=audio 10014 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 pcmu/8000<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014<br>
a=sendrecv<br>
m=video 0 RTP/AVP 34 103 99<br>
a=rtpmap:34 H263/90000<br>
a=rtpmap:103 h263-1998/90000<br>
a=rtpmap:99 H264/90000<br>
a=alt:1 0.9 : user 9kksj== 192.168.1.20 10014<br>
a=sendrecv<br>
a=oldmediaip:192.168.1.20</span><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p></span></p>
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