<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML xmlns:o = "urn:schemas-microsoft-com:office:office"><HEAD>
<META http-equiv=Content-Type content="text/html; charset=iso-8859-1">
<META content="MSHTML 6.00.2900.2604" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<DIV>Look at this:</DIV>
<DIV>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT face="Times New Roman"
size=3>U 84.203.148.146:5060 -> 157.190.74.151:5060</FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes"> </SPAN>SIP/2.0 200
OK..Via: SIP/2.0/UDP
157.190.74.151;rport=5060;branch=z9hG4bKcd</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes">
</SPAN>17ddd1b59ead49..From: "2092"
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes"> </SPAN>..To:
<sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:
8ffc2d18b218</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes">
</SPAN>70b3@157.190.74.151..CSeq: 64735 INVITE..User-Agent: Grandstream BT100
1.0.</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes">
</SPAN>5.18..Contact: <sip:2093@172.16.3.31>..Allow:
INVITE,ACK,CANCEL,BYE,NOTIFY,</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes">
</SPAN>REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported:
rep</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes">
</SPAN>laces..Content-Length: 174....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=SIP</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes"> </SPAN>Call..c=IN
IP4 84.203.148.146..t=0 0..m=audio 35016 RTP/AVP
0..a=sendrecv..</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT size=3><FONT
face="Times New Roman"><SPAN style="mso-spacerun: yes"> </SPAN>a=rtpmap:0
PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..</FONT></FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT
face="Times New Roman"></FONT> </P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT face="Times New Roman">I
assume this is where you get an error message. You haven't called
fix_nated_contact() for this message, and in fact I believe there may be an
error in the ONsip.org example where a line has been lost.</FONT></P><FONT
face="Courier New" size=1>
<P align=left>192. onreply_route[1] {</P>
<P align=left>193.</P>
<P align=left>194. if (isflagset(6) &&
status=~"(180)|(183)|2[0-9][0-9]") {</P>
<P align=left>195. if (!search("^Content-Length:\ 0")) {</P>
<P align=left>196. force_rtp_proxy();</P>
<P align=left>197. };</P>
<P align=left>198. } else if (nat_uac_test("1")) {</P>
<P align=left>199. fix_nated_contact();</P>
<P align=left>200. };</P>
<P align=left>201. }</P></FONT>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT
face="Times New Roman">fix_nated_contact() should go in between line 194 and
195. </FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT
face="Times New Roman">Could you please confirm that this works? I will
look at the config file.</FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"><FONT
face="Times New Roman">g-(</FONT></P>
<P class=MsoNormal style="MARGIN: 0cm 0cm 0pt"> </P></DIV>
<DIV>---- Original Message ----<BR>From: Vivienne Curran<BR>To:
serusers@lists.iptel.org ; greger@teigre.com<BR>Sent: Wednesday, April 06, 2005 04:12
PM<BR>Subject: Re: RTPProxy fails only for Private to Public
communication<BR><BR>> Just as an extra : I have a sniff of the message for
when a public<BR>> client (2092)rings a private client (2093)included at the
bottom of<BR>> this email. I cant see anything wrong with them but maybe it
will<BR>> shed more light on the matter. <BR>> <BR>>
Vivienne Curran <vivcurran@yahoo.co.uk> wrote:<BR>> I changed the line
modparam("nathelper", "rtpproxy_sock",<BR>> "/var/run/rtpproxy.sock") to
modparam("nathelper", "rtpproxy_sock",<BR>> "udp:localhost:22222") and
started the rtpproxy as ./rtpproxy -s udp<BR>> from the relevant directory
and this resulted in a series of<BR>> "rtpp_command: no response from
rtpproxy" and rtpproxy temporarily<BR>> disabled" errors. If I return to the
original modparam and start it<BR>> as ./rtpproxy then it works but like I
said when the private client<BR>> rings the public client, I get "ERROR:
send_rtpp_command: cant read<BR>> reply from a RTP
Proxy". <BR>> <BR>> Any further
ideas? Has anyone on the mailing list experienced this? I<BR>> am using the
script given in the onsip getting started doc for 0.9.0.<BR>> but am using
ser 0.8.14. <BR>> <BR>> BR,<BR>> Vivienne<BR>> <BR>>
<BR>> "Greger V. Teigre" <greger@teigre.com> wrote:<BR>> See
inline.<BR>> <BR>>> Thank you for that Greger. I have altered my script
so that it<BR>>> exactly mimics the one in the onsip document besides the
has_totag<BR>>> and fix_nated register. All is good when I ring from a
private phone<BR>>> to a public phone i.e. the audio is very clear and the
following<BR>>> messages are in /var/log.<BR>>> <BR>>> ERROR:
extract_body: message body has length zero<BR>>> ERROR: force_rtp_proxy2:
cant extract body from the message.<BR>>> <BR>>> I assume this is
because of the 200 OK to a register message where<BR>>> theres no sdp?? Is
this correct?<BR>> <BR>> That's correct. You will find code in the
example configs where we<BR>> test for an empty body before calling
force_rtp_proxy. <BR>> <BR>>> However when I try to phone from public
into private I get:<BR>>> <BR>>> ERROR: send_rtpp_command: cant read
reply from a RTP Proxy.<BR>>> <BR>>> I find this confusing because I
know the rtpproxy is working.<BR>> <BR>> This means that rtpproxy is not
responding to a particular message. I<BR>> have heard some people have had
problems with the socket based<BR>> communication. I only use UDP. This is
what you do to set up udp<BR>> (22222 is default port): <BR>>
modparam("nathelper", "rtpproxy_sock", "udp:localhost:22222")<BR>> rtpproxy
must be started with -s udp:*<BR>> g-)<BR>> <BR>>> BR<BR>>>
Vivienne.<BR>>> <BR>>> "Greger V. Teigre" <greger@teigre.com>
wrote:<BR>>> Yes, you can use fix_nated_contact instead. It is not
entirely<BR>>> RFC-compliant, but that's what you have in
0.8.14.<BR>>> The has_totag() only tests to see if the INVITE has a To
header,<BR>>> which means that it is in-dialog and thus is a
re-INVITE. An INVITE<BR>>> will normally not have loose routing
unless you have another SIP<BR>>> proxy forwarding an INVITE to you (in
which case you should assume<BR>>> that the other proxy handles NAT and
thus not trigger NAT-related<BR>>> code). You can safely remove the
has_totag() if you use<BR>>> force_rtp_proxy("l")<BR>>>
g-)<BR>>> <BR>>> ---- Original Message ----<BR>>> From:
Vivienne Curran<BR>>> To: Greger V. Teigre ;
serusers@lists.iptel.org<BR>>> Sent: Tuesday, April 05, 2005 02:25
PM<BR>>> Subject: Re: [Serusers] Contact Header and SDP not
rewritten<BR>>> <BR>>>> Greger,<BR>>>> <BR>>>>
Since fix_nated_register does not exist with 0.8.14, will<BR>>>>
fix_nated_contact do instead? Also if I am leaving out the<BR>>>>
has_totag() at the start of the script, will this greatly effect
its<BR>>>> functionality?<BR>>>> <BR>>>> Thank
you,<BR>>>> Vivienne<BR>> Send instant messages to your online
friends<BR>> http://uk.messenger.yahoo.com <BR>> U 157.190.74.151:5060
-> 84.203.148.146:5060<BR>> INVITE sip:2093@84.203.148.146
SIP/2.0..Via: SIP/2.0/UDP<BR>> 157.190.74.151;bra <BR>>
nch=z9hG4bKcd17ddd1b59ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aed <BR>> c22bd5a3b510c..To:
<sip:2093@84.203.148.146>..Contact:<BR>> <sip:2092@157.190.74
<BR>> .151>..Supported: replaces..Call-ID:<BR>>
8ffc2d18b21870b3@157.190.74.151..CSeq: <BR>> 64735
INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18..Max-Forwards: 70..Al
<BR>> low:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Typ<BR>>
e: application/sdp..Content-Length: 426....v=0..o=2092 8000 0 IN<BR>> IP4
157.19 <BR>> 0.74.151..s=SIP Call..c=IN IP4 157.190.74.151..t=0
0..m=audio 5004<BR>> RTP/AVP <BR>> 0 8 4 18 2 15 99 9
101..a=sendrecv..a=rtpmap:0<BR>> PCMU/8000/3..a=rtpmap:8 PCMA
<BR>> /8000/3..a=rtpmap:4 G723/8000/3..a=rtpmap:18<BR>>
G729/8000/3..a=rtpmap:2 G726-3 <BR>> 2/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99<BR>> iLBC/8000/3..a=fmtp:99 mode=
<BR>> 20..a=rtpmap:9
G722/8000/3..a=ptime:20..a=rtpmap:101<BR>> telephone-event/8000/3
<BR>> ..a=fmtp:101 0-11..<BR>> <BR>> U 84.203.148.146:5060
-> 157.190.74.151:5060<BR>> SIP/2.0 100 trying -- your call is
important to us..Via:<BR>> SIP/2.0/UDP 157.19 <BR>>
0.74.151;branch=z9hG4bKcd17ddd1b59ead49;rport=5060..From: "2092"<BR>>
<sip:2092@ <BR>>
84.203.148.146>;tag=aedc22bd5a3b510c..To:<BR>>
<sip:2093@84.203.148.146>..Call-I <BR>> D:
8ffc2d18b21870b3@157.190.74.151..CSeq: 64735 INVITE..Server: Sip<BR>> EXpress
<BR>> router (0.8.14 (i386/linux))..Content-Length:
0..Warning: 392<BR>> 84.203.148.1 <BR>> 46:5060 "Noisy
feedback tells: pid=8990 req_src_ip=157.190.74.151<BR>> req_src_
<BR>> port=5060 in_uri=sip:2093@84.203.148.146<BR>>
out_uri=sip:2093@84.203.148.14:506 <BR>> 0
via_cnt==1"....<BR>> <BR>> U 84.203.148.146:5060 ->
84.203.148.14:5060<BR>> INVITE sip:2093@84.203.148.14:5060
SIP/2.0..Via: SIP/2.0/UDP<BR>> 84.203.148.146 <BR>>
;branch=z9hG4bKf51e.b169be72.0..Via: SIP/2.0/UDP<BR>>
157.190.74.151;rport=5060; <BR>>
branch=z9hG4bKcd17ddd1b59ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag= <BR>> aedc22bd5a3b510c..To:
<sip:2093@84.203.148.146>..Contact:<BR>> <sip:2092@157.190
<BR>> .74.151:5060>..Supported: replaces..Call-ID:<BR>>
8ffc2d18b21870b3@157.190.74.15 <BR>> 1..CSeq: 64735
INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18..Max-Forwards
<BR>> : 69..Allow:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Con
<BR>> tent-Type: application/sdp..Content-Length:
445....v=0..o=2092 8000<BR>> 0 IN IP <BR>> 4
157.190.74.151..s=SIP Call..c=IN IP4 84.203.148.146..t=0<BR>> 0..m=audio
35014 <BR>> RTP/AVP 0 8 4 18 2 15 99 9
101..a=sendrecv..a=rtpmap:0<BR>> PCMU/8000/3..a=rtpm <BR>>
ap:8 PCMA/8000/3..a=rtpmap:4 G723/8000/3..a=rtpmap:18<BR>>
G729/8000/3..a=rtpmap <BR>> :2 G726-32/8000/3..a=rtpmap:15
G728/8000/3..a=rtpmap:99<BR>> iLBC/8000/3..a=fmtp <BR>> :99
mode=20..a=rtpmap:9 G722/8000/3..a=ptime:20..a=rtpmap:101<BR>> telephone-eve
<BR>> nt/8000/3..a=fmtp:101 0-11..a=nortpproxy:yes..<BR>>
<BR>> U 84.203.148.14:5060 -> 84.203.148.146:5060<BR>>
SIP/2.0 100 Trying..Via: SIP/2.0/UDP<BR>>
84.203.148.146;branch=z9hG4bKf51e.b169 <BR>> be72.0..Via:
SIP/2.0/UDP<BR>> 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b5
<BR>> 9ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <s
<BR>> ip:2093@84.203.148.146>..Call-ID:<BR>>
8ffc2d18b21870b3@157.190.74.151..CSeq: 64 <BR>> 735
INVITE..User-Agent: Grandstream BT100 1.0.5.18..Content-Length:<BR>> 0....
<BR>> <BR>> U 84.203.148.14:5060 ->
84.203.148.146:5060<BR>> SIP/2.0 180 Ringing..Via:
SIP/2.0/UDP<BR>> 84.203.148.146;branch=z9hG4bKf51e.b16 <BR>>
9be72.0..Via: SIP/2.0/UDP<BR>>
157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b <BR>>
59ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <
<BR>>
sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>>
8ffc2d18b21870b3@15 <BR>> 7.190.74.151..CSeq: 64735
INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18..C <BR>>
ontent-Length: 0....<BR>> <BR>> U 84.203.148.146:5060 ->
157.190.74.151:5060<BR>> SIP/2.0 180 Ringing..Via:
SIP/2.0/UDP<BR>> 157.190.74.151;rport=5060;branch=z9hG <BR>>
4bKcd17ddd1b59ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3 <BR>> b510c..To:
<sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>> 8ffc2d1
<BR>> 8b21870b3@157.190.74.151..CSeq: 64735
INVITE..User-Agent:<BR>> Grandstream BT100 <BR>>
1.0.5.18..Content-Length: 0....<BR>> <BR>> U 84.203.148.14:5060 ->
84.203.148.146:5060<BR>> SIP/2.0 200 OK..Via: SIP/2.0/UDP<BR>>
84.203.148.146;branch=z9hG4bKf51e.b169be72 <BR>> .0..Via:
SIP/2.0/UDP<BR>> 157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b59ead
<BR>> 49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <sip:2
<BR>>
093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>>
8ffc2d18b21870b3@157.190 <BR>> .74.151..CSeq: 64735
INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18..Contac
<BR>> t: <sip:2093@172.16.3.31>..Allow:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION <BR>>
S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported:<BR>>
replaces..Conte <BR>> nt-Length: 152....v=0..o=2093 8000 0 IN IP4
172.16.3.31..s=SIP<BR>> Call..c=IN I <BR>> P4 172.16.3.31..t=0
0..m=audio 5004 RTP/AVP<BR>> 0..a=sendrecv..a=rtpmap:0 PCMU/
<BR>> 8000/3..a=ptime:20..<BR>> <BR>> U 84.203.148.146:5060
-> 157.190.74.151:5060<BR>> SIP/2.0 200 OK..Via:
SIP/2.0/UDP<BR>> 157.190.74.151;rport=5060;branch=z9hG4bKcd
<BR>> 17ddd1b59ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c <BR>> ..To:
<sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>>
8ffc2d18b218 <BR>> 70b3@157.190.74.151..CSeq: 64735
INVITE..User-Agent: Grandstream<BR>> BT100 1.0. <BR>>
5.18..Contact: <sip:2093@172.16.3.31>..Allow:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY, <BR>>
REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:<BR>> application/sdp..Supported:
rep <BR>> laces..Content-Length: 174....v=0..o=2093 8000 0 IN
IP4<BR>> 172.16.3.31..s=SIP <BR>> Call..c=IN IP4
84.203.148.146..t=0 0..m=audio 35016 RTP/AVP<BR>> 0..a=sendrecv..
<BR>> a=rtpmap:0
PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..<BR>> <BR>> U
84.203.148.14:5060 -> 84.203.148.146:5060<BR>> SIP/2.0 200
OK..Via: SIP/2.0/UDP<BR>> 84.203.148.146;branch=z9hG4bKf51e.b169be72
<BR>> .0..Via: SIP/2.0/UDP<BR>>
157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b59ead <BR>>
49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <sip:2
<BR>>
093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>>
8ffc2d18b21870b3@157.190 <BR>> .74.151..CSeq: 64735
INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18..Contac
<BR>> t: <sip:2093@172.16.3.31>..Allow:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION <BR>>
S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported:<BR>>
replaces..Conte <BR>> nt-Length: 152....v=0..o=2093 8000 1 IN IP4
172.16.3.31..s=SIP<BR>> Call..c=IN I <BR>> P4 172.16.3.31..t=0
0..m=audio 5004 RTP/AVP<BR>> 0..a=sendrecv..a=rtpmap:0 PCMU/
<BR>> 8000/3..a=ptime:20..<BR>> <BR>> U 84.203.148.146:5060
-> 157.190.74.151:5060<BR>> SIP/2.0 200 OK..Via:
SIP/2.0/UDP<BR>> 157.190.74.151;rport=5060;branch=z9hG4bKcd
<BR>> 17ddd1b59ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c <BR>> ..To:
<sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>>
8ffc2d18b218 <BR>> 70b3@157.190.74.151..CSeq: 64735
INVITE..User-Agent: Grandstream<BR>> BT100 1.0. <BR>>
5.18..Contact: <sip:2093@172.16.3.31>..Allow:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY, <BR>>
REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:<BR>> application/sdp..Supported:
rep <BR>> laces..Content-Length: 174....v=0..o=2093 8000 1 IN
IP4<BR>> 172.16.3.31..s=SIP <BR>> Call..c=IN IP4
84.203.148.146..t=0 0..m=audio 35016 RTP/AVP<BR>> 0..a=sendrecv..
<BR>> a=rtpmap:0
PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..<BR>> <BR>> U
84.203.148.14:5060 -> 84.203.148.146:5060<BR>> SIP/2.0 200
OK..Via: SIP/2.0/UDP<BR>> 84.203.148.146;branch=z9hG4bKf51e.b169be72
<BR>> .0..Via: SIP/2.0/UDP<BR>>
157.190.74.151;rport=5060;branch=z9hG4bKcd17ddd1b59ead <BR>>
49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c..To: <sip:2
<BR>>
093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>>
8ffc2d18b21870b3@157.190 <BR>> .74.151..CSeq: 64735
INVITE..User-Agent: Grandstream BT100<BR>> 1.0.5.18..Contac
<BR>> t: <sip:2093@172.16.3.31>..Allow:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTION <BR>>
S,INFO,SUBSCRIBE..Content-Type: application/sdp..Supported:<BR>>
replaces..Conte <BR>> nt-Length: 152....v=0..o=2093 8000 2 IN IP4
172.16.3.31..s=SIP<BR>> Call..c=IN I <BR>> P4 172.16.3.31..t=0
0..m=audio 5004 RTP/AVP<BR>> 0..a=sendrecv..a=rtpmap:0 PCMU/
<BR>> 8000/3..a=ptime:20..<BR>> <BR>> U 84.203.148.146:5060
-> 157.190.74.151:5060<BR>> SIP/2.0 200 OK..Via:
SIP/2.0/UDP<BR>> 157.190.74.151;rport=5060;branch=z9hG4bKcd
<BR>> 17ddd1b59ead49..From: "2092"<BR>>
<sip:2092@84.203.148.146>;tag=aedc22bd5a3b510c <BR>> ..To:
<sip:2093@84.203.148.146>;tag=acd725e00242a605..Call-ID:<BR>>
8ffc2d18b218 <BR>> 70b3@157.190.74.151..CSeq: 64735
INVITE..User-Agent: Grandstream<BR>> BT100 1.0. <BR>>
5.18..Contact: <sip:2093@172.16.3.31>..Allow:<BR>>
INVITE,ACK,CANCEL,BYE,NOTIFY, <BR>>
REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type:<BR>> application/sdp..Supported:
rep <BR>> laces..Content-Length: 174....v=0..o=2093 8000 2 IN
IP4<BR>> 172.16.3.31..s=SIP <BR>> Call..c=IN IP4
84.203.148.146..t=0 0..m=audio 35016 RTP/AVP<BR>> 0..a=sendrecv..
<BR>> a=rtpmap:0
PCMU/8000/3..a=ptime:20..a=nortpproxy:yes..<BR>> <BR>> Send instant
messages to your online friends<BR>> http://uk.messenger.yahoo.com
</DIV></BODY></HTML>