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<HTML><HEAD><TITLE>Anyway to steal Media IP/Port from RTPProxy or MediaProxy</TITLE>
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<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Dear Greger & List</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Actually, I want Asterisk to deploy re-INVITE to let the
media flow directly between my UAs and ITSPs (neither relay via RTP/MediaProxy
and Asterisk RTP).</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Case 1:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>If I forward it to Asterisk and use re-INVITE in
sip.conf</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>exten => _X.,1,Answer()</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><SPAN
class=250060105-23062006><FONT face=Arial color=#0000ff size=2>exten =>
_X.,2,Dial(<A
href="mailto:SIP/Number@ITSP">SIP/Number@ITSP</A>)</FONT></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><SPAN
class=250060105-23062006><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>exten => _X.,3,Hangup</FONT></SPAN></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Asterisk actually open RTP with UAs and have the c=/m= info
because of Answer. But I faced the re-INVITE with call drop-off
issue.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>When both parties are talking, if remote phone from ITSP
hang up, things are fine. If UA hang up, remote phone is still in talking
status, and I see no BYE from Asterisk send to ITSP, even it receive BYE from
UA</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Case 2:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>If I forward it to Asterisk and not use re-INVITE in
sip.conf</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>exten => _X.,1,Answer()</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><SPAN
class=250060105-23062006><FONT face=Arial color=#0000ff size=2>exten =>
_X.,2,Dial(<A
href="mailto:SIP/Number@ITSP">SIP/Number@ITSP</A>)</FONT></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><SPAN
class=250060105-23062006><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>exten => _X.,3,Hangup</FONT></SPAN></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Everything is fine</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Case 3:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>If I forward it to Asterisk and use re-INVITE in
sip.conf</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><SPAN
class=250060105-23062006><FONT face=Arial color=#0000ff size=2>exten =>
_X.,1,Dial(<A
href="mailto:SIP/Number@ITSP">SIP/Number@ITSP</A>)</FONT></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><SPAN
class=250060105-23062006><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>exten => _X.,2,Hangup</FONT></SPAN></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Asterisk don't actually open RTP with UAs and don't have
the c=/m= info. At that time, c= and m= from UAs to Asterisk always point to
RFC1918, and also in Asterisk's memory knowledge.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>If this case, when re-INVITE happen, the re-INVITE to ITSP
contain RFC1918 IP, and cause wrong media path.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Pls. advice</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Brgds</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2>Hoa</FONT></SPAN></DIV></SPAN></DIV>
<DIV dir=ltr align=left> </DIV>
<DIV dir=ltr align=left><SPAN class=250060105-23062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> Greger V. Teigre
[mailto:greger@teigre.com] <BR><B>Sent:</B> Thursday, June 22, 2006 7:02
PM<BR><B>To:</B> Hoa Thai Duy<BR><B>Cc:</B>
serusers@iptel.org<BR><B>Subject:</B> Re: [Serusers] Anyway to steal Media
IP/Port from RTPProxy or MediaProxy<BR></FONT><BR></DIV>
<DIV></DIV>I would think you are better off forwarding the INVITE to
Asterisk?!<BR>g-)<BR><BR>Hoa Thai Duy wrote:
<BLOCKQUOTE cite=mid01a101c695d5$ca70a370$fa03a8c0@hangla type="cite">
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<P><FONT face=Arial size=2>Hi List</FONT> </P>
<P><FONT face=Arial size=2>I want to get to c= and m= value after
use_media_proxy or force_rtp_proxy (after real RTP flow between UA and
media/rtpproxy)</FONT></P>
<P><FONT face=Arial size=2>I want this in order to steal this pair of
information, and bypass the RTPProxy/MediaProxy and use this information for
UA to talk with other application server (eg. Asterisk)</FONT></P>
<P><FONT face=Arial size=2>Pls. help</FONT> </P>
<P><FONT face=Arial size=2>Brgds</FONT> </P><BR><PRE wrap=""><HR width="90%" SIZE=4>
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