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I'm hardly no expert :-) And definitely not on Asterisk. I suggest you
try to asterisk mailing list.<br>
g-)<br>
<br>
Hoa Thai Duy wrote:
<blockquote cite="mid005d01c698e0$edbcd6b0$fa03a8c0@hangla" type="cite">
<title>Anyway to steal Media IP/Port from RTPProxy or MediaProxy</title>
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<div dir="ltr" align="left"><span class="390222005-26062006"><font
color="#0000ff" face="Arial" size="2">Dear expert Greger</font></span></div>
<div dir="ltr" align="left"><span class="390222005-26062006"></span> </div>
<div dir="ltr" align="left"><span class="390222005-26062006"><font
color="#0000ff" face="Arial" size="2">Actually, in Case 1, the BYE
from SIP UA reached Asterisk, and Asterisk just hangup the SIP channel,
and don't send it to ITSP. </font></span></div>
<div dir="ltr" align="left"><span class="390222005-26062006"><font
color="#0000ff" face="Arial" size="2">In case 2, It does, and remote
phone hangup normally.</font></span></div>
<div dir="ltr" align="left"><span class="390222005-26062006"></span> </div>
<div dir="ltr" align="left"><span class="390222005-26062006"><font
color="#0000ff" face="Arial" size="2">I captured the Asterisk log, and
see no differrence between the 02 BYE message in Case 1 and Case 2, but
call-id.</font></span></div>
<div dir="ltr" align="left"><span class="390222005-26062006"></span> </div>
<div dir="ltr" align="left"><span class="390222005-26062006"><font
color="#0000ff" face="Arial" size="2">Where could it be the issue, or
it's the limitation of the Asterisk kernel?</font></span></div>
<div dir="ltr" align="left"><span class="390222005-26062006"></span> </div>
<div dir="ltr" align="left"><span class="390222005-26062006"><font
color="#0000ff" face="Arial" size="2">Tks & brgds</font></span></div>
<div dir="ltr" align="left"><span class="390222005-26062006"></span> </div>
<div dir="ltr" align="left"><span class="390222005-26062006"><font
color="#0000ff" face="Arial" size="2">Hoa</font></span></div>
<div dir="ltr" align="left"><span class="390222005-26062006"></span> </div>
<br>
<div class="OutlookMessageHeader" dir="ltr" align="left" lang="en-us">
<hr tabindex="-1"><font face="Tahoma" size="2"><b>From:</b> Greger V.
Teigre [<a class="moz-txt-link-freetext" href="mailto:greger@teigre.com">mailto:greger@teigre.com</a>] <br>
<b>Sent:</b> Friday, June 23, 2006 9:54 PM<br>
<b>To:</b> Hoa Thai Duy<br>
<b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:serusers@iptel.org">serusers@iptel.org</a><br>
<b>Subject:</b> Re: [Serusers] Anyway to steal Media IP/Port from
RTPProxy or MediaProxy<br>
</font><br>
</div>
Ad case 1, you may have a problem with the routing of the BYE. You just
have to track where it stops.<br>
As for the problem in inself: To me it looks more like an Asterisk
question and not a SER question?!<br>
g-)<br>
<br>
Hoa Thai Duy wrote:
<blockquote cite="mid012a01c69683$6e776b40$fa03a8c0@hangla"
type="cite">
<meta content="MSHTML 6.00.2900.2912" name="GENERATOR">
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" size="2">Dear Greger & List</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" size="2">Case 1:</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" size="2">exten => _X.,1,Answer()</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"><span
class="250060105-23062006"><font color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" size="2">exten => _X.,3,Hangup</font></span></span></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" size="2">Case 2:</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" size="2">exten => _X.,1,Answer()</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"><span
class="250060105-23062006"><font color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" size="2">exten => _X.,3,Hangup</font></span></span></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" size="2">Everything is fine</font></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
color="#0000ff" face="Arial" size="2">Case 3:</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" 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"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><span
class="250060105-23062006"><font color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" size="2">exten => _X.,2,Hangup</font></span></span></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" 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
color="#0000ff" face="Arial" 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"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" size="2">Pls. advice</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" size="2">Brgds</font></span></div>
<div dir="ltr" align="left"><span class="250060105-23062006"></span> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"><font
color="#0000ff" face="Arial" size="2">Hoa</font></span></div>
</span></div>
<div dir="ltr" align="left"> </div>
<div dir="ltr" align="left"><span class="250060105-23062006"></span> </div>
</span></div>
<br>
<div class="OutlookMessageHeader" dir="ltr" align="left"
lang="en-us">
<hr tabindex="-1"> <font face="Tahoma" size="2"><b>From:</b>
Greger V. Teigre [<a class="moz-txt-link-freetext"
href="mailto:greger@teigre.com">mailto:greger@teigre.com</a>] <br>
<b>Sent:</b> Thursday, June 22, 2006 7:02 PM<br>
<b>To:</b> Hoa Thai Duy<br>
<b>Cc:</b> <a class="moz-txt-link-abbreviated"
href="mailto:serusers@iptel.org">serusers@iptel.org</a><br>
<b>Subject:</b> Re: [Serusers] Anyway to steal Media IP/Port from
RTPProxy or MediaProxy<br>
</font><br>
</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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name="Generator">
<!-- Converted from text/rtf format -->
<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>
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</pre>
</blockquote>
</blockquote>
</blockquote>
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