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<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff size=2>Dear expert Greger</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff 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 face=Arial
color=#0000ff size=2>In case 2, It does, and remote phone hangup
normally.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff 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><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff 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><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff size=2>Tks & brgds</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff size=2>Hoa</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=390222005-26062006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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<FONT face=Tahoma size=2><B>From:</B> Greger V. Teigre
[mailto:greger@teigre.com] <BR><B>Sent:</B> Friday, June 23, 2006 9:54
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>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">
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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></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></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></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></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></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></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></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></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></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></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 [<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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<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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