Ad case 1, you may have a problem with the routing of the BYE. You just
have to track where it stops.
As for the problem in inself: To me it looks more like an Asterisk
question and not a SER question?!
g-)
Hoa Thai Duy wrote:
Dear Greger & List
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).
Case 1:
If I forward it to Asterisk and use re-INVITE in sip.conf
exten => _X.,1,Answer()
exten => _X.,2,Dial(SIP/Number@ITSP <mailto:SIP/Number@ITSP>)
exten => _X.,3,Hangup
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.
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
Case 2:
If I forward it to Asterisk and not use re-INVITE in sip.conf
exten => _X.,1,Answer()
exten => _X.,2,Dial(SIP/Number@ITSP <mailto:SIP/Number@ITSP>)
exten => _X.,3,Hangup
Everything is fine
Case 3:
If I forward it to Asterisk and use re-INVITE in sip.conf
exten => _X.,1,Dial(SIP/Number@ITSP <mailto:SIP/Number@ITSP>)
exten => _X.,2,Hangup
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.
If this case, when re-INVITE happen, the re-INVITE to ITSP contain
RFC1918 IP, and cause wrong media path.
Pls. advice
Brgds
Hoa
------------------------------------------------------------------------
*From:* Greger V. Teigre [mailto:greger@teigre.com]
*Sent:* Thursday, June 22, 2006 7:02 PM
*To:* Hoa Thai Duy
*Cc:* serusers(a)iptel.org
*Subject:* Re: [Serusers] Anyway to steal Media IP/Port from RTPProxy
or MediaProxy
I would think you are better off forwarding the INVITE to Asterisk?!
g-)
Hoa Thai Duy wrote:
>
> Hi List
>
> 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)
>
> 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)
>
> Pls. help
>
> Brgds
>
>
> ------------------------------------------------------------------------
>
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