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.,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.,3,Hangup
Everything is fine
Case 3:
If I forward it to Asterisk and use re-INVITE in
sip.conf
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
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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