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
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
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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) 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) 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) 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@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
_____
_______________________________________________
Serusers mailing list
Serusers@lists.iptel.org
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@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
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Dear expert Greger
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. In case 2, It does, and remote phone hangup normally.
I captured the Asterisk log, and see no differrence between the 02 BYE message in Case 1 and Case 2, but call-id.
Where could it be the issue, or it's the limitation of the Asterisk kernel?
Tks & brgds
Hoa
_____
From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Friday, June 23, 2006 9:54 PM To: Hoa Thai Duy Cc: serusers@iptel.org Subject: Re: [Serusers] Anyway to steal Media IP/Port from RTPProxy or MediaProxy
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) 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) 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) 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@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
_____
_______________________________________________
Serusers mailing list
Serusers@lists.iptel.org
I'm hardly no expert :-) And definitely not on Asterisk. I suggest you try to asterisk mailing list. g-)
Hoa Thai Duy wrote:
Dear expert Greger
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. In case 2, It does, and remote phone hangup normally.
I captured the Asterisk log, and see no differrence between the 02 BYE message in Case 1 and Case 2, but call-id.
Where could it be the issue, or it's the limitation of the Asterisk kernel?
Tks & brgds
Hoa
*From:* Greger V. Teigre [mailto:greger@teigre.com] *Sent:* Friday, June 23, 2006 9:54 PM *To:* Hoa Thai Duy *Cc:* serusers@iptel.org *Subject:* Re: [Serusers] Anyway to steal Media IP/Port from RTPProxy or MediaProxy
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@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
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Greger V. Teigre -
Hoa Thai Duy