[Serusers] Anyway to steal Media IP/Port from RTPProxy or MediaProxy

[Greger V. Teigre]

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 at teigre.com]
> *Sent:* Friday, June 23, 2006 9:54 PM
> *To:* Hoa Thai Duy
> *Cc:* serusers at 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 at ITSP <mailto:SIP/Number at 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 at ITSP <mailto:SIP/Number at 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 at ITSP <mailto:SIP/Number at 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 at teigre.com]
>> *Sent:* Thursday, June 22, 2006 7:02 PM
>> *To:* Hoa Thai Duy
>> *Cc:* serusers at 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 at lists.iptel.org
>>> http://lists.iptel.org/mailman/listinfo/serusers
>>>   
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