[OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?

[David Loh]

Hi,

Arrggghh .. that's one of my attempts to eliminate the broken "BYE" 
problem... that's ngrep was captured when I set "modparam("rr", 
"enable_double_rr", "0");",
I've paste another ngrep to http://pastebin.ca/674450, this time the 
double RR header is enabled.
And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt 
the post is "openser").

Even though double RR header is enabled, but for BYE it's still doesn't 
process properly :(
For the .cfg file line #130 onward, I did tried t_relay, forward and 
force_send_socket,
but none of this will do the trick (force_send_socket was complaining 
TLS error due to missing certificate (?) )
Would appreciate if anyone could enlighten me why is this happen ?


Thanks,
David Loh



Klaus Darilion wrote:
> But the INVITE you posted at http://pastebin.ca/673392 also has only 
> one Record-Route header.
>
> regards
> klaus
>
> David Loh schrieb:
>> Hi,
>>
>> Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK 
>> ...but when asterisk disconnected the call and send BYE back to OpenSER,
>> the TLS RR header wasn't present, the only 2 RR header was 
>> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" ....
>> I'm puzzled ... is there any command to 'fix' this?
>>
>>
>> Regards,
>> David Loh
>>
>> Klaus Darilion wrote:
>>> The openser proxy should add 2 record-route header (TLS and UDP = 
>>> double record route). This is why it does not work.
>>>
>>> regards
>>> klaus
>>>
>>> David Loh schrieb:
>>>> Hi All,
>>>>
>>>> Greeting.
>>>>
>>>> I've been struggle with OpenSER TLS implementation for more than a 
>>>> week, since I've ported from UDP to TLS, everything work fine 
>>>> except the "BYE" request from Asterisk (loose route), my 
>>>> implementation was something like below:
>>>>
>>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk]
>>>>
>>>> My OpenSER.cfg already configured to listen on two port which is :- 
>>>> "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or 
>>>> even voicemail) having no problem,
>>>> but when the callee disconnect the call, caller will never get hang 
>>>> up :(
>>>>
>>>> I've attached my ethereal trace/ngrep to pastebin,
>>>> http://pastebin.ca/673392
>>>>
>>>> Wondering if anyone can help me with the broken "BYE" that returned 
>>>> from Asterisk ?
>>>> Line #131, supposedly this line should have contain 2 Via header, 
>>>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS",
>>>> but somehow the TLS via header was gone !! (compare to previous ACK 
>>>> (Line #117) /INVITE (Line #51).
>>>> Due to the missing TLS via header, OpenSER log file was complaining 
>>>> "protocol/port mis-match".
>>>>
>>>> The last BYE request (Line #256) is actually firing from Client, 
>>>> which contain the "TLS" via.
>>>>
>>>>
>>>> I've even tried "force_send_socket" to port 5061 (instead of 5060) 
>>>> from loose route, but it complaining TLS certificate error,
>>>> since Asterisk doesn't support TLS natively, I've no clue why is 
>>>> the ACK/INVITE/CANCEL work but not BYE.
>>>> if (loose_route) {
>>>> ....
>>>> if(is_method("BYE")) {   force_send_socket(IP:5061);  }
>>>> }
>>>>
>>>>
>>>> Has any one gone through of this kinda OpenSER over TLS + Asterisk 
>>>> setup,
>>>> I'm really appreciate if you can share your experience with me, or 
>>>> pin point what's the mistakes I made here.
>>>>
>>>> Thanks in advance.
>>>>
>>>> Regards,
>>>> David Loh
>>>>
>>>>
>>>>
>>>>
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>>>
>>>
>>
>>
>
>