29 Aug
2007
29 Aug
'07
3:42 p.m.
Route headers are fine - the problem is the RURI of the BYE:
See the Contact header of the INVITE:
Contact: <sip:davidloh@x.x.80.178:4294;transport=TLS>
This URI must be used in the RURI of the BYE, but Asterisk uses:
BYE sip:davidloh@x.x.80.178:4294 SIP/2.0
Thus, the proxy forwards the request with UDP instead of TLS. Thus, this
is a bug in Asterisk. Try update Asterisk. Try looking at Asterisk Bug
tracker for this bug. If you are unlucky, open a bug report on the
Asterisk bug tracker (bugs.digium.com)
regards
klaus
David Loh schrieb:
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
>
>
>
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users