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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