29 Aug
2008
29 Aug
'08
3:38 p.m.
Ali Jawad schrieb:
Hi Klaus
You are referring to line 17 right ? That part of traffic is from
openser to the pstn gw ..and both of those are UDP..should it be
transport=tls there or transport=udp?
The Contact is the contact of the caller. Thus, there should be the
IP:socket:protocol which is used by the caller (TLS).
klaus
Thanks
With Regards
Ali Jawad
System Administrator
Splendor Telecom (www.splendor.net)
Beirut, Lebanon
Phone: +961 1 373725
Fax: +961 1 375554
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-29 14:34
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
1. INVITE:
Contact:
<sip:username@IP.OF.LAN.GW:2949;transport=UDP;rinstance=D98C1DD404B2008F
980980E97E42F8EC;nat=yes>.
As you see the caller announces UDP as contact. Either a bug in the
caller client or do you rewrite the contact in openser?
Further it is strange that the caller sends frmo 127.0.0.1 (Via header,
SDP) but announces a different IP in contact.
regards
klaus
Ali Jawad schrieb:
<sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
Dear All
Please find below the call setup from my softphone to my cell phone,
The
setup is as follows:
LAN -> Office Gateway <--TLS--->Openser<--UDP-->PSTN GW
I got the trace below by applying
ngrep -W byline -T username -q -d eth0
http://pastebin.com/m2a78a27f
I did the same trace for a udp call and it seemed identical to me, as
you can see in the lower part of the trace that a BYE packet is being
sent to the softphone however the transport is being indicated as UDP
not TLS..is this normal ?
Any clues apart from that ?
Thanks
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: 2008-08-28 18:13
To: Ali Jawad
Cc: users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
working with TLS
Hi!
Send us an ngrep dump: ngrep -P "" -W byline port 5060
Although this will show us only the UDP part (as TLS is encrypted) but
may still show as the problem.
Which SIP client do you use?
regards
klaus
Ali Jawad schrieb:
mode
Hi All
I am using using openser 1.3..if I make a call between two softphones
on
> the same lan or a a pstn call to my mobile phone..and the
> called/receiver party does hang-up the call. It works fine in UDP > and the call get's hang-up. However in
TLS mode this does not work.
> Anything I might have missed here? Since both udp and tls use the
same
routes,
and voice is fine and no one way audio ..etc.
I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in
the
logs of openser I can't see any errors.
However on the wire shark I can see icmp destination
unreachable...port
> unreachable.
>
> I would have said it is a NAT issue. However it works for simple UDP.
>
>
>
> However I did notice the following in the logs
>
>
>
> Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000
>
> Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794
> (45900000)
>
> Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler:
> retransmission_handler : done
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request:
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE>
>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri:
> 8F980980E97E42F8EC;nat=yes>
> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0>
>
> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2
>
> Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type
232,
> <branch> =
>
>
>
> Shouldn't the transport=TLS ?
>
>
>
>
>
> Regards
>
>
>
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>
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