[Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS

toly hippo_big at hotmail.com
Fri Sep 5 03:48:11 CEST 2008




Klaus Darilion-2 wrote:
> 
> 
> 
> 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 at pernau.at] 
>> Sent: 2008-08-29 14:34
>> To: Ali Jawad
>> Cc: users at lists.kamailio.org
>> Subject: Re: [Kamailio-Users] FW: Call Hangup from Caller-End not
>> working with TLS
>> 
>> 1. INVITE:
>> 
>> Contact: 
>> <sip:username at 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:
>>> 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 at pernau.at] 
>>> Sent: 2008-08-28 18:13
>>> To: Ali Jawad
>>> Cc: users at 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:
>>>> 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
>> mode
>>>> 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:     
>>>>
>> <sip:michofr at 193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200
>>> 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
>>>>
>>>>
>>>>
>> ------------------------------------------------------------------------
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.kamailio.org
>>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> 
> 
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> 
> 

If you make tls/tcp call from sip client having tls/tcp transport to GW
which has udp transport, you have to add transport=tls before you forward
bye back to client.

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