[SR-Users] Server silently droping Bye message.
Helena Garcia-Nieto
helena.gnieto at morodo.co.uk
Thu Aug 29 15:31:49 CEST 2013
Sorry I thought I attached the complete trace, I captured it with wireshark
on the server machine so I hope it is enough since you can see both sides.
Yes I realized the alias is added on the invite from the Kamailio to the
PSTN GW:
Contact:
<sip:AndroidAnumber at PrivateAndroidIP:35986;transport=tcp>;alias=PublicAndroi
dIP~35986~2
but it is not present on the BYE the PSTNGW send back to kamailio.
But kamailio has the information on the database, so I though it may be
able to send the message anyway.
Thanks
Helena
From: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] On Behalf Of
Daniel-Constantin Mierla
Sent: jueves, 29 de agosto de 2013 14:44
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Server silently droping Bye message.
Hello,
the BYE does not have information about the public ip of the nat router
behind which is located the android device.
With the default config of kamailio, that should be stored as alias
parameter in the contact header.
You should send the full signaling from first invite to the bye for the case
that does not work, in order to see if something goes wrong there and what
party is responsible for that.
You should use ngrep to take the trace on kamailio server, because it easy
to read and captures the both sides. A sample command would be:
ngrep -d any -qt -W byline port 5060
Of course you can replace IP addresses with suggestive tokens.
Cheers,
Daniel
On 8/29/13 2:00 PM, Helena Garcia-Nieto wrote:
Hello,
Thanks in advanced for the help. I am absolutely new with kamailio and still
struggling through silly problems so please forgive me if the solutions is
so obvious.
I have a network like
Devices --à Kamailio --à pstn GW
The PSTN GW is more or less out of my reach for changing the behaivour.
As devices I have xlite, 3cx and an android app
Kamailio is on version 4.0.2
I have problems with the calls to PSTN, When A is the Android app the server
is not able to relay the Bye coming from the pstngw so the android app does
not disconnect the call. This do not happened when A are xlite or 3cx. Bye
is correctly sent to the PSTNGW if it comes from the android app. For calls
inside the kamailio server there is no problem with the devices and all the
byes are sent ok.
When I call from the android app I have to drop some of the parameters on
the Contact header because the pstngw do not like them and drop the call
otherwise (responding with a 500 IRP error) I droped the parameters using:
subst('/^Contact: (.*)>.*$/Contact: \1>/i'); #delete the parameter that
is anoying pstnGW on contact header
I also change request uri deleting the 00, but I do this modification on all
the calls. I have tried without this ruri modification and the resoult is
the same xlite ok android ko so I do not think that is affecting.
subst_uri('/^sip:00(.*)/sip:\1/i'); # delete the 00
Searching on the kamailio log I found that in the android call:
DEBUG:
tm [t_lookup.c:1395]: t_newtran(): DEBUG: t_newtran: msg id=113 , global msg
id=112 , T on entrance=0xffffffffffffffff
DEBUG:
tm [t_lookup.c:534]: t_lookup_request(): t_lookup_request: start searching:
hash=30121, isACK=0
DEBUG:
tm [t_lookup.c:492]: matching_3261(): DEBUG: RFC3261 transaction matching
failed
DEBUG:
tm [t_lookup.c:716]: t_lookup_request(): DEBUG: t_lookup_request: no
transaction found
DEBUG:
tm [t_hooks.c:374]: run_reqin_callbacks_internal(): DBG:
trans=0x7f60cadc8d10, callback type 1, id 0 entered
While on the xlite call I can see
tm [t_lookup.c:1395]: t_newtran(): DEBUG: t_newtran: msg id=116 , global msg
id=115 , T on entrance=(nil)
tm [t_lookup.c:534]: t_lookup_request(): t_lookup_request: start searching:
hash=64449, isACK=0
tm [t_lookup.c:477]: matching_3261(): DEBUG: RFC3261 transaction matched,
tid=1cbf.617b985b6620c855b501dfbfced5f7a4.0
tm [t_lookup.c:733]: t_lookup_request(): DEBUG: t_lookup_request:
transaction found (T=0x7f60cadb2510)
When processing the BYE. So I’m thinking that the problem is that Kamailio
do not know where to send it. I’m not sure that is the problem.
I’ve checked and tags and session information seem ok to me, so I do not
understand what is wrong with the Bye so the server cannot find where it
should be sent to.
I attached a trace with both calls first xlite and then android
This is the Bye on the XLite call
Frame 59: 788 bytes on wire (6304 bits), 788 bytes captured (6304 bits)
Ethernet II, Src: Cisco_67:50:00 (macAdd), Dst: FujitsuT_80:7d:d8 (macAdd)
Internet Protocol Version 4, Src: PSTNGWIP (PSTNGWIP), Dst: ServerIP
(ServerIP)
User Datagram Protocol, Src Port: sip (5060), Dst Port: ServerPort
(ServerPort)
Session Initiation Protocol (BYE)
Request-Line: BYE sip:XliteAnumber at publicXliteIP:55838;transport=tcp
SIP/2.0
Method: BYE
Request-URI: sip:XliteAnumber at publicXliteIP:55838;transport=tcp
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP
PSTNGWIP;branch=z9hG4bK5d4.f93633a612aca237ad1f5c341c1b2986.0
Via: SIP/2.0/UDP
PSTNGWIP:5061;branch=z9hG4bK-fat3ctwlimdw3lkb;rport=5061
Route: <sip:ServerIP:ServerPort;r2=on;lr=on;nat=yes>
<sip:ServerIP:ServerPort;r2=on;lr=on;nat=yes>
Route: <sip:ServerIP:ServerPort;transport=tcp;r2=on;lr=on;nat=yes>
<sip:ServerIP:ServerPort;transport=tcp;r2=on;lr=on;nat=yes>
Max-Forwards: 16
From: 00Bnumber <sip:00Bnumber at ServerIP:ServerPort>
<sip:00Bnumber at ServerIP:ServerPort>;tag=eihmt4yopie2j4fq.i
To: <sip:XliteAnumber at ServerIP:ServerPort>
<sip:XliteAnumber at ServerIP:ServerPort>;tag=2b8f312e
Call-ID: NGZlMDU3OWE2YjM1ZGQ1OTEyM2NiNmI0YjM1MjA0YTg
CSeq: 515 BYE
Sequence Number: 515
Method: BYE
Contact: Anonymous <sip:PSTNGWIP:5061> <sip:PSTNGWIP:5061>
User-Agent: Sippy
cisco-GUID: 521708479-278139363-3214082078-3387785402
h323-conf-id: 521708479-278139363-3214082078-3387785402
and that one is the one on the Android one:
Frame 115: 766 bytes on wire (6128 bits), 766 bytes captured (6128 bits)
Ethernet II, Src: Cisco_67:50:00 (macAdd), Dst: FujitsuT_80:7d:d8 (macAdd)
Internet Protocol Version 4, Src: PSTNGWIP (PSTNGWIP), Dst: ServerIP
(ServerIP)
User Datagram Protocol, Src Port: sip (5060), Dst Port: ServerPort
(ServerPort)
Session Initiation Protocol (BYE)
Request-Line: BYE
sip:AndroidAnumber at PrivateAndroidIP:35986;transport=tcp SIP/2.0
Method: BYE
Request-URI: sip:AndroidAnumber at PrivateAndroidIP:35986;transport=tcp
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP
PSTNGWIP;branch=z9hG4bKf0a2.679a023eeebfc2b9acae8c683386146f.0
Via: SIP/2.0/UDP
PSTNGWIP:5061;branch=z9hG4bK-ge57vxpelnqy57yg;rport=5061
Route: <sip:ServerIP:ServerPort;r2=on;lr=on;nat=yes>
<sip:ServerIP:ServerPort;r2=on;lr=on;nat=yes>
Route: <sip:ServerIP:ServerPort;transport=tcp;r2=on;lr=on;nat=yes>
<sip:ServerIP:ServerPort;transport=tcp;r2=on;lr=on;nat=yes>
Max-Forwards: 16
From: <sip:00Bnumber at ServerIP:ServerPort>
<sip:00Bnumber at ServerIP:ServerPort>;tag=2yapycnnrb7yjgwj.i
To: <sip:AndroidAnumber at ServerIP:ServerPort>
<sip:AndroidAnumber at ServerIP:ServerPort>;tag=652340227
Call-ID: fa8e5892-e88f-fd9b-8f99-0545ac8e9e8e
CSeq: 438 BYE
Sequence Number: 438
Method: BYE
Contact: Anonymous <sip:PSTNGWIP:5061> <sip:PSTNGWIP:5061>
User-Agent: Sippy
cisco-GUID: 521708482-278139363-3214082078-3387785402
h323-conf-id: 521708482-278139363-3214082078-3387785402
Both Devices are behind NAT.
I hope I explained my problem correctly but do not hesitate to ask any other
thing.
Thanks in advanced
Helena
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--
Daniel-Constantin Mierla - http://www.asipto.com
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Kamailio Advanced Trainings - Berlin, Oct 21-24; Miami, Nov 11-13, 2013
- more details about Kamailio trainings at http://www.asipto.com -
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