Hello Daniel,
Thanks for the reply. That is correct, looping was not my intention, I'm
trying to figure out what's causing it...
BR,
George
On 31 July 2017 at 17:29, Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Hello,
usually you should not loop requests locally, unless a very special case
-- do you do the loop routing because of a need or just happens but you
don't know why?
Cheers,
Daniel
On 31.07.17 13:46, George Diamantopoulos wrote:
Hello all,
I have been toying with kamailio lately, and I thought I had gotten to
the point where I had a mostly working (tm) prototype. I gave it a test
drive with some real calls, however, and an issue manifested at least once,
where homer received packets originating from the kamailio host, and whose
destination was also the kamailio host.
The dialog this manifested in is a generally problematic one, with many
retransmissions occurring because of slow database access (I haven't
implemented htable caching yet). No packet capture over the network is
actually taking place, I'm copying everything to homer with
"trace_mode"set
to 1.
Homer shows these messages like in the screenshot:
https://imagebin.ca/v/3VGJAmovRBmo. Here's an example of a packet:
2017-07-28 14:32:29 +0300 : 2.3.4.5:5060 -> 2.3.4.5:5060
INVITE sip:1234567890@kamailio-server.org SIP/2.0
Record-Route: <sip:2.3.4.5;lr;ftag=as491cec82>
Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bK0cf.0779
9e3eb71f33a9ef91178ac760ebd2.0
Via: SIP/2.0/UDP 2.3.4.5;branch=z9hG4bKsr-BnCVy
rWoUladI14pU7Pry-Pzy-ONy-VSQ74NU7HzeYazq2nFU2Oc5FIWiGIKq-HXe
x1oONpam-C6IrIXkr4FQxZRh7sM
Max-Forwards: 69
From: <sip:subscriber@5.4.3.2:5061>;tag=as491cec82
To: <sip:1234567890@kamailio-server.org>
Contact: <sip:2.3.4.5;line=sr-eNC05xhKefQnON1EglFrOkF0OfpzV2s9UzM9UXM
NQXMn5-B0Q-s*>
Call-ID: 4e5d22c3369704b97d866c8d0f3798f8@192.168.201.2:5061
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2~bpo70+1
Date: Fri, 28 Jul 2017 11:32:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "9876543210" <sip:9876543210@192.168.201.2>
;party=calling;privacy=off;screen=yes
Content-Type: application/sdp
Content-Length: 500
v=0
o=root 242468242 242468243 IN IP4 172.17.130.13
s=Asterisk PBX 11.13.1~dfsg-2~bpo70+1
c=IN IP4 172.17.130.13
t=0 0
m=audio 59426 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:50
a=sendrecv
a=rtcp:59427
a=ice-ufrag:hffIoy6x
a=ice-pwd:nXB8ip7Qz8ZG5yyZRr97kGJbej
a=candidate:igct4zWHEzhGCjWc 1 UDP 2130706431 <21%203070%206431>
172.17.130.13 59426 typ host
a=candidate:igct4zWHEzhGCjWc 2 UDP 2130706430 <21%203070%206430>
172.17.130.13 59427 typ host
Can something like this be triggered by misconfiguration in the routing
scripts? Should it worry me and should I dig into it more? Could it be a
bug of the siptrace module and nothing bad actually took place? I'm not
sure where to start with this, so any input would be greatly appreciated.
Thanks!
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