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 <http://2.3.4.5:5060> ->
2.3.4.5:5060 <http://2.3.4.5:5060>
INVITE sip:1234567890@kamailio-server.org
<mailto:sip%3A1234567890@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.07799e3eb71f33a9ef91178ac760ebd2.0
Via: SIP/2.0/UDP
2.3.4.5;branch=z9hG4bKsr-BnCVyrWoUladI14pU7Pry-Pzy-ONy-VSQ74NU7HzeYazq2nFU2Oc5FIWiGIKq-HXex1oONpam-C6IrIXkr4FQxZRh7sM
Max-Forwards: 69
From: <sip:subscriber@5.4.3.2:5061
<http://sip:subscriber@5.4.3.2:5061>>;tag=as491cec82
To: <sip:1234567890@kamailio-server.org
<mailto:sip%3A1234567890@kamailio-server.org>>
Contact:
<sip:2.3.4.5;line=sr-eNC05xhKefQnON1EglFrOkF0OfpzV2s9UzM9UXMNQXMn5-B0Q-s*>
Call-ID: 4e5d22c3369704b97d866c8d0f3798f8@192.168.201.2:5061
<http://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
<mailto:sip%3A9876543210@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 172.17.130.13 59426 typ host
a=candidate:igct4zWHEzhGCjWc 2 UDP 2130706430 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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