22 Dec
2016
22 Dec
'16
5:02 p.m.
I have tcpdump saving all SIP traffic to files. Based on the
Call-ID: 3a5e35a4-41f6-1235-7db0-000c29e59fdd
we can see at frame 5, I identified the INVITE that crashed kamailio as this one:
INVITE sip:RelayToGateway@202.173.5.181:5060 SIP/2.0
Via: SIP/2.0/UDP 202.173.5.217;rport;branch=z9hG4bK7828rtcXNgZKK
Max-Forwards: 15
From: "05068694099"
<sip:05068694099@softphone.spc.brastel.ne.jp>;tag=60tpj2K802F1r
To: <sip:6285747462406@202.173.5.181:5060>
Call-ID: 3a5e35a4-41f6-1235-7db0-000c29e59fdd
CSeq: 100817517 INVITE
Contact: <sip:mod_sofia@202.173.5.217:5060>
User-Agent: FLIP transcoder
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 293
X-FlipData:
ruri=sip:6285747462406@softphone.spc.brastel.ne.jp;carrier=8;pai_uri=sip:05068694099@softphone.spc.brastel.ne.jp
v=0
o=FreeSWITCH 1482259707 1482259708 IN IP4 202.173.5.217
s=FreeSWITCH
c=IN IP4 202.173.5.217
t=0 0
m=audio 47456 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
This is a request from FreeSWITCH acting as a transcoder and I don't see anything out
of ordinary in it.
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