[SR-Users] truncating sip message
ycaner
yasin.caner at netgsm.com.tr
Mon Apr 11 08:11:36 CEST 2016
After i saw it , i search on net , yes, i got it but one thing
make me think wrong because reassabmle fragmented IPv4 datagrams
on wireshark was disabled by me ,long time ago so it is why i fell
in a block hole.
Thanks for advices
10.04.2016 00:46 tarihinde Andres-27
[via SIP Router] yazdı:
On 4/8/16 8:19 AM, ycaner wrote:
Hello;
I saw it. it is fragmented IP and cant resolve it
wireshark. But i cant understand one thing, First Call Flows
makes fragmantion but second call flows doesnt fragmantaion
even if it is larger than first call flows.
Fragmentation will
happen at the Operating System level, not at the Application
level. If the medium is Ethernet and the entire content
including IP headers exceed 1500 bytes then it gets fragmented.
Wireshark can clearly show you the packet length for any complete
or fragmented packet.
Thanks.
08.04.2016 14:40 tarihinde
Andres-27 [via SIP Router] yazdı:
On 4/8/16 5:27 AM, ycaner wrote:
Hello Daniel;
source and destination is different machines and ips.
they are sniffed by tshark and they are the same and
truncated.
Maybe they are
not truncated by rather fragmented. If you are capturing
packets by UDP Port you will not see the fragments. Try to
capture again by IP only and you will see all packets.
i tried to
remove all extra addes , include Record-Route and send
messsage , no truncate. and then started to add headers
and BOOM. after 5 headers , it is truncated. configuration
become simple to testing one by one. i couldnt find way to
solve it.
i tried to sniff with tcpdump and sdp body result here
, truncated.;
Call Flows ---
UAC1 --> Kamailio1--->kamailio2-->UAC2
--truncated sdp body
UAC1 --> Kamailio2--->UAC2 --No problem
Destination ip
tcpdump:
host-kamailio2ip.sip-tls > uac2ip.sip: SIP,
length: 1472
INVITE sip:1021uac2num at uac2ip:5060 SIP/2.0
Via: SIP/2.0/UDP
kamailio2ip:5061;branch=z9hG4bK61c7.f5edc3dbd9effe58f3d04a6b77b11f13.0
Via: SIP/2.0/UDP
kamailio1:5061;rport=5061;branch=z9hG4bK61c7.189d12e4e39ec53abb760692e8a8afee.0
Via: SIP/2.0/UDP
192.168.0.225:10251;received=uac1ip;branch=z9hG4bK702835784163333;rport=17762
From: uac1
<sip:uac1 at kamailio1:5061>;tag=1848614912
To: "0uac2num"
<sip:0uac2num at kamailio1;user=phone>
Call-ID: 228601536217236-500684316713 at uac1ip
CSeq: 2 INVITE
Contact: <sip:uac1 at uac1ip:17762>
Proxy-Authorization: Digest username="uac1",
realm="kamailio1",
nonce="Vwd5PlcHeBLSAh0pqSU3AvKzvNAD+QGC",
uri="sip:0uac2num at kamailio1;user=phone",
response="d296105a65958144cf1b60c921aef11b", algorithm=MD5
Max-Forwards: 68
Supported: replaces, join, path
User-Agent: Fanvil C58/C58P 2.2.41.13
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER,
NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 276
P-Provider:kamailio1
P-Unique:srid-5707759b-29c4-1
atik:1231345123
atik2:1231345123
atik3:1231345123
atik4:1231345123
atik5:1231345123
v=0
o=8503027337 2007313085 1038110647 IN IP4
192.168.0.225
s=A conversation
c=IN IP4 192.168.0.225
t=0 0
m=audio 10208 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:1[|sip]
08.04.2016 12:03 tarihinde
Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
how did you asserted the fact that the message is
truncated? Is the next SIP hop not retrieving it with full
content? Or just you looked at some traces on the net, if
this one, note that some network sniffing tools are having
their limitations in what they display.
Daniel
On 08/04/16 09:30, ycaner
wrote:
Hello
Daniel;
in request route when invite comes, a header is
added like P-SrcUserid. when i added new header , it
truncate sdpbody. before t_relay(), i try to remove
header, and it didnt. main problem is why truncates
sdpbody when i added new header?
Thanks.
Works fine sip messages: http://pastebin.com/t7v9c2vW
removes coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid
: http://pastebin.com/9tB7Lx3U
truncateting sip message : http://pastebin.com/DNuuJxhM
removes
coming headers cfg,but i couldnt remove P-Provider and P-Uniqueid
and added P-SrcUserid : http://pastebin.com/XxhcV55v
08.04.2016 09:22 tarihinde
Daniel-Constantin Mierla-6 [via SIP Router] yazdı:
Hello,
On 08/04/16 08:06, ycaner wrote:
> Hello;
> i think i found problem. it is textops
module.Appending new header doesnt
> increase content length. in addition , remove_hf
doesnt work correctly that
> doesnt remove headers.
>
the value of content lenght is about the body size, it
is not affected
by new headers or removing old headers.
What do you mean by remove_hf() is not working? How
you try it? Can you
send here how you use remove_hf() in config and the
incoming, plus the
outgoing sip message?
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
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