[SR-Users] truncating sip message
ycaner
yasin.caner at netgsm.com.tr
Fri Apr 8 14:19:21 CEST 2016
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.
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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