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@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@kamailio1:5061>;tag=1848614912
To: "0uac2num" <sip:0uac2num@kamailio1;user=phone>
Call-ID: 228601536217236-500684316713@uac1ip
CSeq: 2 INVITE
Contact: <sip:uac1@uac1ip:17762>
Proxy-Authorization: Digest username="uac1",
realm="kamailio1", nonce="Vwd5PlcHeBLSAh0pqSU3AvKzvNAD+QGC",
uri="sip:0uac2num@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
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http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 -
http://www.kamailioworld.com
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