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@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
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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