[SR-Users] getting fragmented packet when adding record-route header

[Klaus Darilion]

Next time please send only the trace of the relevant SIP dialog (between 
provider and Kamailio/Asterisk). Ther seconds dialog started by Asterisk 
is not relevant.

The problem is rather simple:




U 2011/03/15 15:43:48.237614 6.1.1.1:5060 -> 5.1.1.1:5060
INVITE sip:1231234 at domain.com SIP/2.0
Record-Route: <sip:6.1.1.1;lr=on;ftag=B0432A3C-37B>
Via: SIP/2.0/UDP 6.1.1.1;branch=z9hG4bK4e1f.614446c3.0
Via: SIP/2.0/UDP 6.1.1.2:5060;branch=z9hG4bK47ABD036B
Remote-Party-ID: <sip:1231000 at 6.1.1.2>;party=calling;screen=yes;privacy=off
From: "1231000" <sip:1231000 at 6.1.1.2>;tag=B0432A3C-37B
To: <sip:1231234 at ire.e164.org.uk>
Date: Tue, 15 Mar 2011 15:43:48 gmt
Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B at 6.1.1.2
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: MSSGW
Allow: INVITE, BYE, CANCEL, ACK
CSeq: 101 INVITE
Max-Forwards: 14
Timestamp: 1300203828
Contact: <sip:1231000 at 6.1.1.2:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 417

v=0
o=CiscoSystemsSIP-GW-UserAgent 4797 428 IN IP4 6.1.1.2
s=SIP Call
c=IN IP4 6.1.1.2
t=0 0
m=audio 23382 RTP/AVP 8 18 4 3 98 0 101
c=IN IP4 6.1.1.2
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:98 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


U 2011/03/15 15:43:48.246612 5.1.1.1:5060 -> 6.1.1.1:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 6.1.1.1;branch=z9hG4bK4e1f.614446c3.0;rport=5060
Via: SIP/2.0/UDP 6.1.1.2:5060;branch=z9hG4bK47ABD036B
From: "1231000" <sip:1231000 at 6.1.1.2>;tag=B0432A3C-37B
To: <sip:1231234 at ire.e164.org.uk>
Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B at 6.1.1.2
CSeq: 101 INVITE
Server: kamailio (3.1.2 (i386/linux))
Content-Length: 0



U 2011/03/15 15:43:48.248371 1.2.3.3:5060 -> 1.2.3.1:5060
INVITE sip:1231234 at domain.com SIP/2.0
Record-Route: <sip:1.2.3.3;r2=on;lr=on;ftag=B0432A3C-37B>
Record-Route: <sip:5.1.1.1;r2=on;lr=on;ftag=B0432A3C-37B>
Record-Route: <sip:6.1.1.1;lr=on;ftag=B0432A3C-37B>
Via: SIP/2.0/UDP 1.2.3.3;branch=z9hG4bK4e1f.576ffdd1.0
Via: SIP/2.0/UDP 6.1.1.1;rport=5060;branch=z9hG4bK4e1f.614446c3.0
Via: SIP/2.0/UDP 6.1.1.2:5060;branch=z9hG4bK47ABD036B
Remote-Party-ID: <sip:1231000 at 6.1.1.2>;party=calling;screen=yes;privacy=off
From: "1231000" <sip:1231000 at 6.1.1.2>;tag=B0432A3C-37B
To: <sip:1231234 at ire.e164.org.uk>
Date: Tue, 15 Mar 2011 15:43:48 gmt
Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B at 6.1.1.2
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: MSSGW
Allow: INVITE, BYE, CANCEL, ACK
CSeq: 101 INVITE
Max-Forwards: 13
Timestamp: 1300203828
Contact: <sip:1231000 at 6.1.1.1:5060>

                    ^^^^^^^^^^^^^^^^

Here, Kamailio changed the received contact. As there is another proxy 
between the UAC and Kamailio, Kamailio must not modify the contact. 
(remove fix_nated_contact() for requests coming from the service provider)

By changing the contact, the BYE gets looped in the provider's Openser 
proxy until the message gets rejected due to the size.


regards
Klaus



On 15.03.2011 18:16, Asgaroth wrote:
> On 15/03/2011 14:29, Klaus Darilion wrote:
>> I prefer for ngrep traces (you could replace usernames/IP-addresses)
>
> I have attached 2 text files of ngrep traces. Both are of the same call,
> one trace was performed at the asterisk media server, and the other was
> performed at the proxy. I've replaced all user names and ip address.
>
> Just for info the IP are as follows:
>
> 1.2.3.1 = Asterisk media server (running asterisk 1.8.3)
> 1.2.3.2 = Kamailio location server (running kamailio 3.1.2)
> 1.2.3.3 = Kamailio proxy server (internal interface) (running kamailio
> 3.1.2)
> 5.1.1.1 = Kamailio proxy server (external interface) (running kamailio
> 3.1.2)
> 6.1.1.1 = Provider proxy server
> 6.1.1.2 = Provider media gateway
>
> Please let me know if you require any additional information.
>
> Thank you for taking the time to take a look at the traces.
>
>
>
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