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:1231...@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@6.1.1.2>;party=calling;screen=yes;privacy=off
From: "1231000" <sip:1231000@6.1.1.2>;tag=B0432A3C-37B
To: <sip:1231...@ire.e164.org.uk>
Date: Tue, 15 Mar 2011 15:43:48 gmt
Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B@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@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@6.1.1.2>;tag=B0432A3C-37B
To: <sip:1231...@ire.e164.org.uk>
Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B@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:1231...@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@6.1.1.2>;party=calling;screen=yes;privacy=off
From: "1231000" <sip:1231000@6.1.1.2>;tag=B0432A3C-37B
To: <sip:1231...@ire.e164.org.uk>
Date: Tue, 15 Mar 2011 15:43:48 gmt
Call-ID: DA4501A5-4E5111E0-A17EB0FA-C1EC011B@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@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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_______________________________________________
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