[SR-Users] RTPproxy on Kamailio 3.2.1 difficulty.

[nunu abe]

Hi, 


Thank you for your swift response Sammy:)

I am not sure what you meant about the tcpdump, but what I am doing is capturing packets with wireshark on the pseudo-device to get packets from both interfaces. So here is what I captured. I chopped off the messages I thought are irrelevant. This is a call from IPv4 client to IPv6 client.


INVITE sip:300 at 10.10.10.10;user=phone SIP/2.0 
Via: SIP/2.0/UDP 30.30.30.3:1029;rport;branch=z9hG4bK-gwwnwsm8l1bu 
From: "IPv4 Client" <sip:200 at 10.10.10.10>;tag=p043g0591e 
To: <sip:300 at 10.10.10.10;user=phone> 
Call-ID: 3c267bf62be0-7ffgdvptawad 
CSeq: 1 INVITE 
Max-Forwards: 70 
Contact: <sip:200 at 30.30.30.3:1029;line=nv9lxq4g>;reg-id=1 
P-Key-Flags: resolution="31x13", keys="4" 
User-Agent: snom370/7.3-boco-test 
Accept: application/sdp 
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO 
Allow-Events: talk, hold, refer, call-info 
Supported: timer, replaces, from-change 
Session-Expires: 3600;refresher=uas 
Min-SE: 90 
Content-Type: application/sdp 
Content-Length: 678 
v=0 
o=root 991959232 991959232 IN IP4 30.30.30.3 
s=call 
c=IN IP4 30.30.30.3
t=0 0 
m=audio 55512 RTP/SAVP 8 9 99 3 18 4 101 
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:z2sV7JQKuY0lQ9+6pwyMw9v9g7/ExmM1oVxKiMAM 
a=rtpmap:8 pcma/8000 
a=rtpmap:9 g722/8000 
a=rtpmap:99 g726-32/8000 
a=rtpmap:3 gsm/8000 
a=rtpmap:18 g729/8000 
a=rtpmap:4 g723/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=ptime:20 
a=sendrecv 
m=audio 55512 RTP/AVP 8 9 99 3 18 4 101 
a=rtpmap:8 pcma/8000 
a=rtpmap:9 g722/8000 
a=rtpmap:99 g726-32/8000 
a=rtpmap:3 gsm/8000 
a=rtpmap:18 g729/8000 
a=rtpmap:4 g723/8000 
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-16 
a=ptime:20 
a=sendrecv



INVITE sip:300@[3001:0:0:4:0:0:0:4]:5060;line=fb0371cc0525bb2 SIP/2.0 
Record-Route: <sip:[3001:0:0:1:0:0:0:10];r2=on;lr=on;nat=v46> 
Record-Route: <sip:10.10.10.10;r2=on;lr=on;nat=v46> 
Via: SIP/2.0/UDP [3001:0:0:1:0:0:0:10];branch=z9hG4bK6e01.90956496.0 
Via: SIP/2.0/UDP 30.30.30.3:1029;rport=1028;branch=z9hG4bK-uzp7y9vq7nma 
From: "IPv4 Client" <sip:200 at 10.10.10.10>;tag=p043g0591e 
To: <sip:300 at 10.10.10.10;user=phone> 
Call-ID: 3c267bf62be0-7ffgdvptawad 
CSeq: 2 INVITE 
Max-Forwards: 69 
Contact: <sip:200 at 30.30.30.3:1028;line=nv9lxq4g>;reg-id=1 
P-Key-Flags: resolution="31x13", keys="4"                          ******here expanding the wireshark message, I see "Unrecognised SIP header"  ******
User-Agent: snom370/7.3-boco-test 
Accept: application/sdp 
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO 
Allow-Events: talk, hold, refer, call-info 
Supported: timer, replaces, from-change 
Session-Expires: 3600;refresher=uas 
Min-SE: 90 
Content-Type: application/sdp 
Content-Length: 704 

v=0 
o=root 991959232 991959232 IN IP6 3001:0:0:1::10 
s=call 
c=IN IP6 3001:0:0:1::10 
t=0 0 
m=audio 38450 RTP/SAVP 8 9 99 3 18 4 101 
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:z2sV7JQKuY0lQ9+6pwyMw9v9g7/ExmM1oVxKiMAM 
.
a=sendrecv 
a=nortpproxy:yes 

********************This is an ICMPv6 message type 2 = "Too big".***************
INVITE sip:300@[3001:0:0:4:0:0:0:4]:5060;line=fb0371cc0525bb2 SIP/2.0 
Record-Route: <sip:[3001:0:0:1:0:0:0:10];r2=on;lr=on;nat=v46> 
Record-Route: <sip:10.10.10.10;r2=on;lr=on;nat=v46> 
Via: SIP/2.0/UDP [3001:0:0:1:0:0:0:10];branch=z9hG4bK6e01.90956496.0 
Via: SIP/2.0/UDP 30.30.30.3:1029;rport=1028;branch=z9hG4bK-uzp7y9vq7nma 
From: "IPv4 Client" <sip:200 at 10.10.10.10>;tag=p043g0591e 
To: <sip:300 at 10.10.10.10;user=phone> 
Call-ID: 3c267bf62be0-7ffgdvptawad 
CSeq: 2 INVITE 
Max-Forwards: 69 
Contact: <sip:200 at 30.30.30.3:1028;line=nv9lxq4g>;reg-id=1 
P-Key-Flags: resolution="31x13", keys="4"                        ******here expanding the wireshark message, I see "Unrecognised SIP header".******
User-Agent: snom370/7.3-boco-test 
Accept: application/sdp 
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO 
Allow-Events: talk, hold, refer, call-info 
Supported: timer, replaces, from-change 
Session-Expires: 3600;refresher=uas 
Min-SE: 90 
Content-Type: applicat


These messages keep repeating until the caller receives request timeout response from kamailio.


Thank you for your help :)

Regards,
Maedot.
________________________________
From: Sammy Govind <govoiper at gmail.com>
To: nunu abe <nunu_abe at yahoo.com> 
Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <sr-users at lists.sip-router.org> 
Sent: Monday, January 9, 2012 3:43 PM
Subject: Re: [SR-Users] RTPproxy on Kamailio 3.2.1 difficulty.


Hi again,

How are you taking traces on Kamailio+rtpproxy server !?
Since it has multiple interfaces and SIP packets maybe too big for default packet length in capture so what i do is.

#tcpdump -i any -s 0 -w maycapture.pcap -vvvvv
-i any [listens to both interfaces for traffic] 
-s 0 [let the length of each packet captured reach infinity :) ]


Also check for tcpdump params for IPv6 special flags if any.

Paste the new SIP traces. 

Regards.
Sammy.