I'm not better at IPv6, not yet atleast but if the caller is getting a timeout response and you see repeated SIP traces for IPv6 Client then the obvious is that your server is trying to route the call to IPv6 client and there is not route to destination. Thats why packets are timed out.<div>
<br><br><div class="gmail_quote">On Mon, Jan 9, 2012 at 8:39 PM, nunu abe <span dir="ltr"><<a href="mailto:nunu_abe@yahoo.com">nunu_abe@yahoo.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
<br>
Hi,<br>
<br>
<br>
Thank you for your swift response Sammy:)<br>
<br>
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.<br>
<br>
<br>
INVITE <a href="mailto:sip%3A300@10.10.10.10">sip:300@10.10.10.10</a>;user=phone SIP/2.0<br>
Via: SIP/2.0/UDP 30.30.30.3:1029;rport;branch=z9hG4bK-gwwnwsm8l1bu<br>
From: "IPv4 Client" <<a href="mailto:sip%3A200@10.10.10.10">sip:200@10.10.10.10</a>>;tag=p043g0591e<br>
To: <<a href="mailto:sip%3A300@10.10.10.10">sip:300@10.10.10.10</a>;user=phone><br>
Call-ID: 3c267bf62be0-7ffgdvptawad<br>
CSeq: 1 INVITE<br>
Max-Forwards: 70<br>
Contact: <sip:200@30.30.30.3:1029;line=nv9lxq4g>;reg-id=1<br>
P-Key-Flags: resolution="31x13", keys="4"<br>
User-Agent: snom370/7.3-boco-test<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO<br>
Allow-Events: talk, hold, refer, call-info<br>
Supported: timer, replaces, from-change<br>
Session-Expires: 3600;refresher=uas<br>
Min-SE: 90<br>
Content-Type: application/sdp<br>
Content-Length: 678<br>
v=0<br>
o=root 991959232 991959232 IN IP4 30.30.30.3<br>
s=call<br>
c=IN IP4 30.30.30.3<br>
t=0 0<br>
m=audio 55512 RTP/SAVP 8 9 99 3 18 4 101<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:z2sV7JQKuY0lQ9+6pwyMw9v9g7/ExmM1oVxKiMAM<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:9 g722/8000<br>
a=rtpmap:99 g726-32/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:18 g729/8000<br>
a=rtpmap:4 g723/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
m=audio 55512 RTP/AVP 8 9 99 3 18 4 101<br>
a=rtpmap:8 pcma/8000<br>
a=rtpmap:9 g722/8000<br>
a=rtpmap:99 g726-32/8000<br>
a=rtpmap:3 gsm/8000<br>
a=rtpmap:18 g729/8000<br>
a=rtpmap:4 g723/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=ptime:20<br>
a=sendrecv<br>
<br>
<br>
<br>
INVITE sip:300@[3001:0:0:4:0:0:0:4]:5060;line=fb0371cc0525bb2 SIP/2.0<br>
Record-Route: <sip:[3001:0:0:1:0:0:0:10];r2=on;lr=on;nat=v46><br>
Record-Route: <sip:10.10.10.10;r2=on;lr=on;nat=v46><br>
Via: SIP/2.0/UDP [3001:0:0:1:0:0:0:10];branch=z9hG4bK6e01.90956496.0<br>
Via: SIP/2.0/UDP 30.30.30.3:1029;rport=1028;branch=z9hG4bK-uzp7y9vq7nma<br>
From: "IPv4 Client" <<a href="mailto:sip%3A200@10.10.10.10">sip:200@10.10.10.10</a>>;tag=p043g0591e<br>
To: <<a href="mailto:sip%3A300@10.10.10.10">sip:300@10.10.10.10</a>;user=phone><br>
Call-ID: 3c267bf62be0-7ffgdvptawad<br>
CSeq: 2 INVITE<br>
Max-Forwards: 69<br>
Contact: <sip:200@30.30.30.3:1028;line=nv9lxq4g>;reg-id=1<br>
P-Key-Flags: resolution="31x13", keys="4" ******here expanding the wireshark message, I see "Unrecognised SIP header" ******<br>
User-Agent: snom370/7.3-boco-test<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO<br>
Allow-Events: talk, hold, refer, call-info<br>
Supported: timer, replaces, from-change<br>
Session-Expires: 3600;refresher=uas<br>
Min-SE: 90<br>
Content-Type: application/sdp<br>
Content-Length: 704<br>
<br>
v=0<br>
o=root 991959232 991959232 IN IP6 3001:0:0:1::10<br>
s=call<br>
c=IN IP6 3001:0:0:1::10<br>
t=0 0<br>
m=audio 38450 RTP/SAVP 8 9 99 3 18 4 101<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:z2sV7JQKuY0lQ9+6pwyMw9v9g7/ExmM1oVxKiMAM<br>
.<br>
a=sendrecv<br>
a=nortpproxy:yes <br>
<br>
********************This is an ICMPv6 message type 2 = "Too big".***************<br>
INVITE sip:300@[3001:0:0:4:0:0:0:4]:5060;line=fb0371cc0525bb2 SIP/2.0<br>
Record-Route: <sip:[3001:0:0:1:0:0:0:10];r2=on;lr=on;nat=v46><br>
Record-Route: <sip:10.10.10.10;r2=on;lr=on;nat=v46><br>
Via: SIP/2.0/UDP [3001:0:0:1:0:0:0:10];branch=z9hG4bK6e01.90956496.0<br>
Via: SIP/2.0/UDP 30.30.30.3:1029;rport=1028;branch=z9hG4bK-uzp7y9vq7nma<br>
From: "IPv4 Client" <<a href="mailto:sip%3A200@10.10.10.10">sip:200@10.10.10.10</a>>;tag=p043g0591e<br>
To: <<a href="mailto:sip%3A300@10.10.10.10">sip:300@10.10.10.10</a>;user=phone><br>
Call-ID: 3c267bf62be0-7ffgdvptawad<br>
CSeq: 2 INVITE<br>
Max-Forwards: 69<br>
Contact: <sip:200@30.30.30.3:1028;line=nv9lxq4g>;reg-id=1<br>
P-Key-Flags: resolution="31x13", keys="4" ******here expanding the wireshark message, I see "Unrecognised SIP header".******<br>
User-Agent: snom370/7.3-boco-test<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO<br>
Allow-Events: talk, hold, refer, call-info<br>
Supported: timer, replaces, from-change<br>
Session-Expires: 3600;refresher=uas<br>
Min-SE: 90<br>
Content-Type: applicat<br>
<br>
<br>
These messages keep repeating until the caller receives request timeout response from kamailio.<br>
<br>
<br>
Thank you for your help :)<br>
<br>
Regards,<br>
Maedot.<br>
________________________________<br>
<div class="im">From: Sammy Govind <<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>><br>
To: nunu abe <<a href="mailto:nunu_abe@yahoo.com">nunu_abe@yahoo.com</a>><br>
Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List <<a href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a>><br>
</div>Sent: Monday, January 9, 2012 3:43 PM<br>
<div class="im HOEnZb">Subject: Re: [SR-Users] RTPproxy on Kamailio 3.2.1 difficulty.<br>
<br>
<br>
</div><div class="HOEnZb"><div class="h5">Hi again,<br>
<br>
How are you taking traces on Kamailio+rtpproxy server !?<br>
Since it has multiple interfaces and SIP packets maybe too big for default packet length in capture so what i do is.<br>
<br>
#tcpdump -i any -s 0 -w maycapture.pcap -vvvvv<br>
-i any [listens to both interfaces for traffic] <br>
-s 0 [let the length of each packet captured reach infinity :) ]<br>
<br>
<br>
Also check for tcpdump params for IPv6 special flags if any.<br>
<br>
Paste the new SIP traces.<br>
<br>
Regards.<br>
Sammy. <br>
</div></div></blockquote></div><br></div>