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@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@10.10.10.10;tag=p043g0591e To: sip:300@10.10.10.10;user=phone Call-ID: 3c267bf62be0-7ffgdvptawad CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:200@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@10.10.10.10;tag=p043g0591e To: sip:300@10.10.10.10;user=phone Call-ID: 3c267bf62be0-7ffgdvptawad CSeq: 2 INVITE Max-Forwards: 69 Contact: sip:200@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@10.10.10.10;tag=p043g0591e To: sip:300@10.10.10.10;user=phone Call-ID: 3c267bf62be0-7ffgdvptawad CSeq: 2 INVITE Max-Forwards: 69 Contact: sip:200@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@gmail.com To: nunu abe nunu_abe@yahoo.com Cc: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List sr-users@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.