[SR-Users] RTPproxy on Kamailio 3.2.1 difficulty.

[Sammy Govind]

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.


On Mon, Jan 9, 2012 at 8:39 PM, nunu abe <nunu_abe at yahoo.com> wrote:

>
>
> 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.
>
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