[SR-Users] Fwd: Can Kamailio be used to redirect media between a client that switches from wifi to 3g/gsm

[Shaun Clark]

Forgot to post the response to the list as well.

Date: Fri, Jun 22, 2012 at 6:57 AM
Subject: Re: [SR-Users] Can Kamailio be used to redirect media between a
client that switches from wifi to 3g/gsm
To: Klaus Darilion <klaus.mailinglists at pernau.at>


Thanks for the response! I see a series of what I believe are re-REGISTER
statements:

Message sent: (to dest=75.101.244.XXX:5060)
REGISTER sip:75.101.244.XXX SIP/2.0
Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK1839704852
From: <sip:990XX at 75.101.244.XXX>;tag=1689684502
To: <sip:990XX at 75.101.244.XXX>
Call-ID: 1867622191
CSeq: 1 REGISTER
Contact: <sip:990XX at 10.165.27.161:2407;line=daeb0d9351eff22>
Max-Forwards: 70
User-Agent: Linphone/3.4.0 (eXosip2/unknown)
Expires: 3600
Content-Length: 0

Received message:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61
;rport=2407;branch=z9hG4bK1839704852
From: <sip:990XX at 75.101.244.XXX>;tag=1689684502
To: <sip:990XX at 75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.e10d
Call-ID: 1867622191
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="75.101.244.XXX",
nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX"
Server: Kamailio
Content-Length: 0

REGISTER sip:75.101.244.XXX SIP/2.0
Via: SIP/2.0/UDP 10.165.27.161:2407;rport;branch=z9hG4bK123406454
From: <sip:990XX at 75.101.244.XXX>;tag=1689684502
To: <sip:990XX at 75.101.244.XXX>
Call-ID: 1867622191
CSeq: 2 REGISTER
Contact: <sip:990XX at 10.165.27.161:2407;line=daeb0d9351eff22>
Authorization: Digest username="990XX", realm="75.101.244.XXX",
nonce="4fe476e500000e00aa8700d49b8c668f4c6f1e6a367f2XXX",
uri="sip:75.101.244.XXX", response="1e1d558894f2c05c322c76efbb2f9XXX",
algorithm=MD5
Max-Forwards: 70
User-Agent: Linphone/3.4.0 (eXosip2/unknown)
Expires: 3600
Content-Length: 0

Received message:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.165.27.161:2407;received=32.158.143.61
;rport=2407;branch=z9hG4bK123406454
From: <sip:990XX at 75.101.244.XXX>;tag=1689684502
To: <sip:990XX at 75.101.244.XXX>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8b5a
Call-ID: 1867622191
CSeq: 2 REGISTER
Contact: <sip:990XX at 10.165.27.161:2407;line=daeb0d9351eff22>;expires=120,
<sip:990XX at 50.43.101.83:51879;line=59ecc207f06f4e9>;expires=81
Server: Kamailio
Content-Length: 0

But after this I would expect to see an INVITE but one is never sent, but
if I switch back to the original IP on that device the call is reconnected,
so it proves we're missing an INVITE I believe. What do I need to do on the
server side to force a re-INVITE to be sent after this registration occurs?
Thanks!

On Fri, Jun 22, 2012 at 1:14 AM, Klaus Darilion <
klaus.mailinglists at pernau.at> wrote:

> Hi Shaun!
>
> Your problem description is too short to give you any good help.
>
> Use tcpdump (or other tools) to capture the scenario with Asterisk and
> Kamailio. Then compare them to find out why it doesn't work.
>
> Is media sent directly to Asterisk then it ca not be the problem of
> Kamailio.
>
> I hope the mobile client is smart enough to also send a reINVITE when
> getting the new IP address (of the mobile connection) with proper Contact
> header - otherwise it can not receive SIP requests from Asterisk.
>
> regards
> Klaus
>
>
> On 20.06.2012 18:07, Shaun Clark wrote:
>
>> The use case is that I have a SIP client registered to Kamailio talking
>> to an Asterisk box connected to the PSTN. The client is a mobile phone
>> and the user is connected to wifi. The user then steps out of wifi range
>> and the phone drops the connection and picks up the 3g data connection.
>> I want the media stream to reconnect to the client and the call to
>> resume without having to redial. This works now if the client is
>> directly connected to the Asterisk machine, but not when I am routing
>> through my Kamailio server. How do I go about this, examples are always
>> appreciated, thanks!
>>
>>
>> ______________________________**_________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>>
>>
>
>


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