Within Kamailio there is nothing you can do to trigger a reINVITE. You
need a B2BUA (e.g. Asterisk) to force a reINVITE, and even then it is
not sure that the SIP client sends properly updated SDP and contact
header (I would try this with a manually sent reINVITE).
Further, even if there is no reINVITE, you should still have audio.
How do you handle the media stream? Is it sent directly to Asterisk? Is
there rtpproxy inbetween? (if yes, then you need the reINVITEs so that
rtpproxy will accept the new source IP address of the RTP stream (lock-in)).
regards
Klaus
On 22.06.2012 15:58, Shaun Clark wrote:
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.mailingli...@pernau.at
<mailto:klaus.mailingli...@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:99...@75.101.244.xxx>;tag=1689684502
To: <sip:99...@75.101.244.xxx>
Call-ID: 1867622191
CSeq: 1 REGISTER
Contact: <sip:990XX@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
<tel:32.158.143.61>;rport=2407;branch=z9hG4bK1839704852
From: <sip:99...@75.101.244.xxx>;tag=1689684502
To: <sip:99...@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:99...@75.101.244.xxx>;tag=1689684502
To: <sip:99...@75.101.244.xxx>
Call-ID: 1867622191
CSeq: 2 REGISTER
Contact: <sip:990XX@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
<tel:32.158.143.61>;rport=2407;branch=z9hG4bK123406454
From: <sip:99...@75.101.244.xxx>;tag=1689684502
To: <sip:99...@75.101.244.xxx>;tag=c97b4d1cb1f3d0da549e06a8d482ef63.8b5a
Call-ID: 1867622191
CSeq: 2 REGISTER
Contact:
<sip:990XX@10.165.27.161:2407;line=daeb0d9351eff22>;expires=120,
<sip:990XX@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.mailingli...@pernau.at <mailto:klaus.mailingli...@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!
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