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.mailinglists@pernau.at mailto:klaus.mailinglists@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@75.101.244.XXX;tag=1689684502 To: sip:990XX@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:990XX@75.101.244.XXX;tag=1689684502 To: sip:990XX@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@75.101.244.XXX;tag=1689684502 To: sip:990XX@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:990XX@75.101.244.XXX;tag=1689684502 To: sip:990XX@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.mailinglists@pernau.at mailto:klaus.mailinglists@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@lists.sip-router.org <mailto:sr-users@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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