Greetings all. I have following use case:
0) Peer A and B registers on kamailio via TLS. 1) Peer A establishes call to peer B. 2) Peer A changes network and call has to be refreshed. Peer A reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer.
The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering.
Is there way to solve or avoid this situation?
Thank you :)
The SIP reply is routed based on the address in the VIA header. The address in VIA is set by each node sending a request, if between the request and the response the sender is no longer available at the address it added as VIA, then it is little one can do for that.
You can try not sending the 100 trying, so the sender does retransmissions. Also, maybe the sender should send a new re-invite as it changes to a new network again.
Cheers, Daniel
On 11/08/14 13:10, Dmytro Bogovych wrote:
Greetings all. I have following use case:
- Peer A and B registers on kamailio via TLS.
- Peer A establishes call to peer B.
- Peer A changes network and call has to be refreshed. Peer A
reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer.
The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering.
Is there way to solve or avoid this situation?
Thank you :)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
You can try not sending the 100 trying, so the sender does retransmissions
Sender did not receive 100 answer; anyway it did not retransmit. It is based on resiprocate.
maybe the sender should send a new re-invite as it changes to a new
network again. Is it legal in SIP? I ask because such attempt violates resiprocate's invite session state machine rules; resiprocate will throw exception.
On Mon, Aug 11, 2014 at 3:17 PM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
The SIP reply is routed based on the address in the VIA header. The address in VIA is set by each node sending a request, if between the request and the response the sender is no longer available at the address it added as VIA, then it is little one can do for that.
You can try not sending the 100 trying, so the sender does retransmissions. Also, maybe the sender should send a new re-invite as it changes to a new network again.
Cheers, Daniel
On 11/08/14 13:10, Dmytro Bogovych wrote:
Greetings all. I have following use case:
- Peer A and B registers on kamailio via TLS.
- Peer A establishes call to peer B.
- Peer A changes network and call has to be refreshed. Peer A
reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer.
The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering.
Is there way to solve or avoid this situation?
Thank you :)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
On 11/08/14 15:18, Dmytro Bogovych wrote:
You can try not sending the 100 trying, so the sender does retransmissions
Sender did not receive 100 answer; anyway it did not retransmit. It is based on resiprocate.
Ahh, right, for tls the reliability of transmission is given by transport layer. So no much to try on this option.
maybe the sender should send a new re-invite as it changes to a new
network again. Is it legal in SIP? I ask because such attempt violates resiprocate's invite session state machine rules; resiprocate will throw exception.
I see no problem from sip point of view -- each party can send re-invites as they have something new to negotiate.
Cheers, Daniel
On Mon, Aug 11, 2014 at 3:17 PM, Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com> wrote:
The SIP reply is routed based on the address in the VIA header. The address in VIA is set by each node sending a request, if between the request and the response the sender is no longer available at the address it added as VIA, then it is little one can do for that. You can try not sending the 100 trying, so the sender does retransmissions. Also, maybe the sender should send a new re-invite as it changes to a new network again. Cheers, Daniel On 11/08/14 13:10, Dmytro Bogovych wrote:Greetings all. I have following use case: 0) Peer A and B registers on kamailio via TLS. 1) Peer A establishes call to peer B. 2) Peer A changes network and call has to be refreshed. Peer A reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer. The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering. Is there way to solve or avoid this situation? Thank you :) _______________________________________________ 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-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA _______________________________________________ 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
I see no problem from sip point of view -- each party can send re-invites
as they have something new to negotiate.
Will try to patch resiprocate :) Thanks for your help !
On Mon, Aug 11, 2014 at 4:47 PM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
On 11/08/14 15:18, Dmytro Bogovych wrote:
You can try not sending the 100 trying, so the sender does
retransmissions Sender did not receive 100 answer; anyway it did not retransmit. It is based on resiprocate.
Ahh, right, for tls the reliability of transmission is given by transport layer. So no much to try on this option.
maybe the sender should send a new re-invite as it changes to a new
network again. Is it legal in SIP? I ask because such attempt violates resiprocate's invite session state machine rules; resiprocate will throw exception.
I see no problem from sip point of view -- each party can send re-invites as they have something new to negotiate.
Cheers, Daniel
On Mon, Aug 11, 2014 at 3:17 PM, Daniel-Constantin Mierla < miconda@gmail.com> wrote:
The SIP reply is routed based on the address in the VIA header. The address in VIA is set by each node sending a request, if between the request and the response the sender is no longer available at the address it added as VIA, then it is little one can do for that.
You can try not sending the 100 trying, so the sender does retransmissions. Also, maybe the sender should send a new re-invite as it changes to a new network again.
Cheers, Daniel
On 11/08/14 13:10, Dmytro Bogovych wrote:
Greetings all. I have following use case:
- Peer A and B registers on kamailio via TLS.
- Peer A establishes call to peer B.
- Peer A changes network and call has to be refreshed. Peer A
reregisters, gathers ICE candidates, builds reINVITE and sends it to B. After peer A changes network again, reregisters and waits for answer from old reINVITE. 3) Peer B receives reINVITE and sends answer.
The problem is this answer never gets to peer B. Kamailio cannot route answer as old TLS connection is closed and new one did not established before reregistering.
Is there way to solve or avoid this situation?
Thank you :)
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany ::: Oct 15-17, San Francisco, USA
-
Daniel-Constantin Mierla -
Dmytro Bogovych