Changing route set in SIP outbound

[Avshalom Houri]

Assume that the first SIP proxy that is part of the route set (in SIP
outbound)
crashes and immediately restarts, or a backup proxy takes over. Is there
any way to keep the dialog alive if either of the
endpoints senses this failure and recreates a connection or the dialog is
doomed and needs to be
fully recreated again? The issue is that the route set includes the
connection information, which is no longer valid.

tnx
avshalom

[Worley, Dale R (Dale)]

________________________________________
From: sip-***@ietf.org [sip-***@ietf.org] On Behalf Of Avshalom Houri [***@il.ibm.com]

Assume that the first SIP proxy that is part of the route set (in SIP outbound)
crashes and immediately restarts, or a backup proxy takes over. Is there any way to keep the dialog alive if either of the
endpoints senses this failure and recreates a connection or the dialog is doomed and needs to be
fully recreated again? The issue is that the route set includes the connection information, which is no longer valid.
________________________________________

As you phrase the question, the *dialog* cannot be kept alive if the connection in question is lost because the route set cannot be changed.

Dale
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[aayush]

The route set cannot be changed mid dialog.

Regarding the endpoint detecting the crash of an intermediate proxy,there is
a periodic session refresh mechanism available in SIP which the endpoint
must support.

This mechanism is implemented by examining the Session-Expires and Min-SE
headers.

Conversely if a backup proxy takes over the crashed one, then both proxies
(Active-Stabdby) must support Virtual IP failover for implementing this
fault tolerance use case.

On Jan 3, 2011 8:59 PM, "Worley, Dale R (Dale)" <***@avaya.com> wrote:

________________________________________
From: sip-***@ietf.org [sip-***@ietf.org] On Behalf Of Avshalom
Houri [***@il.ibm.com]


Assume that the first SIP proxy that is part of the route set (in SIP
outbound)
crashes and immedia...
________________________________________

As you phrase the question, the *dialog* cannot be kept alive if the
connection in question is lost because the route set cannot be changed.

Dale
_______________________________________________
Sip mailing list https://www.ietf.org/mailman/listinfo/sip
This list is essentially closed and only used for finishing old business.
Use sip-***@cs.columbia.edu for questions on how to develop a SIP
implementation.
Use ***@ietf.org for new developments on the application of sip.
Use ***@ietf.org for issues related to maintenance of the core SIP
specifications.

[Nataraju A.B]

AFAIU, Any proxy which forms part of the route-set during call setup. It
must support backup and recovery mechanism.

One simple example where the proxy supports billing functionality could be
part of the route-set and it shall support back and recovery. In this case
it shall back back up all the necessary data to recover if there is any
failure in the active unit. Otherwise it does not make sense to to be part
of a route-set.

Post by aayush
The route set cannot be changed mid dialog.
Regarding the endpoint detecting the crash of an intermediate proxy,there
is a periodic session refresh mechanism available in SIP which the endpoint
must support.
This mechanism is implemented by examining the Session-Expires and Min-SE
headers.
Conversely if a backup proxy takes over the crashed one, then both proxies
(Active-Stabdby) must support Virtual IP failover for implementing this
fault tolerance use case.
________________________________________
Assume that the first SIP proxy that is part of the route set (in SIP outbound)
crashes and immedia...
________________________________________
As you phrase the question, the *dialog* cannot be kept alive if the
connection in question is lost because the route set cannot be changed.
Dale
_______________________________________________
Sip mailing list https://www.ietf.org/mailman/listinfo/sip
This list is essentially closed and only used for finishing old business.
_______________________________________________
Sip mailing list https://www.ietf.org/mailman/listinfo/sip
This list is essentially closed and only used for finishing old business.

[Dean Willis]

Post by Worley, Dale R (Dale)
Assume that the first SIP proxy that is part of the route set (in SIP outbound)
crashes and immediately restarts, or a backup proxy takes over. Is there any way to keep the dialog alive if either of the
endpoints senses this failure and recreates a connection or the dialog is doomed and needs to be
fully recreated again? The issue is that the route set includes the connection information, which is no longer valid.

Well, if the proxies share credentials and an IP address by some HSRP-like magic, then the dialog might reasonably stay alive. This further requires either stateless operation or state sharing ( embedded state; encoding it into the message headers is an example) between the proxies.

This is analogous to the route-set failover question in many a routing model: flow switching, MPLS, frame relay, ATM, or even Token-Ring. SIP has no magic to cure the fundamental issue. At best we can reduce it to a known problem.


--
Dean

[Worley, Dale R (Dale)]

________________________________________
From: sip-***@ietf.org [sip-***@ietf.org] On Behalf Of Avshalom Houri [***@il.ibm.com]

Assume that the first SIP proxy that is part of the route set (in SIP outbound)
crashes and immediately restarts, or a backup proxy takes over. Is there any way to keep the dialog alive if either of the
endpoints senses this failure and recreates a connection or the dialog is doomed and needs to be
fully recreated again? The issue is that the route set includes the connection information, which is no longer valid.
________________________________________

Here's an idea: When an outbound REGISTER reaches the registrar, it records the ob URI that the edge proxy created in a hidden location in the registration. As part of the Path, it inserts an ob URI that it creates that contains the registrar's hostport, and a representation of the SIP user-part and sip-instance. Now, when a new INVITE is sent to the phone, or otherwise when the ob URI is used, the request goes to the registrar, which then forwards it to one or another of the real ob URIs for the phone. Indeed, I think that the phone's GRUU can be used for the new ob URI. All that lets the selection of the flow to use to reach the phone to be made on a request-by-request basis.

Dale
_______________________________________________
Sip mailing list https://www.ietf.org/mailman/listinfo/sip
This list is essentially closed and only used for finishing old business.
Use sip-***@cs.columbia.edu for questions on how to develop a SIP implementation.
Use ***@ietf.org for new developments on the application of sip.
Use ***@ietf.org for issues related to maintenance of the core SIP specifications.