[no subject]

Is there any work being done on improving these scenarios?

//Thomas. 


-----Original Message-----
From: Brett Tate [mailto:brett@broadsoft.com] 
Sent: onsdag 5 februari 2003 23:53
To: sip@ietf.org
Subject: RE: [Sip] INVITE, CANCEL and BYE, re-transitions

"This list is for NEW development of the core SIP Protocol.
Use sip-implementors@cs.columbia.edu for questions on current sip.
Use sipping@ietf.org for new developments on the application of sip."

> I would just like to verify my understanding
> regarding 'Backup Routes' with SIP.
>
> Suppose one UA has 2 routes, one primary to
> PROXY1, and one secondary to PROXY2.
>
> Suppose PROXY1 is down.
> When the UA sets up a session, it will
> Send INVITE to Proxy 1
> Timeout
> Eventually send INVITE to Proxy 2, and setup the session.

When advancing to proxy 2, one or more of the
following should change within INVITE to
denote forking:
1) From tag
2) Call-ID
3) Via branch

> Suppose now Proxy 1 and 2 are UP.
> UA send INVITE to Proxy 1
> Proxy sends trying and sets up session.
> Proxy 1 dies.
> UA sends BYE to Proxy 1, though does not receive any response.

The 100 Trying response does not create an early
dialog.  Thus CANCEL must be sent instead of a BYE.

> Shouldn't at this point the UA try to send the
> BYE to the Proxy 2 as well?

The CANCEL should only be sent where the INVITE
was sent.  (This is mainly because the top via entry
is used to help identify the cancelled request.)

If a 101-2xx response was received and the
proxy's Record-Route entry included a hostname
representing both proxies, the BYE could
target advance to proxy 2 when proxy 1 is not
responding.

Please see RFC 3261 sections 9, 12, and 13 for
a more complete description.

_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip
_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip