Re: Retransmission of INVITEs

I don't understand what you don't understand.

The problem is an INVITE gets lost, nobody learns that,
caller gets never a final reply and eventually hangs up
even if there is a callee willing to have a call and
restarted proxy willing to help set up the call.

I think James' T2 proposal is fair enough.

-Jiri

At 06:57 PM 10/3/2002, Medhavi Bhatia wrote:
>James,
>
>I dont understand why the retransmission will be necessary in this case
>after the 100. Since it is a UA, the user is already
>monitoring the call, so he will hangup when no response is received. If its
>a B2BUA/proxy, timer C should take care of this.
>
>-Medhavi.
>
>----- Original Message -----
>From: "James Undery" <jundery <at> ubiquity.net>
>To: "SIP (E-mail)" <sip <at> ietf.org>
>Sent: Thursday, October 03, 2002 7:57 AM
>Subject: [Sip] Retransmission of INVITEs
>
>
>Hi,
>
>Whilst refreshing my knowledge of retransmission I came across a problem I
>thought had been resolved, and yet can't find it in the SIP archives.
>Retransmission of INVITEs cease upon receipt of provisional responses (the
>reliability of final responses is left to the server transaction), however,
>this introduces a problem for 100 responses (not generic 1xxs). The problem
>is a UAC sends an INVITE, a proxy responds with 100 and then promptly dies.
>The UAC has ceased retransmission of the INVITE waiting for a response
>that'll never come. My favoured solution would be for 100s to INVITEs to
>step the retransmission to T2 and have non 100 provisional responses stop
>retransmission. Anyway is this a non problem / stupid solution?
>
>James
>
>P.S. I am far less concerned about proxy failure once the request has
>actually reached the UAS, so ignored that.
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>
>
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--
Jiri Kuthan            http://iptel.org/~jiri/

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