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Re: [Sipping] Liaison Statement on offer/answer procedures
Hi,
When thinking about which alternative is the best, I think it's also
good to keep the following in mind:
It is not only the SDP that may be "updated" by the "intermediate"
transactions. For example, I guess the remote target could also be
updated (see example below). So, what happens to that if the re-INVITE
then fails?
So, whatever alternative we go for, I think that we should apply the
same rule to whatever data (SDP, remote target etc) has been updated
between the re-INVITE request and error response. Otherwise we will
really create a mess with confusion and interop problems.
Does anyone have a problem with taking the-same-rule-applies-to-all-data
approach?
Example with updated remote target:
re-INVITE ------------------->
UPDATE with new rem.target--->
<- 200 (UPDATE) --------------
<- 4xx (re-INVITE) -----------
What happens to the new remote target? Is it kept, or is there a
fallback to whatever remote target was used before the UPDATE was sent?
Regards,
Christer
-----Original Message-----
From: sipping-bounces at ietf.org [mailto:sipping-bounces at ietf.org] On
Behalf Of Christer Holmberg
Sent: 9. toukokuuta 2008 10:19
To: Paul Kyzivat; Gonzalo Camarillo
Cc: mmusic-chairs at tools.ietf.org; sipping; Mary Barnes
Subject: Re: [Sipping] Liaison Statement on offer/answer procedures
Hi,
I agree with Paul. We need to choose *some* solution. So, I think it
would be good that anyone who has a preference indicates that.
Regards,
Christer
-----Original Message-----
From: sipping-bounces at ietf.org [mailto:sipping-bounces at ietf.org] On
Behalf Of Paul Kyzivat
Sent: 8. toukokuuta 2008 15:51
To: Gonzalo Camarillo
Cc: mmusic-chairs at tools.ietf.org; sipping; Mary Barnes
Subject: Re: [Sipping] Liaison Statement on offer/answer procedures
I agree that sipping is the right place for the work, and that it ought
to be done, eventually. It hasn't seemed like especially pressing, but
apparently 3gpp thinks otherwise.
I agree that, at least on the surface, it is less important which
solution is chosen than it is that *some* solution be chosen. But there
are pros and cons to the two obvious approaches, so it is best to do
more than just flip a coin.
Thanks,
Paul
Gonzalo Camarillo wrote:
> Folks,
>
> the SIPPING WG has received the following LS from 3GPP:
>
> https://datatracker.ietf.org/liaison/444/
>
> We have talked to the MMUSIC chairs (MMUSIC was in the cc: of this LS)
> and they are OK with working on this LS in SIPPING. In any case, we
> will send our answer to the MMUSIC folks for comments before getting
> back to 3GPP.
>
> This LS has to do with the offer/answer method, which was developed by
> MMUSIC, but is also related to the SIP machinery associated to
> offer/answer (i.e., rejected re-INVITEs). That is why we consider
> SIPPING the right place to discuss it.
>
> Comments on this LS are appreciated.
>
> Thanks,
>
> Gonzalo
> SIPPING co-chair
> _______________________________________________
> Sipping mailing list https://www.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP Use
> sip-implementors at cs.columbia.edu for questions on current sip Use
> sip at ietf.org for new developments of core SIP
>
_______________________________________________
Sipping mailing list https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP Use
sip-implementors at cs.columbia.edu for questions on current sip Use
sip at ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP Use
sip-implementors at cs.columbia.edu for questions on current sip Use
sip at ietf.org for new developments of core SIP
_______________________________________________
Sipping mailing list https://www.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors at cs.columbia.edu for questions on current sip
Use sip at ietf.org for new developments of core SIP