So it is useful if one of UAS or UAC requires it, but it does not have to be mandatory. Some comments: -- RFC3261 mentions early dialog without mentioning RFC3262. Then it seems logical to me that it needs to be made clear in this RFC3261 that early dialog must mean Contact and Record-Route (if Record-Route was received in INVITE) headers is mandatory in 1xx without reference to 100rel.
-- A UAS could always send 1xx with headers that are required for early dialog but it doesn't have to enforce 100rel (eg because the origination or UAS side itself may not support reliable provisional response handling, or reliable provisioning not really required for its operation). UAS could send "support:100rel" if it supports it, or it would not send it if it doesn't support this. In my opinion, if UAC hasn't sent 100rel required, it should be up to the UAS to decide whether to enforce 100rel (with "required:100rel") if its application really requires SIP requests before call answer. If the origination side (UAC) side has a need to send early requests, like UPDATE, then the UAC should require 100rel from the termination side (UAS) by sending this in INVITE. In a VoIP service provider world, these kind of capabilities are configured during interconnect turn up. -- I notice that some vendors gateway implementations, even if gateway is the termination side, require 100rel for the gateway to receive pre-answer requests such as UPDATE. This really didn't have to be this way. I have always seen these gateways, when it is the termination side, respond back SIP 183 with the headers that create early dialog. So if the origination side received the SIP 183 response, then there is no reason for the origination side to now not be able to send UPDATE request. Also, no reason for the termination gateways to not accept the SIP UPDATE without requiring PRACK. Thanks. On Wed, Aug 3, 2011 at 11:46 AM, Robert Sparks <rjspa...@nostrum.com> wrote: > (removing the rfc-editor and trimming the distribution to the lists) > > On Aug 2, 2011, at 5:24 PM, Iñaki Baz Castillo wrote: > > > 2011/8/2 Robert Sparks <rjspa...@nostrum.com>: > >> Further, they're only going to make sense for 1xx that is sent using > 100rel. > > > > This has been discussed in sip-implementors, and that assertion seems > > incorrect. As I've reported in the errata: > > > > > > Section 12.1: "Dialogs are created through the generation of > > non-failure responses to requests with specific methods. Within this > > specification, only 2xx and 101-199 responses with a To tag, where the > > request was INVITE, will establish a dialog." > > > > Section 12.1.1: "When a UAS responds to a request with a response that > > establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy all > > Record-Route header field values from the request into the response > > [...]. The UAS MUST add a Contact header field to the response." > > > > So it's clear that a 1xx response to an INVITE creates a dialog and > > then it MUST contain a Contact header and mirrored Record-Route > > headers, *regardless* the usage of 100rel. > > > > Am I wrong? if so, why? > > Not wrong, just incomplete. This will create an (early) dialog at the UAS. > It may or may not create a dialog at the UAC without 100rel since the > message may never get to the UAC. Where I said "make sense" above, > it might have been better if I had said "be useful". > > > > > Regards. > > > > > > -- > > Iñaki Baz Castillo > > <i...@aliax.net> > > _______________________________________________ > > sipcore mailing list > > sipc...@ietf.org > > https://www.ietf.org/mailman/listinfo/sipcore > > _______________________________________________ > Sip mailing list https://www.ietf.org/mailman/listinfo/sip > This list is essentially closed and only used for finishing old business. > Use sip-implement...@cs.columbia.edu for questions on how to develop a SIP > implementation. > Use dispa...@ietf.org for new developments on the application of sip. > Use sipc...@ietf.org for issues related to maintenance of the core SIP > specifications. >
_______________________________________________ Sip mailing list https://www.ietf.org/mailman/listinfo/sip This list is essentially closed and only used for finishing old business. Use sip-implement...@cs.columbia.edu for questions on how to develop a SIP implementation. Use dispa...@ietf.org for new developments on the application of sip. Use sipc...@ietf.org for issues related to maintenance of the core SIP specifications.
