RE: [Sipping] Draft on SIP Interface to VoiceXML
Darshan Bildikar <darshanb@huawei.com> Wed, 14 December 2005 05:04 UTC
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Date: Wed, 14 Dec 2005 10:33:33 +0530
From: Darshan Bildikar <darshanb@huawei.com>
Subject: RE: [Sipping] Draft on SIP Interface to VoiceXML
In-reply-to: <156d01c5ffcf$00236540$0a01a8c0@db01.voxpilot.com>
To: 'Dave Burke' <david.burke@voxpilot.com>, sipping@ietf.org
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Dear Dave, Some comments "On receipt of the INVITE, the VoiceXML Media Server issues a provisional response, 100 Trying, and commences the fetch of the initial VoiceXML document. The 200 OK response indicates that the VoiceXML document has been fetched and parsed correctly and is ready for execution" 1) Many SIP flows rely upon media server answering the INVITE immediately. In my opinion it would be better to begin parsing the VXML after generating the answer to the INVITE request by the Application Server. If there is a problem with the VXML then it the SIP dialog can be ended with a BYE. 2) Instead of using "400 Bad Request" wouldn't it be better to use "488 Not Acceptable Here"? 3) The offer answer exchange in section 3.2 is not as per the offer answer standard. The initial INVITE is without an offer and thus the initial offer is contained in a reliable provisional response. In this case the answer should be in PRACK. Also, Note that RTP will start immediately after the reliable provisional response 4) I don't think it should be mandatory to support a minimal set of codecs. Interoperability is ensured by the offer answer standard. If the media server doesn't support a particular codec; so be it. The request is rejected based on SDP. 5) In the REFER flow it doesn't make sense to retrieve and process the XML until the referee(?) has answered the INVITE, it would be a waste of n/w and computing resources to fetch and process the VXML if it wont be used at all. Regards, Darshan -----Original Message----- From: sipping-bounces@ietf.org [mailto:sipping-bounces@ietf.org] On Behalf Of Dave Burke Sent: Tuesday, December 13, 2005 3:52 PM To: sipping@ietf.org Subject: [Sipping] Draft on SIP Interface to VoiceXML Hello, I would like to notify folks of a recent I-D: http://www.ietf.org/internet-drafts/draft-burke-vxml-00.txt The specification is principally concerned with the binding of SIP and VoiceXML behaviours and represents a standardisation of similar but somewhat non-interoperable interfaces in production in many of today's SIP networks. Thanks, Dave
_______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP
- [Sipping] Draft on SIP Interface to VoiceXML Dave Burke
- RE: [Sipping] Draft on SIP Interface to VoiceXML Francois Audet
- RE: [Sipping] Draft on SIP Interface to VoiceXML Darshan Bildikar
- Re: [Sipping] Draft on SIP Interface to VoiceXML RJ Auburn
- Re: [Sipping] Draft on SIP Interface to VoiceXML Dave Burke