Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09

"Parthasarathi R" <partha@parthasarathi.co.in> Tue, 18 June 2013 16:52 UTC

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From: Parthasarathi R <partha@parthasarathi.co.in>
To: 'Iñaki Baz Castillo' <ibc@aliax.net>
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Date: Tue, 18 Jun 2013 22:22:32 +0530
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Cc: "'SIPCORE (Session Initiation Protocol Core) WG'" <sipcore@ietf.org>
Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-09
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Hi Inaki,

I agree with you that RFC 4168 (SIP SCTP) does not explain the contact header usage. But it does not meant that it is the best practice. My comments is based on the later RFC namely RFC 5630 (The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)) and look into the draft for the explanation and Normative statements around contact header w.r.t SIPS URI. 

The current text in the draft is clear to explain the usage of contact header. It will confuse in case anybody implement by just looking into this draft. I think that it is required for SIP over WebSocket to explain the usage of contact header field with example and normative statement. 

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Iñaki Baz Castillo
> Sent: Tuesday, June 18, 2013 12:21 AM
> To: Parthasarathi R
> Cc: SIPCORE (Session Initiation Protocol Core) WG
> Subject: Re: [sipcore] Open issues in draft-ietf-sipcore-sip-websocket-
> 09
> 
> 2013/6/16 Parthasarathi R <partha@parthasarathi.co.in>:
> > I could not understand how this update resolve the issue in the
> following scenario:
> >
> >  SIP UA1 (WS only)----Proxy-----SIP UA2 (UDP/TCP)
> >
> > Here, the dialog is established with contact details for SIP UA1 &
> SIP UA2 as follows:
> >
> > 1) contact: sip:sipua1@example.com;transport=ws
> > 2) contact: sip:sipua2@example.com;transport=tcp
> >
> > In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, how does it
> possible now as per the current update in Sec 5.2.4 of draft-ietf-
> sipcore-sip-websocket-09.
> 
> 
> Hi Parthasarathi,
> 
>   SIP UA1 (SCTP/UDP/TCP)----Proxy-----SIP UA2 (UDP/TCP)
> 
> Here, the dialog is established with contact details for SIP UA1 & SIP
> UA2 as follows:
> 
> 1) contact: sip:sipua1@example.com;transport=sctp
> 2) contact: sip:sipua2@example.com;transport=tcp
> 
> In case SIP UA2 wishes to send RE-INVITE towards SIP UA1, please let
> me know in which section of RFC 4168 (SIP STCP) it is stated that the
> proxy must use Record-Route for the initial INVITE for this scenario
> to work.
> 
> Regards.
> 
> 
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>
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