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

"Parthasarathi R" <partha@parthasarathi.co.in> Mon, 17 June 2013 17:40 UTC

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From: Parthasarathi R <partha@parthasarathi.co.in>
To: 'Iñaki Baz Castillo' <ibc@aliax.net>
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Date: Mon, 17 Jun 2013 23:10:32 +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,

As you mentioned, "record-route" is the right solution and not documented in the draft as of now. My comment is to explicitly add the statement for clarity.

Thanks
Partha

> -----Original Message-----
> From: sipcore-bounces@ietf.org [mailto:sipcore-bounces@ietf.org] On
> Behalf Of Iñaki Baz Castillo
> Sent: Monday, June 17, 2013 2:45 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>:
> > Hi Inaki,
> >
> > Open Item 2 comment: The current statement states that
> >
> >      " All SIP elements MUST implement at least one of the following:
> >
> >       *  Both UDP and TCP transports.
> >
> >       *  SIP WebSocket transport."
> >
> > 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, this subject was heavily discussed during the past
> year. It's clear that in 99% or 100% of cases, a SIP WebSocket Client
> (i.e. a browser) will not be able to receive incoming WS connections
> (it is not a SIP WebSocket Server) and thus, the proxy in your flow
> has to perform record-routing, a *standard* mechanism defined in RFC
> 3261 for making your scenario feasible.
> 
> IMHO it is not so hard to assume that when SIP over WebSocket devices
> are present, the proxy they connect to must perform record-routing.
> The same is true for clients behind NAT.
> 
> 
> Regards.
> 
> 
> --
> Iñaki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> sipcore mailing list
> sipcore@ietf.org
> https://www.ietf.org/mailman/listinfo/sipcore

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