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Re: [rtcweb] draft-ibc-rtcweb-sip-websocket -- WebSocket Transport for Session Initiation Protocol (SIP)

"Dan Wing" <dwing@cisco.com> Fri, 16 September 2011 18:01 UTC

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From: Dan Wing <dwing@cisco.com>
To: 'Randell Jesup' <randell-ietf@jesup.org>, rtcweb@ietf.org
References: <CABw3bnO+85i-TtuqS+P4n+rYgyxyoASc8HXpADhy4QPTC0_szA@mail.gmail.com> <6F469757-6B5C-4DC9-BC34-026F34C7E508@phonefromhere.com> <CAD5OKxvCSJWS+F72P_WOFapmtffkLCSSe3A-rDEUOhNjWcoh4A@mail.gmail.com> <E44893DD4E290745BB608EB23FDDB7620B39D8@008-AM1MPN1-043.mgdnok.nokia.com> <CAD5OKxtt9phWr9Vrn5J1STxA9airR8g-tHMnddFP=QwrTHuPcA@mail.gmail.com> <4E724E25.1000204@jesup.org>
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Date: Fri, 16 Sep 2011 11:04:02 -0700
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Subject: Re: [rtcweb] draft-ibc-rtcweb-sip-websocket -- WebSocket Transport for Session Initiation Protocol (SIP)
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Randell Jesup
> Sent: Thursday, September 15, 2011 12:13 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] draft-ibc-rtcweb-sip-websocket -- WebSocket
> Transport for Session Initiation Protocol (SIP)
> 
> On 9/15/2011 11:57 AM, Roman Shpount wrote:
> > Actually SIP over UDP is what is typically used for mobile apps now.
> > If you are doing SIP from the public IP, you do not need to maintain
> > the connection even if TCP/TLS is used.
> 
> Markus had a point: SIP over UDP requires keepalives if it's behind a
> NAT as well.  So maybe it's not such a big difference, depending on the
> keepalive rates needed (and for UDP circa 30 sec is typical).

PCP could help with that, 
http://tools.ietf.org/html/draft-ietf-pcp-base-13#section-7.3

-d

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