Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as SIPREC session recordingclient]

"Olle E. Johansson" <oej@edvina.net> Thu, 08 September 2011 05:49 UTC

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From: "Olle E. Johansson" <oej@edvina.net>
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Date: Thu, 8 Sep 2011 07:51:08 +0200
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To: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as SIPREC session recordingclient]
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8 sep 2011 kl. 03:15 skrev Silvia Pfeiffer:

> If implementations count for anything, check out:
> 
> http://phono.com/
> and
> https://docs.google.com/present/view?id=0AcNAXuadeXIxZGY4NDl4Z2RfNTlod3BrOXNjZA&hl=en&pli=1
> 
> They use SIP with web sockets.

Nice. That will make it easy to hook in the rtcweb/webrtc media layer.

Now, I don't see clearly if you want to argue for or against SIP as part of rtcweb standards with these examples, but that may be my biased eyes reading... ;-)

/O
> 
> Cheers,
> Silvia.
> 
> 
> On Thu, Sep 8, 2011 at 8:30 AM, Matthew Kaufman
> <matthew.kaufman@skype.net> wrote:
>> On 9/7/11 12:20 PM, Randell Jesup wrote:
>>> 
>>> I also started from the same point - assume SIP.  SIP gives you all the
>>> things that the zillions of hours and emails to define it and define
>>> extensions and secure it provides, without having to reinvent all those
>>> wheels (or ask app developers to reinvent them).  Why go through the
>>> horrible pain of choosing something else, or why throw the app developers to
>>> the wolves to fend for themselves?
>>> 
>>> However...
>>> 
>>> Two things have swayed me.  The primary one is the suggestion of
>>> Offer/Answer in the browser.  This breaks out the important negotiation
>>> piece that almost any application would need, and while not perfect, SDP O/A
>>> is a zillion times simpler than SIP with all the extensions one could use.
>> 
>> I agree with this. While I am also opposed to SDP O/A, these are two
>> unrelated arguments to have... and not baking a SIP phone into the browser
>> is *more* important than avoiding a repeat of the offer/answer problems.
>> 
>>> 
>>> The other thing that swayed me was thinking about federation and the apps
>>> that will be built with this.  A webrtc app talks to its (web)server, other
>>> webrtc clients running the app that talk to the server, and to other webrtc
>>> applications/networks that federate with it (and their clients).
>>> 
>>> Federation is in the same hands as the person who provides/wrote the app.
>>>  If they have no interest in federation you can't force it, and they may
>>> have no use for all the fancy SIP standards.
>> 
>> And for numerous types of apps (think: server-based augmented reality
>> systems), "federation" doesn't even make sense.
>> 
>>> 
>>> On the other hand, if they *want* to either provide access to the wider
>>> communication net that is the PSTN network, now or in the future, or they
>>> want easy federation with other networks, it behooves them to use SIP or
>>> something very close to it or equivalent/convertible (at a basic level at
>>> least) to it.
>>> 
>>> So what conclusions do I draw from this?
>>> 
>>> 1) O/A via SDP in the browser simplifies a lot of things (including
>>> handling new codecs, etc).  It doesn't extremely limit an application,
>>> though we should think about how an application can interact with the
>>> fmtp/etc parameters used.
>> 
>> I agree that it would simplify some interop cases, but at an unfortunate
>> cost in lack of flexibility and functionality. Still not nearly as bad as if
>> we put a full SIP stack in there though.
>> 
>>> 
>>> 2) SIP as a *separate* item that can be cleanly and easily *added* to a
>>> webrtc app to handle the call setup/etc is a good idea.
>> 
>> I would be open to looking at this again, *after* RTC is already in browsers
>> and successful, to see if it actually solves a real use case.
>> 
>> Matthew Kaufman
>> 
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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* Olle E Johansson - oej@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden