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

Silvia Pfeiffer <silviapfeiffer1@gmail.com> Thu, 08 September 2011 01:14 UTC

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From: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
Date: Thu, 8 Sep 2011 11:15:46 +1000
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To: Matthew Kaufman <matthew.kaufman@skype.net>
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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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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.

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
>
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