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

"Ravindran Parthasarathi" <> Thu, 08 September 2011 17:10 UTC

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Date: Thu, 8 Sep 2011 22:41:17 +0530
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Thread-Topic: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remoterecording - RTC-Web client acting as SIPREC session recordingclient]
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From: "Ravindran Parthasarathi" <>
To: "Randell Jesup" <>, <>
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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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In principle, I like your idea.


>-----Original Message-----
>From: [] On
>Of Randell Jesup
>Sent: Thursday, September 08, 2011 12:50 AM
>Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE:
>Remoterecording - RTC-Web client acting as SIPREC session
>On 9/7/2011 3:07 AM, Olle E. Johansson wrote:
>> 6 sep 2011 kl. 21:24 skrev Asveren, Tolga:
>>> What about semantics (adding just a X-header won't help there) or
>much SIP would be left anyhow if all semantical control is exposed
>through the API?
>>> I think bridged line appearance is a good test to run against
>different models.
>> Well, I tried to stay neutral but examples likes this makes me not
>want SIP in the browser. DTMF, Early Media, bridge line apperances and
>other PSTN legacy will make the implementation too focused on classical
>telephony and we'll spend too much time implementing features that are
>application specific and we can implement in controlling applications,
>client or server-side.
>> Cullen tried to make a draft with "limited" SIP (maybe "SIP Lite")
>it will be hard to protect that from the myriad of extensions that add
>PSTN functionality that's not really needed to set up multimedia calls
>between two browser users. It may be needed for gatewaying to legacy
>systems, but if we don't "stop Olle in the gate" - verbatim translation
>of a Swedish saying that propably doesn't mean much to most people on
>the list - I think we will never be done.
>> Of course, being a SIP developer, I started off with thinking that
>in the browser was the default route. I am beginning to understand that
>the browser is the user interface part we all need, the media handler.
>We all have different requirements on how to control that media GUI and
>to get anywhere I am beginning to think the logic for signalling to set
>up rendevouz points and manage sessions has to move "somewhere else"
>where we can implement SIP, XMPP or some other protocol that fulfills
>the need of our respective application.
>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?
>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,
>O/A is a zillion times simpler than SIP with all the extensions one
>could use.
>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.
>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.
>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.
>This means a webrtc app could use something else, or roll its own.
>would use SIP.
>This would require (limited) SIP to be available as part of webrtc in
>the browser - but as an option, not as a mandate.  An application could
>use an extended SIP client via JS or other mechanism.  Basic SIP would
>need to be in all webrtc implementations so the apps could rely on
>having the option.
>This would make building apps & services that can (optionally) call
>phones easy (very easy perhaps), while not limiting the ability of app
>developers to innovate.  It also makes it easier to build servers to
>handle webrtc apps.
>Randell Jesup
>rtcweb mailing list