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

Justin Uberti <juberti@google.com> Thu, 08 September 2011 19:28 UTC

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From: Justin Uberti <juberti@google.com>
Date: Thu, 08 Sep 2011 15:29:52 -0400
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To: igor.faynberg@alcatel-lucent.com
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Subject: Re: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remote recording - RTC-Web client acting as SIPREC session recording client]
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On Thu, Sep 8, 2011 at 3:22 PM, Igor Faynberg <
igor.faynberg@alcatel-lucent.com> wrote:

> **
>
> On 09/08/11 16:54, Igor Faynberg wrote:
> ...
>
>
> If the issue is getting a signal from a Web server to a client, there's
> approximately 100 ways to get notifications from the server to the client
> using HTTP (hanging GET being one of them).
>
>
> I thought  that COMET-like polling is inefficient. Hanging GETs
> require server resources to hold a TCP session o open. Firewalls and IE7
> time out  a GET after 30-60 seconds.
>

In that case, you start a new hanging GET. If Gmail can work using this
approach, telephony applications should be able to use it too.

>
>
> Now that WS is getting standardized, there will be 101.
>
>
>  101st, seems to be a solution, I agree.  But it has not finished
> standardization, while SIP has.
>

I would argue that WebSocket is considerably more mature than WebRTC.

>
> Igor
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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