Re: [rtcweb] Another consideration about signaling

Kevin,

2011/10/7 Kevin P. Fleming <kpfleming@digium.com>:
> On 09/16/2011 10:42 AM, Iñaki Baz Castillo wrote:
>>
>> Hi all,
>>
>> Let's imagine that rtcweb defines a specific signaling protocol (i.e.
>> SIP) so browsers MUST use it natively for signaling the media streams.
>> Of course this would require a SIP proxy/server in server side (think
>> about NAT) which IMHO seems a showstopper (how to deploy such a SIP
>> proxy in shared web hostings? a "mod_sip" for Apache? should I make a
>> XMPP<->SIP protocol gateway in order to intercommunicate web-browsers
>> with pure XMPP clients?)
>>
>> But there is another important drawback with this assumption:
>>
>> A web site could be interested in drawing in the web the status of the
>> different audio/video streams between users connected to the web. This
>> could mean displaying in the web the active streams (audio/video),
>> when a session is on hold, when it's resumed again, when a new stream
>> is added to a multimedia session (i.e. offering video within an
>> already established audio session).
>>
>> If the signaling uses a separate channel (i.e. SIP) then there is no
>> way for the web server to know what happens during multimedia sessions
>> (or it would be really difficult to achieve). So multimedia sessions
>> would be completely separated from the web page itself. Is that what
>> we want?
>
> (resurrecting this old thread)
>
> Regardless of the signaling protocol in use, it seems unlikely to me that a
> web application developer is going to want to be *involved* in the call
> signaling itself; they are going to pass that responsibility over to some
> existing code/library/server/system (which could mean just shuttling SIP
> messages back and forth between WebSocket connections and UDP connections).
>
> If this assumption is true, then the web application developer who wants to
> achieve the result you describe above is going to choose a SIP
> library/server that allows them access to the call states and other pieces
> of information that they need, in preference to those who don't make that
> information available. I can't imagine that a practical solution would be to
> choose a SIP library/server that handles the calls in an 'opaque' fashion
> but then build a web application that is required to snoop on all the
> signaling to be able to monitor activity.
>

Web developers have to use well defined APIs so they don't need to
know how the underlaying protocol works. Instead, they rely on the API
methods and information to achieve what Iñaki is stating.

> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>