Re: [rtcweb] Another consideration about signaling

[Dzonatas Sol <[e-mail]>]

On 09/16/2011 01:56 PM, Henry Sinnreich wrote:
> +1
>
> These are indeed critical items to consider
>
>    
>> if the signaling is implemented within HTTP or WebSocket (by using
>> any custom mechanism), it would be easy for the web server to know the active
>> sessions status in detail, and it could use such a information for rendering
>> it in the webpage (so others web visitors can see the status of my calls, for
>> example).
>>      
>    
>> I just see advantages in *non* mandating a separate and specific
>> signaling protocol within rtcweb, even more taking into account
>> that this is supposed to be an added value for the web. This is: a
>> web-browser MUST NOT be a native SIP phone (IMHO).
>>      
> Though I would add the "other web visitors" may well be other app components
> that creative developers may assemble into rich apps.
>
> Thanks, Henry
>    

Let JSON be the default for websockets, and we can call those lambda 
expressions.



>
> On 9/16/11 10:42 AM, "I�aki Baz Castillo"<[e-mail]>  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?
>
> In the
>    
>> other side, if the signaling is implemented within HTTP or
>>      
> WebSocket (by using
>    
>> any custom mechanism), it would be easy for the
>>      
> web server to know the active
>    
>> sessions status in detail, and it could
>>      
> use such a information for rendering
>    
>> it in the webpage (so others web
>>      
> visitors can see the status of my calls, for
>    
>> example).
>>      
> I just see advantages in *non* mandating a separate and
>    
>> specific
>>      
> signaling protocol within rtcweb, even more taking into account
>    
>> that
>>      
> this is supposed to be an added value for the web. This is: a
> web-browser
>    
>> MUST NOT be a native SIP phone (IMHO). I wouldn't like to
>>      
> see a competition
>    
>> between Firefox 12 and Chrome 17 in next SIPit
>>      
> (Session Initiation Protocol
>    
>> Interoperability Test).
>>      
> Best regards.
>
>    


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