Re: [rtcweb] Forking & Early Media - Was Re: Minimal SDP negotiation mechanism

On 09/20/2011 01:39 PM, Cullen Jennings wrote:
> On Sep 20, 2011, at 2:17 PM, Olle E. Johansson wrote:
>
>    
>> 20 sep 2011 kl. 22:07 skrev Cullen Jennings:
>>
>>      
>>> On Sep 20, 2011, at 7:07 AM, Olle E. Johansson wrote:
>>>
>>>        
>>>> 20 sep 2011 kl. 15:00 skrev Harald Alvestrand:
>>>>
>>>>          
>>>>> Once we start requiring that the PeerConnection know the difference between "early" media and "late" media, it seems to me we're slipping down a slippery slope.
>>>>>            
>>>> The difference between early and late media is purely a billing decision in PSTN. I don't think we should separate these on the rtcweb side. It's a PSTN gateway issue, not something to be bothered with in rtcweb.
>>>>
>>>> "The campaign to simplify call states"
>>>>          
>>> Simplification is good - just make sure that it is possible to gateway to a SIP call to 1-800-go-fedex. A simplification that broke that would, in my personal opinion, would not be acceptable. (and I think that early media is a bit more than just a billing artifact but that's a longer story). You agree?
>>>        
>> Absolutely, but if the call is in UP state on the rtcweb side as soon as the PSTN/SIP gateway receives early media we will have no issue with sending DTMF in early media, which is what you are trying to trick me with :-)
>>      
> really I was not going to mention DTMF :-)
>
> I was thinking a a call where the browser initiates the call, sends something to the web2sip signaling gateway that sends a SIP invite to the PSTN GW. Now the PSTN gateway sends a 180 and starts ICE.

The PSTN GW sounds like a room pressure aware device that may encounter 
said vacuum space on return. That resolves the potential error on GLARE 
exceptions.

Instead of "stable", the mobile units wants ubiquity with real-time 
GLARE-to-FIXATED states, smoothly.

Or; in return, the display unit and mobile unit become actors in 
use-cases (and used-cases), so we clarified (the two) for rtp-usage.

JIT repository is FIXATED when there is no HTTP sent/delivered due to 
non-expired certificates. Browsers that have no encryption provided by 
the certificate is not obligated to store such bytecode in non-obvious 
encryption; thus, raises the bar a little on reason to federate to 
FIXATED-known headers; no X-, no 0day.

JIT repository that stay FIXATED for more than one day are then 
valuable. The sites' bytecodes simply tune-out the origin servers that 
deal only with unified user-agents, as the FIXATED state cares more 
about the agent than the user (note-well vs. E911 recommendations).

SIP presents RTCWeb with many ways to undo, untwist, or simply focus on 
the user more, yet professionally there is no FIXATED state, or we have 
no professional reason why such FIXATED state of JIT does not exist and 
displays continue to consume power while on-the-... federated signal.

Elevators that just want the SSRC for MMUSIC, there are professionally 
known geolocations for that range and length of signal. We think of 
treats, not tricks.

>   Then the PSTN GW starts sending RTP with the fedex prompts in them and then it eventually sends a 200.  The key thing is the browser needed to be able to receive RTP before the signaling gateway received the 200. As long as we are on the same page there, it's good with me. The PSTN GW sends the RTP directly to the the browser but sends the SIP to the signaling gateway.
>
>
>    
>> Now, if we force early media which is supposed to be one-way, not many implementations would support fedex. I don't see this as a problem if we just forget the early-media and late-media difference and just talk about media and if it's up or not up. That would handle ringback as well.
>>      
> I could care less about if the media is 1 way or two way. I just care about hearing the prompts when I call 1-800-fedex. (I care about early media, not one way media :-)
>
>    
>> In my head, rtcweb is only the handset of a phone. The phone knows about early media, but the handset just has a mike and a speaker and it's simply on or off.
>>      
> sure - not problem with that
>
>    
>>
>>      
>>> That said, I think that doing both forking and early media is hard. Lets assume we are using a signaling gateway that is not a media gateway to translate between a SIP call on one side and whatever is happening over on the browser side. The basic issue is the browser initiating the communications needs to be able to start receiving multiple RTP streams before it even has signaling information to tell it how many it might receive. To simplify this problem, Cary and my draft proposes not allowing forking on the SIP side of the signaling gateway but still allowing early media. If you wanted to do do forking in this case, one would need a SBC that processed media and turned the forked medial legs into one media leg.
>>>        
>> SBS or b2bua - in this case rtcweb to SIP ua. Yep, that's my way of thinking.
>>      
>>> What I'm trying to say is I agree that early media + forking is hard, and the way I would like to simplify that is not allow forking. Note that none of this has much to do with if the PeerConnection object knows if media is early or late - what it needs to deal with is receiving RTP regardless os if the SIP signaling GW had sent it any signaling that indicated what had been in the SIP answer of the SIP side of a signaling GW.
>>>        
>> Agree.
>>
>> /O
>>      
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