[cisco-voip] Part 2: Conferencing Codecs Across WAN

[Pavan]

Rhodium,

Let me try to clarify a bit

Assume phone b1 is in a conference with b2 and a1,

If a1 adds another party a2 to the conference, (conference extension)  
a new rtp stream would flow from a2 to the cfb in site b

If a1 places the existing call on hold and creates a new conference  
with a2 and a3, now we have a conference resource in site a which has  
rtp streams from a1,a2& a3. Assume that a1 now joins both these  
conferences together with the active call being the second conference  
call( conference chaining), we will end up with conference1 containing  
b1,b2 and cfb in site a. Conference2 contains a1,a2,a3 and cfb from  
site b.

In essence both the conference bridges are chained together. Things  
start getting more complicated when you start chaining 3 or more  
conferences. Take a look at nonlinear adhoc conference chaining  
service parameter in ccm.

Let me know if you still have a question.

-pavan

Sent from my phone

On Jan 19, 2010, at 6:26 PM, Rhodium <rhodium_uk at yahoo.co.uk> wrote:

> Thank you for replying Craig.
>
> Let me put it another way, what happens if a phone in site B (for  
> argument's sake Phone B1) already in a conference with phones in  
> site A as per your reply starts another second conference with  
> another phone in site B, therefore invoking the MRGL for itself.
>
> Does that mean, the phone in site B that invoked the second  
> conference will have two distinct RTP streams, one to site A for its  
> resources, and one locally for the second conference it initiated.  
> So the second conference would  then get mixed again at the primary  
> conference's PVDMs or vice versa?
>
> I don't understand how the RTP streams would get mixed together or  
> how they would flow?
>
> I am getting a headache just thinking about the RTP flow and how  
> they would all mix.
>
> Jason
>
>
>
>
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