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PVDM upgrade and conference resources allocation

Kalliopi Vazima
Beginner
Beginner

Hello everyone,

 

i got stuck in a case where in an ISR4351 i have configured a dspfarm profile for conference like below

 

dspfarm profile 1 conference 

 codec g729br8

codec g729r8

codec g729abr8

codec g729ar8

codec g711alaw

codec g711ulaw

maximum conference-participants 16

maximum sessions 2

associate application SCCP

 

the PVDM resources are used for both ad hoc conference and for conference now option too.

I have been asked for an upgrade of the resources in order to cover two cases

1) add one more session with 32 maximum participant in the current infrastructure

2) add one more session with 64 maximum participants for a new meet me option

 

The question is if for the first requirement the only option is to configure a second dspfarm profile like

dspfarm profile 2 conference 

codec...

maximum conference-participants 32

maximum sessions 1

associate application SCCP

 

and add the additional cfb to the MRGL already used by existing users?

What about the second requirement? I was thinking about configuring one more dspfarm profile like

dspfarm profile 3 conference 

codec...

maximum conference-participants 64

maximum sessions 1

associate application SCCP

 

but i m not really sure how this cfb resource will be allocated from the call manager to users of the new meet me conference.

 

Any thoughts?

 

Thanks in advance

 

Kalliopi

 

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3 Replies 3

Jonathan Schulenberg
Hall of Fame
Hall of Fame

That is not going to work. CUCM will allocate the bridge when a call initially needs the media resource (i.e. the third participant in an ad-hoc conference or the first person joins for rendezvous Conference Now or MeetMe). It has no idea how many people will eventually join and is not able to change resource bridges mid-call. You need to choose a consistent maximum participant limit.

Also, conferences that large are going to be unwieldy since Conference Now and MeetMe lack active speaker indications and the ability to mute others. You may want to look at Webex, CMS, or even an FOSS option such as Jitsi which, I believe, has a SIP trunk option for dial-in.

Thank you Jonathan for you prompt reply. Is there any document that refers to your reply?

I was thinking that if i create a difference profile, different resources will be available for the call manager in order to be allocated through MRGL to suitable users. i understand this is not going to happen, correct?

 

Regards

 

Kalliopi

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