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Polycom Soundstation Duo Conference Call Issue

syed.mohsin86
Level 1
Level 1

Hi,

 

I have Polycom Soundstation duo integrated with CUCM 8.0(3) as third party SIP device (advance). Problem is when i create Ad hoc conference call from Polycom soundstation only 2 participants join the conference, how can i increase number of participants or is there any limitations on CUCM for Third party SIP Device?

From my normal 6921 IP phone I can add upto 16 participants.

 

Thanks

7 Replies 7

Sreekanth Narayanan
Cisco Employee
Cisco Employee

Hi Syed,

Does the polycom endpoint have a built-in-bridge for conferencing? Does that have any limitations?

The reason you are able to conference 16 ppl for a 6921 phone is because it's using the CUCM software CFB fo this purpose and not its own BIB. You should check if the polycom also invokes the CUCM CFB or if it's trying to do this on its own resources.

 

Thanks

Sreekanth

Thanks for the response, can u guide me how can i check that polycomm is using CFB ? even i have assigned MRGL to it on device configuration page.

 

Thanks.

You'll need to use RTMT to check the usage of the CFB resources on the server where the CFB is located. This can be done using the Performance monitor tab.

Go under the respective server --> Cisco Media Streaming App --> CFBConferencesActive and CFBStreamsActive. Check if this increases as you start the conference on the polycom endpoint.

Another way of checking this is to check the CCM detailed traces.

Ok i have tried this as you suggested and i see CFB graphs going up as i initiate conference from 6921 but as i initiate conference from polycom soundstation there is not activity in graph and it remains at 0 which confirms that Polycom soundstation is not using CFB.

I need to know how can i enforce it to use CFB and not to use its local resource?

 

Thanks for your support.

Can you check what the setting for the "Built In Bridge" field is on the polycom endpoint ? Is it set to Off?

 

I have searched alot for this option but unfortunately couldnt find it...

We should make sure if the behavior we are observing is expected. So, please configure an Xlite or any other 3rd party sip phone and make a conference, and check if the conference bridge is invoked. If not, I think we are hitting a limitation.

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