SIP CONFERENCE - audio mixing

Unanswered Question
Apr 9th, 2008

Hello,

I had a weird question about the SIP adhoc conference with the ccm 5. Here is the scenario:

- I start a SIP adhoc conference with a 7961GGE. I call many types of phone (7940, 7970, gsm). I can't add more than 6 participants.

I thought that this limitation was due to the fact that the initiator makes the audio mixing. But in fact I can see on wireshark that all the RTP stream go to the PBX.

Moreover, if I unplug the 7961, the other participants are still on the conference.

It means that the PBX is doing the audio mixing right ? How to be sure about that ?

Therefore, why is there a limitation on the phone ?

Thanks,

David

I have this problem too.
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