In Asterisk 220.127.116.11, I’m using
preprocess => true
pp_vad => true
in codecs.conf to stop the outgoing rtp media stream when the caller is silent. This works great for standard calls but I need to extend the feature to MeetMe() conference calls. I had thought (hoped) that if all calling parties in the conference were silent, then no rtp would be transmitted to any of the callers. However this does not appear to be the case. Even if all phones are muted, there is still an rtp stream transmitted to every party in the conference.
Can anybody confirm if it is possible/impossible to use pp_vad to stop the rtp stream being transmitted in a MeetMe() conference? Or is there another way I can achieve the same thing? I need to have the RTP stream transmitted out from Asterisk only when there is voice activity on one or more calling parties in the conference.
Thanks in advance if anyone is able to help.