On 5 July 2017 at 18:08, Mark Melling <mark.melling@savageminds.com> wrote:
Thanks Giovanni for the suggestion.
I tried some more experiments and basically if I call a verto client and add them to a conference then they don't hear the audio (although the conference is detecting audio when they speak).
But if they dial into the conference then everything appears fine and they do hear audio.
Specifically from fs_cli I entered:
originate <call-url> &conference(<conf-name>@default)
If call-url is a sip client then you hear the conference music, but if call-url is a verto client you don't hear any conference music. But the conference does detect when the verto client is speaking (at least the status in the verto web page indicates the user is talking).
Whereas if you dialled into a conference from a verto client then you would hear the conference music.
So I'm not sure how I can work around this.
Have you tried what I suggested?
Maybe this is because verto (webrtc) takes time to establish audio because of stun, etc etc
Try this: instead of generating autocall from inside conference (eg instead of using autocall),originate call to user, wait for her to answer, then (after she answer) sleep for 2 seconds, then transfer her to the conf
-giovanni