[Freeswitch-users] WebRTC calls one way with custom sip messages
davidswalkabout at gmail.com
Sat Nov 27 01:40:35 UTC 2021
If you use Verto, you will need to embed the userID and password of a
Freeswitch user in the page resources so it can be passed in the
$.verto.init(...) call. You can protect this signaling channel somewhat by
rotating the password frequently.
However, I don't know if there is any way to protect the large range of
ports that FS needs to be open to handle exchange of audio and video. I
asked here a few weeks ago if it would be possible to configure FS to
ignore requests on these ports from all addresses except those that have an
active login on the signaling channel.
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