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InnovateAsterisk / Browser-Phone

A fully featured browser based WebRTC SIP phone for Asterisk
https://www.innovateasterisk.com
GNU Affero General Public License v3.0
499 stars 245 forks source link

direct media #502

Open prathibhacdac opened 7 months ago

prathibhacdac commented 7 months ago

Why is direct_media set to no in pjsip.conf?

InnovateAsterisk commented 7 months ago

You can try enable this, and check the media flow, but from what I understand of Asterisk, because it's not a proxy, it actually sets up media exchange between the client (the browser) and itself, and this is done with an encrypted key exchange as part of the media (RTP, not SIP). So this means that on an RTP level, the A and B side is encrypted. This would mean that it's not possible to now re-invite a new B side after the call is established, as the encryption would fail.