Hey all- I've been working on an Asterisk 13 PJSIP Realtime project that has a requirement to have a WebSocket WSS client attached to the same ps_endpoint device as a "standard" UDP device.
The following settings will make a WebRTC client work properly: ice_support = yes use_avpf = yes force_avpf = no media_use_received_transport = yes media_encryption = dtls media_encryption_optimistic = yes A normal UDP endpoint must have these settings: ice_support = no use_avpf = no force_avpf = no media_use_received_transport = yes media_encryption = dtls media_encryption_optimistic = yes So the net of it is that ice_support and use_avpf being enabled will break a standard unencrypted SIP/UDP endpoint and vice versa. Back in the chan_sip days, we had these settings: Set(CHANNEL(secure_bridge_signaling)=1) Set(CHANNEL(secure_bridge_media)=1) Which could somewhat perform the function of ensuring encrypted calls, but those don't quite match up to PJSIP stack and weren't implemented on this channel type anyway. Just curious if you guys would have any ideas about implementing some sort of a knob to change those two settings on the fly in something like a predial hook in dialplan, or have other ideas on how to make this work better. We can somewhat hack around this with SQL views and other magic, but there are other negative effects from doubling the size of our endpoint and AOR tables for two fields. Happy to do some development, but want to make sure I'm thinking about the problem correctly and the solution would have some utility outside my particular use case. Thoughts? Josh
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