I am trying to bridge calls from SIP providers (such as sipgate.de) to jingle-capable jabber clients, using self-hosted asterisk and ejabberd.
If works, but only for clients that agree to accept unencrypted rtp stream (such as dino). However, I want to use Conversations (android xmpp client) in the first place, and that client insists that the media stream is encrypted, and refuses to accept sessions initiated by Motif.
In the documentation, I can find instructions how to enable srtp over sip. But not how to enable it over jingle.
Asterisk version 18.10, but I cannot find any pointers in the modern documentation either.
Any advise? Thanks!