How to make asterisk work correctly in the media path

Asterisk Asterisk SIP
1

I am connecting from webrtc client to Media Server via Asterisk using Proprietary codec
I want asterisk to decrypt the audio and send unencrypted RTP data (without modifyin RTP data) to Media Server with Payload Type G729 and payload ID :18( I am spoofing proprietary codec with G729)

I have configured on the outgoing trunk towards media server (egress side)

directmedia=yes
directrtpsetup=yes

I can see the call establishment successfull , but audio is not heard, none of my clients are behind NAT

Please let me know , Do i need to configure any extra parameters to get this working

2

directrtpsetup has never been supported.

directmedia cannot be supported on encrypted streams.

You are trying to abuse Asterisk and trying to work out whether, for example, it will try to use the G.729 format module is more work than Iā€™m prepared to do for a deliberate misuse. The correct way of handling this is to write a proper format module.

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