"directrtpsetup=yes" does not work :(

I am trying to pass the SDP payloads as received to both sip endpoints, but “directrtpsetup=yes” in sip.conf does not work :frowning:
SDPs are still be altered.
The asterisk version is
Could you experts tell me how to config this feature, please? Thanks!

I was under the impression that this option causes Asterisk to negotiate an external bridge during the INVITE, if possible, and that is not a request pass the SDP unaltered.

It’s still in testing I believe, so expect problems with it

I read about it from here:
voip-info.org/wiki/view/Aste … ctrtpsetup

So it seems that the article above is misleading.
I read the source code, and indeed, found nothing about unaltered SDP with this config, or I got wrong source code version?