[Freeswitch-users] Codec Settings and some other WebRTC FreeSWITCH v1.4beta questions

[Karsten Horsmann]

Hello all,

branch  v1.4.beta
version 8ca6fe728d34cf4d7431b44c4ff6cada639b8a98
Centos 6.4 64bit
OpenSSL 1.0.0-fips

What are the recommended settings of "global_codec_prefs" and
"outbound_codec_prefs" for webrtc use?

Could you show me an example dialplan snippet how to by-pass-media for
webrtc video from browser to browser?

Is it possible to make an condition if video is send or not?

Thanks!

-- 
Cheers

*Karsten Horsmann* aka witchdoc on irc :)
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