[asterisk-users] [SOLVED] Re: asterisk 13 webrtc

[Marek Cervenka]

dtlsenable=yes was missing

thank you joshua

Dne 21.5.2015 v 22:53 Marek Cervenka napsal(a):
> hi,
>
> is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
>
> or is chan_pjsip better supported?
>
> or the recommended way for asterisk is use respoke.io?
>
>
> my problem with asterisk13+chan_sip+sipml5(the same problem is with 
> SIP.js)
> chan_sip.c:10496 process_sdp: Can't provide secure audio requested in 
> SDP offer "
>
> sip.conf (important parts)
> [vr1a882]
> ...
> nat=force_rport,comedia
> canreinvite=no
> encryption=yes
> avpf=yes
> force_avp=yes
> icesupport=yes
> directmedia=yes
> transport=wss,ws
> dtlsrekey=60
> dtlsverify=no
> dtlscertfile=/etc/pki/tls/certs/rapidssl.crt
> dtlsprivatekey=/etc/pki/tls/private/rapidssl.key
> dtlssetup=actpass
>

-- 
---------------------------------------
Marek Cervenka
=======================================

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