Asterisk 13 Webrtc

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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`
dtlsverify=no dtlscertfile=/etc/pki/tls/certs/rapidssl.crt dtlsprivatekey=/etc/pki/tls/private/rapidssl.key dtlssetup

2 thoughts on - Asterisk 13 Webrtc

  • Hi Marek

    Yes, here is a person with (mostly) working Asterisk 13 (chan_sip) +
    WebRTC (using sipml5 js lib) setup

    You can contact me directly, if you wish, I will try to help if I can

    As of the issue you have.. is it because you’re working with FF 37 as
    browser ?

    I have not come across such issues since last summer, when FF (or Asterisk, don’t remember exactly) had problems with proper DTLS-SRTP
    implementation

    Yours, Kirill

    22.05.2015 23:00, asterisk-users-request@lists.digium.com пишет:

    —

  • dtlsenable=yes was missing

    thank you joshua

    Dne 21.5.2015 v 22:53 Marek Cervenka napsal(a):