Hi bhavik, By following the same tutorial I am getting this error currently
*Can't provide secure audio requested in SDP offer* I think it is related to the srtp issue of asterisk Please help me in this I am struggling with this form a long time On Wed, Jul 2, 2014 at 8:21 PM, bhavik patel <[email protected]> wrote: > Hi, > > For SIpml5 tried to configure by this way : > https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 > This is working fine for me. > > > > > On Wed, Jul 2, 2014 at 8:06 PM, Sameer Rathod <[email protected]> > wrote: > >> Hi, >> >> I am getting >> *Can't provide secure audio requested in SDP offer* >> >> with sipml5 client hosted on my local system >> >> >> >> [1060] ; This will be WebRTC client >> type=friend >> username=1060 ; The Auth user for SIP.js >> host=dynamic ; Allows any host to register >> secret=sameer ; The SIP Password for SIP.js >> encryption=yes ; Tell Asterisk to use encryption for this peer >> avpf=yes ; Tell Asterisk to use AVPF for this peer >> icesupport=yes ; Tell Asterisk to use ICE for this peer >> ignorecryptolifetime=yes >> context=sameer ; Tell Asterisk which context to use when this peer is >> dialing >> ;directmedia=yes ; Asterisk will relay media for this peer >> transport=udp,ws ;Asterisk will allow this peer to register on UDP or >> WebSockets >> ;disallow=allow >> ;allow=vp8 >> canreinvite=yes >> ;directrtpsetup=yes >> nat=force_rtp,comedia >> dtmfmode=rfc2833 >> qualify=yes >> >> [1061] ; This will be the legacy SIP client >> type=friend >> username=1061 >> host=dynamic >> secret=sameer >> context=sameer >> ignorecryptolifetime=yes >> nat=force_rtp,comedia >> encryption=yes >> avpf=yes ; Tell Asterisk to use AVPF for this peer >> icesupport=yes ; Tell Asterisk to use ICE for this peer >> ;context=default ; Tell Asterisk which context to use when this peer is >> dialing >> ;directmedia=yes ; Asterisk will relay media for this peer >> transport=udp,ws ; Asterisk will allow this peer to register on UDP or >> WebSockets >> ;disallow=allow >> ;allow=vp8 >> canreinvite=yes >> ;directrtpsetup=yes >> dtmfmode=rfc2833 >> qualify=yes >> >> >> >> >> This is my sip.conf >> >> >> on the one side I am using zoiper client with 1060 (same pc with ip >> 192.168.1.191) >> and for second client I am using sipml5 on chrome >> >> both the client displays a message Not acceptable here >> >> I am using asterisk 12.3 >> >> == WebSocket connection from '192.168.1.191:55561' for protocol 'sip' >> accepted using version '13' >> -- Registered SIP '1061' at 192.168.1.191:55561 >> > Saved useragent "IM-client/OMA1.0 sipML5-v1.2014.04.18" for peer >> 1061 >> == Using SIP RTP CoS mark 5 >> [Jul 2 19:57:04] WARNING[26672][C-00000071]: chan_sip.c:10648 >> process_sdp: Can't provide secure audio requested in SDP offer >> >> >> If any more information is needed please let me know >> >> My goal is do do peer to peer calling with asterisk+webrtc (i.e. webphone) >> >> >> >> >> >> >> >> >> >> -- >> Regards >> Sameer Rathod >> 8109413462 >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Thanks, > Bhavik Patel > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Regards Sameer Rathod 8109413462
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
