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
