Hi Sameer, Provide me your Asterisk Configuration,may be i can help you. Also provide me system configuration.
If you need more help then you can post Sipml5 forum https://groups.google.com/forum/#!forum/doubango. That way your issue may resolve. On Wed, Jul 2, 2014 at 8:35 PM, Sameer Rathod <[email protected]> wrote: > 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 > -- 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
