Hi Sameer, I think you should try using public ip rather then local and latest chrome browser. I have also tried with same configuration and same OS with same asterisk version and working fine for me.
On Thu, Jul 3, 2014 at 11:59 AM, Sameer Rathod <[email protected]> wrote: > Hi Bhavik, > > > > This is sip.conf > [general] > > context=public > allowguest=yes > allowoverlap=no > realm=192.168.1.151 > udpbindaddr=0.0.0.0 > icesupport=yes > dtmfmode=rfc2833 > transport=udp,ws > srvlookup=yes > > > [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 > canreinvite=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 > ;directmedia=yes ; Asterisk will relay media for this peer > transport=udp,ws ; Asterisk will allow this peer to register on UDP or > WebSockets > canreinvite=yes > ;directrtpsetup=yes > dtmfmode=rfc2833 > qualify=yes > > > >> http.conf > > [general] > enabled=yes > bindaddr=192.168.1.151 > bindport=8088 > > > > >> rtp.conf > > [general] > rtpstart=10000 > rtpend=20000 > icesupport=true > stunaddr=stun.l.google.com:19302 > > > I am using asterisk 12.3 on centos 6.5 > > > > > > On Thu, Jul 3, 2014 at 10:31 AM, bhavik patel <[email protected]> > wrote: > >> 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 >> > > > > -- > 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
