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
