I think it is some thing related to strp Could you please send me your configuration file? That will be helpful for me.
On Thu, Jul 3, 2014 at 3:25 PM, bhavik patel <[email protected]> wrote: > 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 > -- 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
