I had also tried with asterisk 11.10.2 no I am getting
== Using SIP RTP CoS mark 5 [Jul 3 15:45:10] WARNING[29686][C-00000001]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 9191 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126 followed this link http://sipjs.com/guides/server-configuration/asterisk/ following are the configuration I did [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=1060 ; 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 context=sameer ; Tell Asterisk which context to use when this peer is dialing directmedia=no ; Asterisk will relay media for this peer transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets [1061] ; This will be the legacy SIP client type=friend username=1061 host=dynamic secret=1061 context=sameer On Thu, Jul 3, 2014 at 3:34 PM, Sameer Rathod <[email protected]> wrote: > 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 > > -- 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
