yes I had configured icesupport=yes ;
on both the client in sip.con as well as did the setting of ice in rtp.conf also here is my sip configuration [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 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 ;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 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 dtmfmode=rfc2833 qualify=yes On Wed, Jul 2, 2014 at 8:00 PM, Joshua Colp <[email protected]> wrote: > Sameer Rathod wrote: > >> = Using SIP RTP CoS mark 5 >> -- Executing [1061@sameer:1] Dial("SIP/1060-00000088", "SIP/1061") >> in new stack >> == Using SIP RTP CoS mark 5 >> -- Called SIP/1061 >> -- SIP/1061-00000089 is ringing >> > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to >> 192.168.1.176:8000 <http://192.168.1.176:8000> >> >> -- SIP/1061-00000089 answered SIP/1060-00000088 >> -- Channel SIP/1061-00000089 joined 'simple_bridge' basic-bridge >> <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4> >> -- Channel SIP/1060-00000088 joined 'simple_bridge' basic-bridge >> <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4> >> > Bridge 1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4: switching from >> simple_bridge technology to native_rtp >> > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to >> 192.168.1.176:8000 <http://192.168.1.176:8000> >> >> > 0x7f6780047090 -- Probation passed - setting RTP source address to >> 192.168.1.191:8000 <http://192.168.1.191:8000> >> >> == WebSocket connection from '192.168.1.191:54390 >> <http://192.168.1.191:54390>' closed >> > > Are either side using encryption or ICE? > > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- 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
