I think you will find that direct audio between two endpoints does not work when NAT is involved.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sameer Rathod Sent: Tuesday, July 08, 2014 11:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] packet2packet bridging Hi Joshua, I had disabled ice support and remover encryption= yes Then also it is showing the same native_rtp in log Could you help me in bypassing asterisk server for audio? please help me I am struggling with it form a long time. On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod <sam...@hostnsoft.com<mailto:sam...@hostnsoft.com>> wrote: -- Channel SIP/1060-0000008e left 'native_rtp' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b> -- Channel SIP/1061-0000008f left 'native_rtp' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b> == Spawn extension (sameer, 1061, 1) exited non-zero on 'SIP/1060-0000008e' here are more generated when I cut the call On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod <sam...@hostnsoft.com<mailto:sam...@hostnsoft.com>> wrote: so In this case If I disable ice support ie commented the icesuppot=yes from all files then also I am getting this output -- Executing [1061@sameer:1] Dial("SIP/1060-0000008e", "SIP/1061") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1061 -- SIP/1061-0000008f is ringing -- SIP/1061-0000008f answered SIP/1060-0000008e -- Channel SIP/1061-0000008f joined 'simple_bridge' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b> -- Channel SIP/1060-0000008e joined 'simple_bridge' basic-bridge <3c12ca41-e180-4fc1-80cf-1339b96da42b> > Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from simple_bridge technology to native_rtp > 0x7f6800039020 -- Probation passed - setting RTP source address to 192.168.1.176:8000<http://192.168.1.176:8000> > 0x7f6780045810 -- Probation passed - setting RTP source address to 192.168.1.191:8000<http://192.168.1.191:8000> On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp <jc...@digium.com<mailto:jc...@digium.com>> wrote: Sameer Rathod wrote: yes I had configured icesupport=yes ; Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com<http://www.digium.com> & www.asterisk.org<http://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 -- 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
