No way to avoid bw charges for any of the client if it is behind any sort of NAT. On 08-Jul-2014 8:52 PM, "Sameer Rathod" <sam...@hostnsoft.com> wrote:
> Hi Eric, > > > I am behind nat > > Is there any solution for the same. > > My goal is to deduct the balance > for the call but free my asterisk server from audio packet load. > > > On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling <ewiel...@nyigc.com> wrote: > >> 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> >> 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> >> 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 >> > 0x7f6780045810 -- Probation passed - setting RTP source address >> to 192.168.1.191:8000 >> >> >> >> >> >> >> On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp <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 & 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 >> > > > > -- > 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 >
-- _____________________________________________________________________ -- 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