Hi, Please clear me on this topic I am confused
My log show "switching to native rtp". Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? Am I right or wrong ? On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani <[email protected]> wrote: > 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" <[email protected]> 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 <[email protected]> wrote: >> >>> I think you will find that direct audio between two endpoints does not >>> work when NAT is involved. >>> >>> >>> >>> *From:* [email protected] [mailto: >>> [email protected]] *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 <[email protected]> >>> 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 <[email protected]> >>> 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 <[email protected]> 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 > -- 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
