Put sip debug on to know if reinvite packets are sent. On 09-Jul-2014 1:17 PM, "Sameer Rathod" <[email protected]> wrote:
> 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 >
-- _____________________________________________________________________ -- 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
