Hi Mitul, I checked that the re-invite packet are sent what I want to check is whether the audio packets is going through the server or not ?
On Wed, Jul 9, 2014 at 1:42 PM, Mitul Limbani <[email protected]> wrote: > 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 > -- 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
