use tcpdump on the server to see if the RTP traffic is passing through it.
On 9 July 2014 10:48, Sameer Rathod <[email protected]> wrote: > 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 > -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: [email protected] w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552
-- _____________________________________________________________________ -- 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
