[asterisk-users] packet2packet bridging

Ishfaq Malik ish at pack-net.co.uk
Wed Jul 9 04:54:47 CDT 2014 

use tcpdump on the server to see if the RTP traffic is passing through it.


On 9 July 2014 10:48, Sameer Rathod <sameer at hostnsoft.com> 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 <mitul at enterux.in> wrote:
>
>> Put sip debug on to know if reinvite packets are sent.
>>  On 09-Jul-2014 1:17 PM, "Sameer Rathod" <sameer at hostnsoft.com> 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 <mitul at enterux.in> 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" <sameer at 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 <EWieling at nyigc.com>
>>>>> wrote:
>>>>>
>>>>>> I think you will find that direct audio between two endpoints does
>>>>>> not work when NAT is involved.
>>>>>>
>>>>>>
>>>>>>
>>>>>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>>>>>> asterisk-users-bounces at 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 <sameer at 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 <sameer at 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 at 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 <jcolp at 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
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>>>>>> 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
>>>>
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>>>>    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: ish at pack-net.co.uk
w: http://www.pack-net.co.uk

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