[asterisk-users] packet2packet bridging

[Sameer Rathod]

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:
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>>>
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>>>   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
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