[asterisk-users] packet2packet bridging

Wed Jul 9 03:12:09 CDT 2014

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 --
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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
>>>>
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>>>> 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
>>>
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>>> 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
>
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