[asterisk-users] packet2packet bridging

Mitul Limbani mitul at enterux.in
Tue Jul 8 13:20:22 CDT 2014


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
>>
>> --
>> _____________________________________________________________________
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>>
>>
>>
>> --
>>
>> 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
>
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