[asterisk-users] packet2packet bridging

Sameer Rathod sameer at hostnsoft.com
Tue Jul 8 09:18:09 CDT 2014


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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>>
>>
>>
>> --
>> Regards
>> Sameer Rathod
>> 8109413462
>>
>>
>
>
> --
> Regards
> Sameer Rathod
> 8109413462
>
>


-- 
Regards
Sameer Rathod
8109413462
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