[asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

[Ilya Awesome]

Ok, if this is normal why I have oneway audio when nat endpoint calling to local.
if mixmonitor or srtp is enabled audio is ok. 
Issues with native_rtp for sure

Sent from my iPhone

> On 19 Mar 2015, at 23:08, Matthew Jordan <mjordan at digium.com> wrote:
> 
>> On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome <jleed at me.com> wrote:
>> NAT endpoint calling local endpount - switching to native_rtp then no audio,
>> both of them have direct_media=no, Verbose log:
>> 
>>    -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in
>> new stack
>>    -- Launched AGI Script /pbx/agi.php
>>    -- AGI Script Executing Application: (Dial) Options:
>> (PJSIP/99/sip:99 at 192.168.1.73:5060,20)
>>    -- Called PJSIP/99/sip:99 at 192.168.1.73:5060
>>    -- PJSIP/99-00000023 is ringing
>>    -- PJSIP/99-00000023 answered PJSIP/304-00000022
>>    -- Channel PJSIP/304-00000022 joined 'simple_bridge' basic-bridge
>> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28>
>>    -- Channel PJSIP/99-00000023 joined 'simple_bridge' basic-bridge
>> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28>
>>> Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from
>> simple_bridge technology to native_rtp
>>> Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in
>> stack
>>> Locally RTP bridged 'PJSIP/99-00000023' and 'PJSIP/304-00000022' in
>> stack
>>> 0x7f4b50145420 -- Probation passed - setting RTP source address to
>> 194.204.157.200:8972
>>> 0x7f4b5014f140 -- Probation passed - setting RTP source address to
>> 192.168.1.73:5004
>>    -- Channel PJSIP/304-00000022 left 'native_rtp' basic-bridge
>> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28>
>>    -- Channel PJSIP/99-00000023 left 'native_rtp' basic-bridge
>> <da8840bc-9b71-4ca6-b1d8-9565bf8e5e28>
>>    -- <PJSIP/304-00000022>AGI Script /pbx/agi.php completed, returning 4
> 
> Correct - and per the log, they shouldn't be in a direct media bridge:
> 
>> Locally RTP bridged 'PJSIP/99-00000023' and
> 'PJSIP/304-00000022' in stack
>> Locally RTP bridged 'PJSIP/99-00000023' and
> 'PJSIP/304-00000022' in stack
> 
> Locally RTP bridged means media is still flowing through Asterisk, it
> just isn't being decoded and passed through the core.
> 
> 
> -- 
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
> 
> -- 
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