asterisk users - Mar 2015 - Asterisk switching bridge to native_rtp even with direct_media=no

Well, it breaks audio for all NAT endpoints, how can I fix this?
> On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com>
wrote:
> 
> On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com
<mailto:jleed at me.com>> wrote:
>> Hey guys,
>> 
>> have issues with reinvite, no matter what endpoint is calling asterisk
>> always tries switch simple_bridge to native_rtp
>> 
>> Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from
simple_bridge
>> technology to native_rtp
>> 
>> in endpoints table ?direct_media? sets to ?no? on all endpoints but it
>> doesn?t help.
>> 
>> if native_rtp not work for some reason I have oneway audio. how can I
fix
>> this? if I add mix_monitor it works, but it?s not a right way to fix
this
>> issues.
>> 
> 
> A native_rtp bridge is used for more than direct media. It is also
> used for local native bridging, that is, when you have two RTP capable
> channels in a bridge and Asterisk does not require the media to flow
> through its core. The bridge then just performs a packet to packet
> swap between the two RTP capable channels.
> 
> Note that on verbosity 4, Asterisk will tell you if the bridge is
> locally or remotely bridging the two channels.
> 
> -- 
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com <http://digium.com/> &
http://asterisk.org <http://asterisk.org/>
> 
> -- 
> _____________________________________________________________________
> -- 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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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com>
wrote:
> Well, it breaks audio for all NAT endpoints, how can I fix this?
>
Local (packet to packet) bridging should not do that. Remote (direct
media) can do that.

Can you confirm - by looking at a verbose level 4 log - how Asterisk
is bridging the two channels?

-- 
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

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

> On 18 Mar 2015, at 18:26, Matthew Jordan <mjordan at digium.com>
wrote:
> 
> On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome <jleed at me.com>
wrote:
>> Well, it breaks audio for all NAT endpoints, how can I fix this?
>> 
> 
> Local (packet to packet) bridging should not do that. Remote (direct
> media) can do that.
> 
> Can you confirm - by looking at a verbose level 4 log - how Asterisk
> is bridging the two channels?
> 
> -- 
> 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
> 
> -- 
> _____________________________________________________________________
> -- 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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