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Nick Awesome

2015-Mar-19 06:47 UTC

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

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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Matthew Jordan

2015-Mar-19 20:08 UTC

head link

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

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

Ilya Awesome

2015-Mar-23 04:43 UTC

head link

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

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
> 
> -- 
> _____________________________________________________________________
> -- 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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