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

2015-03-22 Thread 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  wrote:
> 
>> On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome  wrote:
>> NAT endpoint calling local endpount - switching to native_rtp then no audio,
>> both of them have direct_media=no, Verbose log:
>> 
>>-- Executing [99@dialmap:1] AGI("PJSIP/304-0022", "/pbx/agi.php") in
>> new stack
>>-- Launched AGI Script /pbx/agi.php
>>-- AGI Script Executing Application: (Dial) Options:
>> (PJSIP/99/sip:99@192.168.1.73:5060,20)
>>-- Called PJSIP/99/sip:99@192.168.1.73:5060
>>-- PJSIP/99-0023 is ringing
>>-- PJSIP/99-0023 answered PJSIP/304-0022
>>-- Channel PJSIP/304-0022 joined 'simple_bridge' basic-bridge
>> 
>>-- Channel PJSIP/99-0023 joined 'simple_bridge' basic-bridge
>> 
>>> Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from
>> simple_bridge technology to native_rtp
>>> Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
>> stack
>>> Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' 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/304-0022 left 'native_rtp' basic-bridge
>> 
>>-- Channel PJSIP/99-0023 left 'native_rtp' basic-bridge
>> 
>>-- 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/99-0023' and
> 'PJSIP/304-0022' in stack
>> Locally RTP bridged 'PJSIP/99-0023' and
> 'PJSIP/304-0022' 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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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

2015-03-19 Thread Matthew Jordan 
On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome  wrote:
> NAT endpoint calling local endpount - switching to native_rtp then no audio,
> both of them have direct_media=no, Verbose log:
>
> -- Executing [99@dialmap:1] AGI("PJSIP/304-0022", "/pbx/agi.php") in
> new stack
> -- Launched AGI Script /pbx/agi.php
> -- AGI Script Executing Application: (Dial) Options:
> (PJSIP/99/sip:99@192.168.1.73:5060,20)
> -- Called PJSIP/99/sip:99@192.168.1.73:5060
> -- PJSIP/99-0023 is ringing
> -- PJSIP/99-0023 answered PJSIP/304-0022
> -- Channel PJSIP/304-0022 joined 'simple_bridge' basic-bridge
> 
> -- Channel PJSIP/99-0023 joined 'simple_bridge' basic-bridge
> 
>> Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from
> simple_bridge technology to native_rtp
>> Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
> stack
>> Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' 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/304-0022 left 'native_rtp' basic-bridge
> 
> -- Channel PJSIP/99-0023 left 'native_rtp' basic-bridge
> 
> -- 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/99-0023' and
'PJSIP/304-0022' in stack
   > Locally RTP bridged 'PJSIP/99-0023' and
'PJSIP/304-0022' 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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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

2015-03-18 Thread Nick Awesome 
NAT endpoint calling local endpount - switching to native_rtp then no audio, 
both of them have direct_media=no, Verbose log:

-- Executing [99@dialmap:1] AGI("PJSIP/304-0022", "/pbx/agi.php") in 
new stack
-- Launched AGI Script /pbx/agi.php
-- AGI Script Executing Application: (Dial) Options: 
(PJSIP/99/sip:99@192.168.1.73:5060,20)
-- Called PJSIP/99/sip:99@192.168.1.73:5060
-- PJSIP/99-0023 is ringing
-- PJSIP/99-0023 answered PJSIP/304-0022
-- Channel PJSIP/304-0022 joined 'simple_bridge' basic-bridge 

-- Channel PJSIP/99-0023 joined 'simple_bridge' basic-bridge 

   > Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from 
simple_bridge technology to native_rtp
   > Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in 
stack
   > Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' 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/304-0022 left 'native_rtp' basic-bridge 

-- Channel PJSIP/99-0023 left 'native_rtp' basic-bridge 

-- AGI Script /pbx/agi.php completed, returning 4


> On 18 Mar 2015, at 18:26, Matthew Jordan  wrote:
> 
> On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome  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
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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

2015-03-18 Thread Matthew Jordan 
On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome  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

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

2015-03-18 Thread Nick Awesome 
Well, it breaks audio for all NAT endpoints, how can I fix this?

> On 18 Mar 2015, at 15:48, Matthew Jordan  wrote:
> 
> On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome  > 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://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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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

2015-03-18 Thread Matthew Jordan 
On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome  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://asterisk.org

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

2015-03-18 Thread Nick Awesome 
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.

Asterisk 13.2.0-- 
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