Re: packet2packet bridging

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Not that I am aware of.

 

From: asterisk-users-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Sameer Rathod
Sent: Tuesday, July 08, 2014 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] packet2packet bridging

 

Hi Eric,

I am behind nat

Is there any solution for the same.

My goal is to deduct the balance
for the call but free my asterisk server from audio packet load.

 

On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling <EWieling@xxxxxxxxx> wrote:

I think you will find that direct audio between two endpoints does not work when NAT is involved.  

 

From: asterisk-users-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Sameer Rathod
Sent: Tuesday, July 08, 2014 11:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] packet2packet bridging

 

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@xxxxxxxxxxxxx> 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@xxxxxxxxxxxxx> 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@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@xxxxxxxxxx> 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

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--

Regards

Sameer Rathod

8109413462 

 




--

Regards

Sameer Rathod

8109413462 

 




--

Regards

Sameer Rathod

8109413462 

 


--
_____________________________________________________________________
-- 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




--

Regards

Sameer Rathod

8109413462 

 

-- 
_____________________________________________________________________
-- 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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To UNSUBSCRIBE or update options visit:
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