[asterisk-users] packet2packet bridging

[Eric Wieling]

Not that I am aware of.

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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 at nyigc.com<mailto:EWieling at nyigc.com>> wrote:
I think you will find that direct audio between two endpoints does not work when NAT is involved.

From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [mailto:asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] 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 at hostnsoft.com<mailto:sameer at hostnsoft.com>> 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 at hostnsoft.com<mailto:sameer at hostnsoft.com>> 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 at 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<http://192.168.1.176:8000>
       > 0x7f6780045810 -- Probation passed - setting RTP source address to 192.168.1.191:8000<http://192.168.1.191:8000>




On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp <jcolp at digium.com<mailto:jcolp at digium.com>> 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<http://www.digium.com> & www.asterisk.org<http://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

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