[asterisk-users] packet2packet bridging

[Eric Wieling]

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] 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

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