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[sameer at hostnsoft.com]

Hi,


I am new to asterisk I want to configure my asterisk server such that it only establishes the call

rest the audio must bypass the server and transmitted directly to the peer


In my config file I did changes which are below


canreinvite=yes

nat=force_rtp

dirtectmedia=yes

directsetup=yes


I am using asterisk version 12.3

--
Regards
Sameer Rathod8109413462 

[jcolp at digium.com]

Sameer Rathod wrote:

[sameer at hostnsoft.com]

= Using SIP RTP CoS mark 5
    -- Executing [1061@sameer:1] Dial("SIP/1060-00000088", "SIP/1061") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1061
    -- SIP/1061-00000089 is ringing
       > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to 192.168.1.176:8000
    -- SIP/1061-00000089 answered SIP/1060-00000088
    -- Channel SIP/1061-00000089 joined 'simple_bridge' basic-bridge <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
    -- Channel SIP/1060-00000088 joined 'simple_bridge' basic-bridge <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
       > Bridge 1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4: switching from simple_bridge technology to native_rtp
       > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to 192.168.1.176:8000
       > 0x7f6780047090 -- Probation passed - setting RTP source address to 192.168.1.191:8000
  == WebSocket connection from '192.168.1.191:54390' closed



It is giving me following output on asterisk console




On Wed, Jul 2, 2014 at 6:30 PM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:

[jcolp at digium.com]

Sameer Rathod wrote:

[sameer at hostnsoft.com]

yes I had configured

icesupport=yes ;


on both the client in sip.con


as well as did the setting of ice in rtp.conf also


here is my sip configuration

[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=sameer ; The SIP Password for SIP.js
icesupport=yes ; Tell Asterisk to use ICE for this peer
ignorecryptolifetime=yes
context=sameer ; Tell Asterisk which context to use when this peer is dialing
;directmedia=yes ; Asterisk will relay media for this peer
transport=udp,ws ;Asterisk will allow this peer to register on UDP or WebSockets
canreinvite=yes
;directrtpsetup=yes
;nat=force_rtp,comedia
dtmfmode=rfc2833
qualify=yes

[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=sameer
context=sameer
ignorecryptolifetime=yes
;nat=force_rtp,comedia
icesupport=yes ; Tell Asterisk to use ICE for this peer
;directmedia=yes ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
canreinvite=yes
dtmfmode=rfc2833
qualify=yes






On Wed, Jul 2, 2014 at 8:00 PM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:

[jcolp at digium.com]

Sameer Rathod wrote:

[sameer at hostnsoft.com]

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@digium.com (jcolp@digium.com)> wrote:

[sameer at hostnsoft.com]

 -- 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@hostnsoft.com (sameer@hostnsoft.com)> wrote:

[sameer at hostnsoft.com]

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@hostnsoft.com (sameer@hostnsoft.com)> wrote:

[EWieling at nyigc.com]

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

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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@hostnsoft.com (sameer@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@hostnsoft.com (sameer@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@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>

[sameer at hostnsoft.com]

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@nyigc.com (EWieling@nyigc.com)> wrote:

[EWieling at nyigc.com]

Not that I am aware of.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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@nyigc.com (EWieling@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@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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@hostnsoft.com (sameer@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@hostnsoft.com (sameer@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@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>

[mitul at enterux.in]

No way to avoid bw charges for any of the client if it is behind any sort of NAT. On 08-Jul-2014 8:52 PM, "Sameer Rathod" <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:

[sameer at hostnsoft.com]

Hi,


Please clear me on this topic I am confused


My log show "switching to native rtp". 
Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ?


Am I right or wrong ?



On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani <mitul@enterux.in (mitul@enterux.in)> wrote:

[mitul at enterux.in]

Put sip debug on to know if reinvite packets are sent.
On 09-Jul-2014 1:17 PM, "Sameer Rathod" <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote: