[asterisk-users] packet2packet bridging
Sameer Rathod
sameer at hostnsoft.com
Wed Jul 9 04:48:49 CDT 2014
Hi Mitul,
I checked that the re-invite packet are sent what I want to check is
whether the audio packets is going through the server or not ?
On Wed, Jul 9, 2014 at 1:42 PM, Mitul Limbani <mitul at enterux.in> wrote:
> Put sip debug on to know if reinvite packets are sent.
> On 09-Jul-2014 1:17 PM, "Sameer Rathod" <sameer at hostnsoft.com> wrote:
>
>> 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 at enterux.in> wrote:
>>
>>> 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 at hostnsoft.com> wrote:
>>>
>>>> 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>
>>>> 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] *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>
>>>>> 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>
>>>>> 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
>>>>> > 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 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 & www.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
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>>
>>>>> 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
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> --
> _____________________________________________________________________
> -- 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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