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[asterisk-users] packet2packet bridging

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sameer at hostnsoft.com
Guest

Posted: Wed Jul 09, 2014 4:49 am Post subject: [asterisk-users] packet2packet bridging

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@enterux.in (mitul@enterux.in)> wrote:

Quote:

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:

Quote:

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:

Quote:

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:

Quote:

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:

Quote:

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>
       > 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 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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New to Asterisk? Join us for a live introductory webinar every Thurs:
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--
Regards

Sameer Rathod
[url=tel:8109413462]8109413462[/url]

--
Regards

Sameer Rathod
[url=tel:8109413462]8109413462[/url]

--
Regards

Sameer Rathod
[url=tel:8109413462]8109413462[/url]

--
_____________________________________________________________________
-- 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:
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--
Regards
Sameer Rathod [url=tel:8109413462]8109413462[/url]

--
_____________________________________________________________________
-- 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

ish at pack-net.co.uk
Guest

Posted: Wed Jul 09, 2014 4:55 am Post subject: [asterisk-users] packet2packet bridging

use tcpdump on the server to see if the RTP traffic is passing through it.

On 9 July 2014 10:48, Sameer Rathod <sameer@hostnsoft.com (sameer@hostnsoft.com)> wrote:

Quote:

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:

Quote:

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:

Quote:

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish@pack-net.co.uk (ish@pack-net.co.uk)
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552

mjordan at digium.com
Guest

Posted: Wed Jul 09, 2014 7:59 am Post subject: [asterisk-users] packet2packet bridging

On Wed, Jul 9, 2014 at 2:47 AM, Sameer Rathod <sameer@hostnsoft.com> wrote:

Quote:

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 ?

You are wrong (sorry).

All that means is that the bridging has switched to a native RTP
bridge. That bridge comes in two variants: a local packet to packet
bridge (where the media flows through Asterisk but is not decoded -
RTP is merely swapped between ports) and a remote bridge. The remote
bridge is where the two channels are in a bridge in Asterisk, but
media flows directly between the endpoints.

If your endpoints are behind a NAT, then no, you cannot use a remote
bridge. No amount of hoping or tinkering will make it so.

--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://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

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