asterisk users - Apr 2018 - Asterisk behind NAT Early Media Video

Yes, media is flowing through Asterisk because both client's are behind
different NAT's.

Do I need to do something special in the Call Flow? Or anything additional
to the pjsip.conf?

2018-04-09 16:50 GMT+02:00 Joshua Colp <jcolp at digium.com>:
> On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote:
> > Hello,
> >
> > I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2).
> >
> > Now I would like to get Early Media Video working between clients in
> > different NATed networks. The 183 signalling goes trough perfectly,
but
> > asterisk doesn't forward the Early Media RTP stream from the
caller to
> the
> > recipent.
>
> You would need to examine things specifically and see where media is
> flowing. Is the recipient behind NAT? If so then until we receive media
> from them (wich may or may not occur with early media) we may not have the
> correct target of media.
>
> Cheers,
>
> --
> 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 --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote:
> Yes, media is flowing through Asterisk because both client's are behind
> different NAT's.
This doesn't answer the question of what is ACTUALLY happening in the
scenario you describe which is very important.
 
> Do I need to do something special in the Call Flow? Or anything additional
> to the pjsip.conf?
The "rtp_symmetric" option as you've used causes Asterisk to send
media to the source of media, but it requires us to receive media. If we
don't receive it then we send media to where they've told us to send it,
which as I've mentioned can be wrong.

-- 
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

My understanding based on Wireshark analysis is that the signaling works
(also the recipent phone is displaying the video frame before accepting the
call), also the calling phone send video (i see that also via Wireshark)
but the recipent phone doesn't get any video from the Asterisk before the
call.

2018-04-09 17:02 GMT+02:00 Joshua Colp <jcolp at digium.com>:
> On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote:
> > Yes, media is flowing through Asterisk because both client's are
behind
> > different NAT's.
>
> This doesn't answer the question of what is ACTUALLY happening in the
> scenario you describe which is very important.
>
> > Do I need to do something special in the Call Flow? Or anything
> additional
> > to the pjsip.conf?
>
> The "rtp_symmetric" option as you've used causes Asterisk to
send media to
> the source of media, but it requires us to receive media. If we don't
> receive it then we send media to where they've told us to send it,
which as
> I've mentioned can be wrong.
>
> --
> 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 --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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