asterisk users - Aug 2015 - One way audio - doesn't seem to be NAT issue - SOLVED!

I have been banging my head against the wall for weeks now on this
one.  I have a switch running NetBSD and Asterisk 11.19.0 although I
have had this problem on older versions as well.  I, and my users, can
call out, we can receive calls, quality is excellent but I cannot talk
with one user.  The different elements are as follows:

The switch as described above which is in a server room on the Internet
backbone with a public IP address.

My home system which is behind a bridged modem through a Linksys
WRT54GS with priority given to my ATA.  The ATA is a Cisco SPA112.  I
also have an actual SIP phone.  The problem happens with both.
Obviously I am using NAT but both devices work just fine if I am going
to the PSTN.

My user who is also going through a bridged modem to a Linksys SPA-2102
which is doing the PPPOE so it has a public IP address and no NAT
involved although it serves NAT for the connected computer.

So here is the problem.  While both of us have no problems externally,
when we call each other we get one way audio and it is always from me
to him no matter who initiates the call.

A further test, I can call from the SIP phone to the ATA connected
phone and vice versa just fine.  That involves two devices behind the
same NAT but since they still need to use the server as an intermediary
I can't see how that would matter.

Given that both of us can make and accept calls and the server is
simply connecting two separate channels I can't see where the problem
might lie.  Can anyone suggest a possible setup issue?

I have tried so many things but I am willing to try them again.  Feel
free to make any suggestion no matter how silly.  I really need to fix
this.

Cheers.


-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net

On Tue, Aug 11, 2015, at 04:10 PM, D'Arcy J.M. Cain
wrote:
> I have been banging my head against the wall for weeks now on this
> one.  I have a switch running NetBSD and Asterisk 11.19.0 although I
> have had this problem on older versions as well.  I, and my users, can
> call out, we can receive calls, quality is excellent but I cannot talk
> with one user.  The different elements are as follows:
<snip>

I'd suggest getting a packet capture to see the RTP traffic to see the
actual path of things, not just thinking of what it should be. Media
doesn't just get lost. It's told to go somewhere ultimately and either
that is incorrect for some reason or something is blocking it.

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

Not 100% ure, but maybe play with the canreinvite or directmedia settings.

On Wed, Aug 12, 2015 at 3:10 AM, D'Arcy J.M. Cain <darcy at vex.net>
wrote:
> I have been banging my head against the wall for weeks now on this
> one.  I have a switch running NetBSD and Asterisk 11.19.0 although I
> have had this problem on older versions as well.  I, and my users, can
> call out, we can receive calls, quality is excellent but I cannot talk
> with one user.  The different elements are as follows:
>
> The switch as described above which is in a server room on the Internet
> backbone with a public IP address.
>
> My home system which is behind a bridged modem through a Linksys
> WRT54GS with priority given to my ATA.  The ATA is a Cisco SPA112.  I
> also have an actual SIP phone.  The problem happens with both.
> Obviously I am using NAT but both devices work just fine if I am going
> to the PSTN.
>
> My user who is also going through a bridged modem to a Linksys SPA-2102
> which is doing the PPPOE so it has a public IP address and no NAT
> involved although it serves NAT for the connected computer.
>
> So here is the problem.  While both of us have no problems externally,
> when we call each other we get one way audio and it is always from me
> to him no matter who initiates the call.
>
> A further test, I can call from the SIP phone to the ATA connected
> phone and vice versa just fine.  That involves two devices behind the
> same NAT but since they still need to use the server as an intermediary
> I can't see how that would matter.
>
> Given that both of us can make and accept calls and the server is
> simply connecting two separate channels I can't see where the problem
> might lie.  Can anyone suggest a possible setup issue?
>
> I have tried so many things but I am willing to try them again.  Feel
> free to make any suggestion no matter how silly.  I really need to fix
> this.
>
> Cheers.
>
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:darcy at Vex.Net
> VoIP: sip:darcy at Vex.Net
>
> --
> _____________________________________________________________________
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-- 
Michael Dupree jr.
p: +1-248-935-4147
f: +1-866-671-6867
Skype: MichaelDupreeJr
PGP Pub Key: http://www.michaeldupree.net/?page_id=53









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On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree <michael at easybitllc.com> wrote:
> Not 100% ure, but maybe play with the canreinvite or directmedia
> settings.
Yes!  That was it.  Just for future searches here is what I did.  I
added "directmedia = no" in sip.conf.  This fixed the issue.

I believe that Asterisk was getting confused when one leg was inside
NAT and the other was outside.  Perhaps there was an "OR" where there
should be an "AND".  It makes sense because the other user was the one
outside NAT and he could hear me and I could not hear him no matter who
initiated the call.  He could make outside calls because both he and my
provider were on public IPs.

I am not sure why this hasn't bit anyone else.  Perhaps most Asterisk
systems are in one of two classes, connecting to all NAT phones or
connecting to all public phones, and I am in a minority situation where
I am talking to a mix of setups.

Thanks for that.  I was going nuts trying to figure this out.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net