VoIP Mailing List Archives

[darcy at Vex.Net]

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@Vex.Net
VoIP: sip:darcy@Vex.Net

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[jcolp at digium.com]

On Tue, Aug 11, 2015, at 04:10 PM, D'Arcy J.M. Cain wrote:

[viljoens at verishare....]

Hi D'arcy

Have you checked your RTP port ranges (I'm sure you have), and also that the
server IP for RTP as specified in the initial SIP is correct?

Not sure how this will relate to your setup, but we had something similar
here using Asterisk 1.8.11.0 on both sides of the connection, via a VOIP
service provider in the middle.

We had slightly different parameters, e. g. that we would have no RTP at
all, but a call that did connect to total silence, dialed from either side.

We subscribe to two trunk numbers provided by the VOIP service provider at
each site in Asterisk.

It turned out after carefully looking at the SIP flowing back and forth that
the service provider was providing an RTP server IP that specified not the
same IP as the SIP server (which is their standard practice) but a
-different- RTP server IP.

Due to the routing we have, neither system on either side of the SIP
negotiated call could send packets to this "new" RTP server IP.

We therefore added a route that specifically allowed that "new" RTP server
IP to be reached by both machines on both sides of the VOIP service provider
link.

So can you carefully check that the SIP-negotiated RTP streams are going to
IPs that are reachable in BOTH directions?

Also check what RTP port ranges are being used - I have had this
one-directional problem where the port range in /etc/asterisk/rtp.conf was
too broad, and the firewall on my server was only allowing a smaller subset
of RTP ports.

E. g. /etc/asterisk/rtp.conf specified 10000 - 50000 as allowable RTP ports,
but my firewalld firewall under Centos was only allowing 10000 - 20000 - so
I'd regularly get that my SECOND call to test the server would have audio in
one direction - because
Asterisk was allocating an RTP port on one side of the SIP call that was
outside the range my firewalld was allowing.

It might require some careful tracing of SIP messages, maybe you can try
this? Specifically try to determine what RTP port number is being negotiated
when you have your zero-audio back from the remote party - what RTP port and
RTP server IP is he using at that moment on his side?

Is that port allowed through all the PPP / network segments between you? Is
the IP / IPs between you used to transfer RTP reachable from his side?

Message: 1
Date: Tue, 11 Aug 2015 15:10:44 -0400
From: "D'Arcy J.M. Cain" <darcy@Vex.Net>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] One way audio - doesn't seem to be NAT issue
Message-ID: <20150811151044.79872ce9@imp>
Content-Type: text/plain; charset=US-ASCII

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.


--
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[darcy at Vex.Net]

On Thu, 13 Aug 2015 10:41:31 +0200
"Stefan Viljoen" <viljoens@verishare.co.za> wrote:

[viljoens at verishare....]

Hi D'Arcy

[michael at easybitllc.com]

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@vex.net (darcy@vex.net)> wrote: