asterisk users - Dec 2006 - NAT and Dial to two channels at once

We all love Asterisk's ability to Dial(chan1&chan2) and take the first
that
answers.

However, I have been encountering a problem when one of the channels
is an external phone behind NAT and another is a local phone on the
same net as the asterisk server.

All have canreinvite=yes, and the phone behind NAT is correctly
using Stun to give its external ports, which are opened to it
in the firewall, in addition.

When an incoming call comes in for the extension, and it is
sent to a plain old Dial(SIP/chan1) it works -- two way audio,
native bridged.  All happy.  When it's Dial(SIP/outside&SIP/inside)
and the outside channel answers, bad news.

Here's what happens.

a) The invite comes in from DID with DID-SDP.

b) Asterisk sends out two invites, one to SIP/outside and another to
SIP/inside.  However, it put an Asterisk server SDP in the
invite to SIP/outside instread of DID-SDP.

c) The phone answers, and provides asterisk with Outside-SDP.
The phone however got an invite that makes it think it should
be sending audio to Asterisk-SDP, which it does, where I think
it's discarded, as I don't hear it.

d) The DID should be sending audio to the outside phone but
I'm not hearing that either.  (Not sure why as this SDP looks
fine.)


Note that if I press 'hold' on the outside phone, Asterisk
starts music on hold and the caller to the DID hears it.
Press hold again and -- bingo -- the audio is now working.


So, the solution to this long term is that Asterisk needs
to either send a reinvite to the target phone after it answers,
to properly connect its audio with the source, or asterisk has
to decide to locally bridge the audio, and send each endpoint
only an Asterisk SDP.

Any good ways to make this happen, or do we just have to
hope for a fix?


(This may be getting complicated by the inside phone
being on a different network interface, an internal LAN,
while the outside phone is reached via the external
internet.)

Someone may have a more elegant solution, but I have found that allowing
reinvite on a phone connected via NAT gateway causes too many problems,
especially with the difference in the various NAT implementations.  I
set canreinvite=no host=dynamic nat=yes qualify=yes for all phones that
connect from the Internet. 

A note to anyone doing Internet telephony -- the latest Linksys routers
seem to ignore the small UDP keepalive that some phones like the
grandstreams send, so the NAT hole closes.  I've had to start using
qualify=yes to get asterisk to keep the NAT hole open.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brad
Templeton
Sent: Sunday, December 10, 2006 4:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] NAT and Dial to two channels at once


We all love Asterisk's ability to Dial(chan1&chan2) and take the first
that answers.

However, I have been encountering a problem when one of the channels is
an external phone behind NAT and another is a local phone on the same
net as the asterisk server.

All have canreinvite=yes, and the phone behind NAT is correctly using
Stun to give its external ports, which are opened to it in the firewall,
in addition.

When an incoming call comes in for the extension, and it is sent to a
plain old Dial(SIP/chan1) it works -- two way audio, native bridged.
All happy.  When it's Dial(SIP/outside&SIP/inside) and the outside
channel answers, bad news.

Here's what happens.

a) The invite comes in from DID with DID-SDP.

b) Asterisk sends out two invites, one to SIP/outside and another to
SIP/inside.  However, it put an Asterisk server SDP in the invite to
SIP/outside instread of DID-SDP.

c) The phone answers, and provides asterisk with Outside-SDP.
The phone however got an invite that makes it think it should be sending
audio to Asterisk-SDP, which it does, where I think it's discarded, as I
don't hear it.

d) The DID should be sending audio to the outside phone but I'm not
hearing that either.  (Not sure why as this SDP looks
fine.)


Note that if I press 'hold' on the outside phone, Asterisk starts music
on hold and the caller to the DID hears it.
Press hold again and -- bingo -- the audio is now working.


So, the solution to this long term is that Asterisk needs to either send
a reinvite to the target phone after it answers, to properly connect its
audio with the source, or asterisk has to decide to locally bridge the
audio, and send each endpoint only an Asterisk SDP.

Any good ways to make this happen, or do we just have to hope for a fix?


(This may be getting complicated by the inside phone being on a
different network interface, an internal LAN, while the outside phone is
reached via the external
internet.)
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