[Asterisk-Users] Peer to Peer calls

[Obaid Siddiqui]

If I have both clients and Asterisk in the same nat, it is working fine with
internal addressing.

When using outside IP with appropriate ports open(5060,10000-2000), with
following call flow,

X-Lite->asterisk < -- nat-- -> as5400->pstn
canreinvite=no
nat=yes

with this setting RTP's should be between ata186 and astersik over nat.

I can see bi-directional RTP streams in Ethereal ( *asterisk<->x-lite* ),
very few of them from Asterisk to X-Lite, resulting one-way audio, and the
call is disconnected abruptly after that.

I have setup g711, on X-Lite and SIP.conf, but still it is negotiating "gsm"
with AS5400.
Eventually I wan to use clients on different nats, to work with Asterisk on
different nat.

Is this a codec issue, or asterisk problem or nat? can some body help,
probably I need proxy.

Obaid Siddiqui.
Network Engineer,
Prizm Communications, LP
Austin, Texas.




----- Original Message ----- 
From: "Michael J. Tubby G8TIC" <mike.tubby at thorcom.co.uk>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Sunday, May 29, 2005 7:13 PM
Subject: Re: [Asterisk-Users] Peer to Peer calls


>
> ----- Original Message ----- 
> From: "Michiel van Baak" <michiel at vanbaak.info>
> To: <asterisk-users at lists.digium.com>
> Sent: Sunday, May 29, 2005 10:41 PM
> Subject: Re: [Asterisk-Users] Peer to Peer calls
>
>
> > On 00:32, Mon 30 May 05, Cenk Yabas wrote:
> >> Can anybody please answer this.
> >> Both clients are behind different NAT's.
> >> One of them starts a SIP call to the other through Asterisk.
> >> Asterisk sets up the call.
> >> Issues reinvite and connects them together.
> >> After this point does the media stream flow through Asterisk or Peer to
> >> Peer?
> >> Does such a call use any system resources of Asterisk server after
> >> connection?
> >> Thank you in advance.
> >
> > Did you test this ?
> > My experience is the 'reinvite' does not work in the setup
> > you descripted. I always have to set 'canreinvite=no' in
> > asterisk config or the audio will not come through.
> > If you have only one phone on both NAT's and you can do
> > port-forwording on both firewalls, it can work, but that
> > scenario is highly uncommon.
> > The audio stream is setup on some random port, so your
> > firewall will block this by default.
> >
>
> *But* If your firewall is SIP-aware - for example a Cisco 837 ADSL
> router with IOS 12.3 - then it should be able to fix up the firewall rules
> dynamically so that when the phones in the inside (behind the firewall)
> re-invite it should inspect the SIP on udp/5060 and see the invitation and
> open the appropriate UDP port(s) for the RTP stream.
>
> Mike
>
>
>
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