[Asterisk-Users] conditional canreinvite

[hugolivude]

Guys,

Have a look for my posting:

"How to keep Asterisk (1.2) out of the media path"

A gentleman named Tony Jago provided some awesome info.  I've posted it
below, but you might want to look at my posting for context:

1) Could someone confirm that I'll need to have canreinvite=yes in
sip.conf for both the Xlite and the Polycoms in order to bypass * from
the media path?

This is correct.

2) Does the Polycom & XLite support reinvite?

I believe so.

3) Does reinvite work if you're behind a nat?  i.e. if I have nat=yes,
does this mean I _have_ to have canreinvite=no?

No. You need to have the nat set up correctly. This means you need to put
port forwarding rules in for each and every phone for it's sip and rtp
ports. This means you will have to reconfigure each phone to use a different
port. eg.

phone 10.0.0.1. SIP port 5060 and RTP ports 8001-8010.
phone 10.0.0.2  SIP port 5061 and RTP ports 8011-8020.
phone 10.0.0.3. SIP port 5062 and RTP ports 8021-8030.
etc etc.

on your firewall, you need to map incoming ports 5060 -> 10.0.0.1 and
8001-8010 -> 10.0.0.1
5061 -> 10.0.0.2 and 8011-8020 -> 10.0.0.2 etc etc.

You need to turn on NAT support on each phone.

What you are doing here is allowing each and every phone to work in its own
right across the NAT gateway. After you have finished. Each and every phone
should be able to make and receive calls from anywhere on the internet
(without going through Asterisk).

At this point, if you sacrifice a few chickens and a walrus you may get it
all to work.

Finally:

I have a suspicion that using a NAT router will prevent me from
eliminating Asterisk from the media path.  I am currently running a
Linksys WRT54G with Talisman to get QOS.  Any recommendations for an
alternate QOS router?  Ideally it will also support multiple
sub-domains...

You can do all sorts of stuff with your WRT54G. Running openser on your
WRT54G could in theory do what your looking for. There are plenty of WRT54G
firmwares that let you do nifty VoIP things. You can even install asterisk
on your WRT54G. Check out www.openwrt.org

Hope this is some help.

PS: I found a bug in asterisk's re-invite code that in some cases makes
asterisk push out an invalid SIP packet. If you see anything like this, let
me know and I can send you the patch.

On 1/24/06, David Thomas <punknow at gmail.com> wrote:
> That is the way way SER works. I too am very interested to know if
> this can be done with Asterisk.
>
> David
>
> On 1/12/06, Pavel Jezek <pavel.jezek at i.cz> wrote:
> > Hi, I have asterisk on public IP and phones in two locations behind
> > firewall/nat,
> > - when I have nat=yes and canreinvite=no, this is working fine, but rtp
> > stream must go _always_ through asterisk, even if phones talk inside
> > their locations
> > - when I have nat=yes and canreinvite=yes, phones can speak only inside
> > their location and rtp stream is connected directly between phones (this
> > is, imho, correct and logical), but,
> > is possible to combine both, so do reinvite only "within" e.g. one
> > context and disable reinvite when connecting phones between two context,
> > or any better option exist/planned how to solve?
> > thanks
> > PJ
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