[Asterisk-Users] conditional canreinvite

[sdcharly at gmail.com]

Hi,

Thanks for that cool info. It will help me in the days to come

Great going...

Dan


On 25/01/06, hugolivude <hugolivude at gmail.com> wrote:
>
> 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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