[Asterisk-Users] SIP reinvite asterisk and NAT

[BJ Weschke]

 IAX will use less than individual SIP calls when trunked, yes. I'm not sure 
it's a significant savings with the number of streams we may be talking 
about for your particular scenario, but for larger carrier trunking 
scenarios, it could be quite significant.
  On your original question, if you want to do "canreinvite=yes" for the 
phones, you will most certainly need to make sure that those phones are on 
static IPs and aren't NAT'd from each other or the * server. This raises new 
security questions as well along the lines of making sure that only the IPs 
you're expecting to get traffic from to those IPs are actually able to reach 
those IPs, but you might solve this problem by using internal IP addressing 
if you've already got a VPN link established between the site that has * now 
and the remote office where you're looking to bring up these phones. UDP 
based VPNs are better than TCP based VPNs for VoIP within VPN from a latency 
perspective, but I've deployed plenty of VPNs that weren't UDP based that 
were able to sponsor VoIP traffic through them without any problem.
 On 9/16/05, Jason Walker <desktophero at gmail.com> wrote: 
> 
> I am curious...are you saying to use SIP locally and IAX from point to 
> point
> (over a WAN or VPN tunnel)? With that in mind, do you think that using a
> lesser compressed codec over the IAX trunk would give an okay amount of
> bandwidth savings?
> 
> Thanks.
> 
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mark 
> Phillips
> Sent: Thursday, September 15, 2005 7:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SIP reinvite asterisk and NAT
> 
> If these phones are all to be in a single location I'd deploy a remote
> Asterisk box and run an IAX trunk between remote and local sites.
> That'll save more bandwidth than having a potential 5 individual SIP
> sessions running over your link.
> 
> Also, with the addition of an analogue card such as the TDM400 series 
> you'll
> have survivability should your link go down.
> 
> If you don't add a phone line to the remote site how will they be able to
> call 911 etc?
> 
> Mark
> 
> Damon Estep wrote:
> > I would like to setup up a remote office with a half dozen or so SIP
> > phones connected to an asterisk server via a WAN link. To conserve
> > bandwidth I would like the phones to be able to re-invite when they
> > call each other.
> >
> >
> >
> > The phones will be Polycom, Cisco, or Snom.
> >
> >
> >
> > I may or may not use NAT. Seems like the NAT would really mess up
> > re-invites, any experience with that?
> >
> >
> >
> > Assuming no NAT, what should be expected in this setup?
> >
> >
> >
> > I know the transfer option in asterisk would not work, but I do not
> > think that is a big deal since any re-invited calls would be user to
> > user, with little or no need to transfer.
> >
> >
> >
> > As long as the SIP termination peers I am using are set to
> > canreinvite=no then a call between the users and a remote party would
> > not be re-invited, since the peer terminating the call is set to no,
> > correct?
> >
> >
> >
> > Can someone share some experiences wit this type of setup? Are there
> > other real issues to look out for or be aware of?
> >
> >
> >
> > I am really just trying to avoid having another asterisk box in the
> > remote site to maintain, but do not want to waste bandwidth on calls
> > going across the office.
> >
> >
> >
> > Thanks for taking the time to share your wisdom.
> >
> >
> >
> >
> >
> >
> >
> >
> > ----------------------------------------------------------------------
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> >
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> --
> 
> Mark, G7LTT/KC2ENI
> Randolph, NJ
> http://www.g7ltt.com
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