[Asterisk-Users] Re: Media Path Optimization & NAT

Rich Adamson radamson at routers.com
Thu Jan 20 05:58:23 MST 2005


> >>Let me restate my problem. I have a group of users behind a constrained 
> >>pipe to the public network. There are a few mobile users that will 
> >>mostly be working from their home offices. I *really* want to avoid 
> >>having a call from a mobile user to a public number cause double the 
> >>traffic on the corporate link. Am I making any kind of sense?
> > 
> > You're making sense, but trying to use the canreinvite=yes is not going
> > to be the answer in my opinion. As stated previously, for that to work
> > as you'd like, the sip provider would need to initiate the reinvite and
> > its certainly not in their best interest to do that (not to mention the
> > time they would consume trying to make it work with unknown nat 
> > functions at your user's multiple locations).
> > 
> > There are lots of other ways to address the issue, but in my opinion
> > each approach will require spending additional funds. You really need
> > to identify the different ways to handle the requirement and the costs
> > associated with each. Don't know of any way around that.
> 
> Sorry to be a bother, but other ways to you see to address the issue? 
> I'm certainly willing to invest time and funds into this, that isn't an 
> issue.
> 
> Is SER really the solution to having greater control over the SIP 
> transactions and their associated RTP streams?

I'm not a SER user, therefore others on this list might have a better
understanding as to its appropriateness.

Other possible approaches:
- two * systems, one of which is colocated outside your corp structure
  with iax link, and a sip client with two proxy registration definitions
  (for internal system, if sip client isn't registered, send call to
  colocated system)
- two sip accounts; one internal and one with a sip provider, sip client
  with two different registrations, dialplan to support both
- second internet pipe at your corp location dedicated to outbound calls
  to your sip provider (iax-gsm across broadband?)
- existing config but use a lower-bandwidth codec and increase the size
  of your broadband pipe to support required bandwidth
- two broadband pipes; one for basic internet use, second dedicated only
  to * (remote sip client registration and calls via sip provider). If
  * configured with registered IP, sip client only needs one registration

Obviously, having a good understanding as to the maximum number of 
simultanous calls (to your sip provider) is needed to size pipes, etc.





More information about the asterisk-users mailing list