[Asterisk-Users] avoiding rtp triangle

[John Todd]

At 6:30 PM -0700 on 5/19/04, Randy Bush wrote:
>so, is it safe to put
>
>    canreinvite=yes
>
>on a 7960?  on a 1750?  on a spa-x000?  an xten?
>how the heck do i find out other than the hard way?
>
>randy
>
>--
>
>ps: pun intended

I believe all of those support native SIP bridging with reinvites. 
However, NAT may make your life more difficult with failures of this 
native bridging, so the answer is "the hard way" and even that 
depends on the NAT behind which your UA is located, if any.  (If you 
don't have any NAT to mess with, then I've had nothing but success 
with allowing reinvites everywhere.)

Small offices (1-10 active RTP sessions at a time) will probably 
suffer no great loss in having all of the audio piped through a 
central Asterisk server (canreinvite=no).  More network-distributed 
systems would be well-served to have RTP flowing from 
endpoint-to-endpoint, but that breeds more difficulties when there is 
no control over the NAT or firewall circumstances.  Additionally, in 
a service provider environment it may be worth the extra 
latency/bandwidth to haul everything back to a central location with 
an RTP proxy in the middle so that quality issues can at least be 
quantified (which leg of the call sucks?  Start banging heads there.)

I really base things on a case-by-case basis.  Doing RTP the "right" 
way is not always necessarily the "right" way for the environment.

JT