[Asterisk-Users] Endpoint-to-Endpoint RTP Packets

[Steven Critchfield]

On Wed, 2003-09-10 at 10:42, Olle E. Johansson wrote:
> Eric Wieling wrote:
> 
> > That would be reinvite= and canreinvite= in the user entry for each SIP
> > endpoint.  Asterisk will allow the endpoints to talk directly to each
> > other if both those settings are = yes (the default, I think) AND both
> > endpoints use the same protocol (SIP) AND the same codec.
> 
> I tried to document this on the Wiki. Did I get it right? Please check!
> http://tinyurl.com/mvny
> 
> (If it's wrong, just change it, it's a Wiki)

Minor nitpick, second paragraph 4th word should probably be "two" not
"to".

Also you may want to point out that there are SIP phones that do not
react well to the reinvites like the ata186.

-- 
Steven Critchfield  <critch at basesys.com>