[Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

[Olle E. Johansson]

Chris Albertson wrote:
> This is the big problem with using Asterisk for SIP.  With Asterisk
> the audio data between two SIP extensions has to actualy go into
> then out of the Asterisk box.  This does not scale well to
> thousands of users like in a university campus or a comercial
> SIP service.  
> 
http://www.voip-info.org/wiki-Asterisk+sip+reinvite

As I understand this Asterisk sets up the call with itself as endpoints,
then moves the stream tobypass the PBX and go directly with a SIP reinvite.
Some clients does not support this, and with those you have to configure
asterisk to stay in the media path for this client with canreinvite=no in SIP.conf.

/O