Re: [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

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