[Asterisk-Users] SIP RTP

[Peder at NetworkOblivion]

It just re-directs the RTP stream.  The SIP stream still goes through *.


Mike Hammett wrote:
> According to this page:  http://www.asterisk.org/doxygen/Config_sip.html
>  
> canreinvite=yes redirects just the RTP.  I was under the impression that 
> the entire SIP connection got redirected, therefore losing accounting 
> ability.  Could someone clarify this?
>  
> --Mike
> 
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