[Asterisk-Users] SIP RTP

[BJ Weschke]

On 1/14/06, Mike Hammett <asterisk-users at ics-il.net> 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?
>

 This isn't correct. While RTP goes away on a successful reinvite,
Asterisk never gets out of the middle of the SIP signaling path
because chan_sip is a B2BUA and not a SIP proxy.

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