On 9/19/06, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
No it won't
The SIP messages control the call, irrespective of where the RTP goes so Asterisk can terminate/set-up calls exactly as if the RTP was being handled.
Personally, I would always handle the RTP on quality/accountability/consitency grounds but, yes, that will incur a bandwidth overhead.
I am thinking if re-invite will interfere accounting.
No it won't
Please help me to figure it out:
Phone A is registered at asterisk and calls a gateway. If the gateway
allows re-invite than the rtp would go directly from phone A to the
gateway, while the sip messages are still going through Asterisk.
Asterisk will be informed when the call ended.
If it is a postpaid accounting, just bill the customer, however, how is
it for a pre-paid (calling card user)?
I think Asterisk will have no power to turn off the call from A to the
gateway.
Even more, if the gateway would allow to end a call and continue with a
new call, the new call would not be billed (or would it)?
The SIP messages control the call, irrespective of where the RTP goes so Asterisk can terminate/set-up calls exactly as if the RTP was being handled.
I guess the solution must be re-invite=no
However, re-invite=no means that each call is going with rtp also
through my server, what means for a remote phone, I have to provide for
both legs the bandwidth.
Personally, I would always handle the RTP on quality/accountability/consitency grounds but, yes, that will incur a bandwidth overhead.
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