Hi Andy,
The original config was built with Yate in mind
due to openser incapacity (before release 1.3) of disconnecting the calls.
Since 1.3.0 the dialog module should be able to timeout the calls, in theory
you should no longer need extra software like Yate.
I would still
recommend using Yate combined with OpenSER in the case you are doing some sort
of "Carrier business", for the following reasons:
1. It creates two
different legs for your call (in and out) same as Cisco does, and hides one
side from the other (eg. removes the via headers and any revealing ip
information inside SDP - so your partners should not know where the traffic
comes from).
2. You have more protocols available in.
3. Accounting it
is bit more accurate (you have the session total duration inside the
accounting packets), so radius will be no longer responsible of calculating
the session durations from timestaps.
4. Yate can work in rtp_forward mode,
therefore no extra overhead given by rtp processing.
So basically what
the connector does (as specified in the documentation), for each call which is
authorized by radius, the connector will ask permission from cdrtool. If
permission is granted, it will return in a avp to openser the maximum duration
allowed for the call (timeout value) plus credit available, for the case of
special uas able to display that.
By receiving accounting stop packet, the
connector will inform cdrtool about call disconnection therefore clearing the
lock and debiting the balance inside cdrtool. The rtp stream has nothing to do
with this scenario, so you don't need to touch your NAT support configuration,
it's all in the signaling.
Let me know if you need further
info.
Cheers,
DanB
On Feb 13, 2008 12:53 PM, A.smith <
a.smith@ukgrid.net> wrote:
Hi
Dan/List,
I was reading the post below and trying to understand
how your config
works. If
you are implementing this with something
like a Cisco PSTN then you need all
of
these: PSTN, OpenSER,
Mediaproxy and Yate involved in the SIP route? Does
the RTP
stream
have to route via Yate and mediaproxy? :S
thanks for any help! cheers
Andy.
>Hey Marc,
>
>I use Yate for doing that. It is
simple and works out of the box (with
adding few
>lines in configs
of course).
>
>I take Session timeout returned from connector
and pass it to yate in a sip
header
>Process that header in regex
routing and define the value as timeout for
session.
>Yate knows by
default that when a session has a parameter
"timeout"
returned
>from routing to disconnect the call when
timeout is hit.
>
>Let me know if you need further info, so I
can send you some config files
if you
>want to. You can contact me
on IRC for live support (DanB).
>
>
>All the
best,
>DanB
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