13 Feb
2008
13 Feb
'08
7:26 p.m.
Andy,
I would say both methods are having disadvantages and advantages.
1. The mediaproxy timeout is a plus if this turns to be stable . I had some
not so good experiences in the past and not really responsive support for my
issues, so I have dropped the idea. I will need to recheck, perhaps the
issues were solved.
2. Yate has no rtp detection, therefore will not detect your dead sessions.
I preferred to use it due to prepaid stuff and automatic header masking
features I told you about.
Accounting issues were discussed and rediscussed over and over on this list,
so I will not pop up the subject again.
I think the best accounting technique would be still the last device which
is in touch with your carrier which charges you, so if you send it to PSTN,
then I would say use accounting provided by your PSTN gateway.
Cheers,
DanB
On Feb 13, 2008 6:06 PM, Andy Smith <a.smith(a)ukgrid.net> wrote:
Hi Dan,
one other query on the below, regarding Yate providing more accuarate
accounting, if OpenSER is used with mediaproxy will this not provide the
same level of accuracy (as Mediaproxy actually sits in the RTP stream)?
thanks Andy.
----- Original Message -----
*From:* Dan-Cristian Bogos <danb.lists(a)googlemail.com>
*To:* A.smith <a.smith(a)ukgrid.net>
*Cc:* users(a)lists.openser.org
*Sent:* Wednesday, February 13, 2008 1:07 PM
*Subject:* Re: [OpenSER-Users] FreeRADIUS-CDRTool Prepaid Connector 1.1Released
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(a)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.
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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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