Andy,<br><br>I would say both methods are having disadvantages and advantages. <br><br>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.<br>
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.<br>Accounting issues were discussed and rediscussed over and over on this list, so I will not pop up the subject again.<br>
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.<br><br>
Cheers,<br>DanB<br><br><br><div class="gmail_quote">On Feb 13, 2008 6:06 PM, Andy Smith <<a href="mailto:a.smith@ukgrid.net" target="_blank">a.smith@ukgrid.net</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff">
<div><font face="Arial" size="2">Hi Dan,</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2"> 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)?</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2"> thanks
Andy.</font></div>
<blockquote style="border-left: 2px solid rgb(0, 0, 0); padding-right: 0px; padding-left: 5px; margin-left: 5px; margin-right: 0px;"><div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">----- Original Message ----- </div>
<div style="background: rgb(228, 228, 228) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">
<b>From:</b>
<a title="danb.lists@googlemail.com" href="mailto:danb.lists@googlemail.com" target="_blank">Dan-Cristian Bogos</a> </div></div><div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>To:</b> <a title="a.smith@ukgrid.net" href="mailto:a.smith@ukgrid.net" target="_blank">A.smith</a> </div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Cc:</b> <a title="users@lists.openser.org" href="mailto:users@lists.openser.org" target="_blank">users@lists.openser.org</a> </div>
</div><div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Sent:</b> Wednesday, February 13, 2008 1:07
PM</div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Subject:</b> Re: [OpenSER-Users]
FreeRADIUS-CDRTool Prepaid Connector 1.1 Released</div>
<div><br></div></div><div><div></div><div>Hi Andy,<br><br>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.<br><br>I would still
recommend using Yate combined with OpenSER in the case you are doing some sort
of "Carrier business", for the following reasons:<br>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). <br>2. You have more protocols available in.<br>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.<br>4. Yate can work in rtp_forward mode,
therefore no extra overhead given by rtp processing.<br><br>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.<br>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.<br><br>Let me know if you need further
info.<br><br>Cheers,<br>DanB<br><br><br><br><br><br>
<div class="gmail_quote">On Feb 13, 2008 12:53 PM, A.smith <<a href="mailto:a.smith@ukgrid.net" target="_blank">a.smith@ukgrid.net</a>> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi
Dan/List,<br><br> I was reading the post below and trying to understand
how your config<br>works. If<br>you are implementing this with something
like a Cisco PSTN then you need all<br>of<br>these: PSTN, OpenSER,
Mediaproxy and Yate involved in the SIP route? Does<br>the RTP<br>stream
have to route via Yate and mediaproxy? :S<br><br>thanks for any help! cheers
Andy.<br><br>>Hey Marc,<br>><br>>I use Yate for doing that. It is
simple and works out of the box (with<br>adding few<br>>lines in configs
of course).<br>><br>>I take Session timeout returned from connector
and pass it to yate in a sip<br>header<br>>Process that header in regex
routing and define the value as timeout for<br>session.<br>>Yate knows by
default that when a session has a parameter
"timeout"<br>returned<br>>from routing to disconnect the call when
timeout is hit.<br>><br>>Let me know if you need further info, so I
can send you some config files<br>if you<br>>want to. You can contact me
on IRC for live support (DanB).<br>><br>><br>>All the
best,<br>>DanB<br><br>________________________________________________<br>Message
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