2 Nov
2007
2 Nov
'07
5:23 p.m.
If you've stopped using MediaProxy, what do you use now? I'm currently
seeking an alternative to MediaProxy, and would appreciate some ideas. All
I need is NAT traversal - but something does doesn't route media (voice)
through the OpenSER server...
Thanks,
Brian
----- Original Message -----
From: "Jeremy McNamara" <jj(a)nufone.net>
To: "Mike O'Connor" <mike(a)pineview.net>
Cc: <users(a)lists.openser.org>
Sent: Friday, November 02, 2007 7:30 AM
Subject: Re: [OpenSER-Users] MediaProxy 1.9.0 - Radius
Mike O'Connor wrote:
Hi Guys
I really think there is something wrong with MediaProxy, I can not get
the radius or the mysql accounting to report any details, as best as I
can see there is never any attempt to even try.
I totally scratched MediaProxy off of the list when the author of the
code told me to take my questions to the openser mailing list.
Good luck finding support - I even offered to pay and was completely
ignored.
My suggestion would be to use Asterisk as your media proxy and use one
of the various radius connectors (if you absolutely require radius)
Jeremy
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users 
2 Nov
2 Nov
6:17 p.m.
New subject: [OpenSER-Users] MediaProxy 1.9.0 - Radius
Hi Brian,
The other alternative is rtpproxy. Check the nathelper module.
It can control multiple rtpproxy servers hosted on several machines:
http://www.openser.org/docs/modules/1.3.x/nathelper.html#AEN160
This feature will be available on the next 1.3 release (scheduled by
the end of the year)
Regards,
Ovidiu Sas
On 11/2/07, Brian Heath <brian(a)telethra.com> wrote:
If you've stopped using MediaProxy, what do
you use now? I'm currently
seeking an alternative to MediaProxy, and would appreciate some ideas. All
I need is NAT traversal - but something does doesn't route media (voice)
through the OpenSER server...
Thanks,
Brian
----- Original Message -----
From: "Jeremy McNamara" <jj(a)nufone.net>
To: "Mike O'Connor" <mike(a)pineview.net>
Cc: <users(a)lists.openser.org>
Sent: Friday, November 02, 2007 7:30 AM
Subject: Re: [OpenSER-Users] MediaProxy 1.9.0 - Radius
Mike O'Connor wrote:
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
Hi Guys
I really think there is something wrong with MediaProxy, I can not get
the radius or the mysql accounting to report any details, as best as I
can see there is never any attempt to even try.
I totally scratched MediaProxy off of the list when the author of the
code told me to take my questions to the openser mailing list.
Good luck finding support - I even offered to pay and was completely
ignored.
My suggestion would be to use Asterisk as your media proxy and use one
of the various radius connectors (if you absolutely require radius)
Jeremy
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users 
10:08 p.m.
New subject: [OpenSER-Users] MediaProxy 1.9.0 - Radius
Brian Heath wrote:
If you've stopped using MediaProxy, what do
you use now? I'm currently
seeking an alternative to MediaProxy, and would appreciate some ideas. All
I need is NAT traversal - but something does doesn't route media (voice)
through the OpenSER server...
As I suggested to Mike, use Asterisk. You get full transcoding
support, proper call accounting and the ability to do calling
applications (ivr, etc) , if you want.
You can fix the NAT problems in the SDP from within openser.cfg.
Jeremy
3 Nov
3 Nov
2:54 a.m.
New subject: [OpenSER-Users] MediaProxy 1.9.0 - Radius
> If you've stopped using MediaProxy, what
do you use now? I'm
currently
> seeking an alternative to MediaProxy, and would
appreciate some ideas.
All
I need is NAT
traversal - but something does doesn't route media (voice)
through the OpenSER server...
As I suggested to Mike, use Asterisk. You get full transcoding
support, proper call accounting and the ability to do calling
applications (ivr, etc) , if you want.
4 Nov
4 Nov
8:15 p.m.
New subject: [OpenSER-Users] MediaProxy 1.9.0 - Radius
CSB wrote:
Can you elaborate a little on "proper call
accounting" as I am battling with
this currently. An example:
SER sends INVITE to Asterisk. Depending on the circumstances we might want
to record a voicemail message, hit an IVR or queue, or pass the call on to
the PSTN.
Since OpenSER only manages the SIP sessions, OpenSER does not
specifically know if any particular call is active or not. If you were
only relying OpenSER's accounting a crafty voip user could simply never
send you a BYE message, thus effectively never hanging up any calls.
This is why mediaproxy has a call accounting process. I believe RTP
Proxy has also been updated or is scheduled to be updated with an
accounting process.
In my opinion, the only truly viable way to do call accounting is along
with the media stream (RTP), then you will know if the call is still in
session and have the ability to specifically terminate the call for any
given reason (by stopping the RTP from flowing.)
Since Asterisk can't deal with OpenSER's
authentication limitations we can
only have one effective SIP peer (based on the IP of OpenSER) and therefore
one context for accounting purposes. This makes even routing the call a
challenge (how do you make sure that only certain users can get out to the
PSTN whereas others stay internal to Asterisk). How do you get usable
accounting records? If there are 5 calls from different users being passed
to Asterisk they will all be accounted in the same way and it is not
possible to bill them separately. If anyone has any advice on what I'm
missing or how to get useful accounting records in Asterisk I would
appreciate it.
One option might be to send custom SIP header(s) to Asterisk. I wrote
my own CDR module for Asterisk, to deal with my own particular environment.
Jeremy McNamara
9:51 p.m.
New subject: [OpenSER-Users] MediaProxy 1.9.0 - Radius
On Nov 4, 2007, at 5:15 PM, Jeremy McNamara wrote:
CSB wrote:
> Can you elaborate a little on "proper call accounting" as I am
> battling with
> this currently. An example:
> SER sends INVITE to Asterisk. Depending on the circumstances we
> might want
> to record a voicemail message, hit an IVR or queue, or pass the
> call on to
> the PSTN.
Since OpenSER only manages the SIP sessions, OpenSER
does not
specifically know if any particular call is active or not. If you
were
only relying OpenSER's accounting a crafty voip user could simply
never
send you a BYE message, thus effectively never hanging up any calls.
This is why mediaproxy has a call accounting process. I believe RTP
Proxy has also been updated or is scheduled to be updated with an
accounting process.
In my opinion, the only truly viable way to do call accounting is
along
with the media stream (RTP), then you will know if the call is still
in
session and have the ability to specifically terminate the call for
any
given reason (by stopping the RTP from flowing.)
What about the scenario where the call is placed on hold and the BYE
isn't properly generated? We actually ran into this issue last week
where it appeared that an on hold call was active for more than 5
hours (yes not impossible, but in this case the call "ended" after
only a few seconds. Since the call was on hold the RTP wasn't taken
down when no audio was not passed since none was expected. Really it
seems the only way to do this properly is not with RTP but with SIP
session timers but unfortunately not many devices have them implemented.
I guess the question is there really any other options?
> Since Asterisk can't deal with OpenSER's authentication limitations
> we can
> only have one effective SIP peer (based on the IP of OpenSER) and
> therefore
> one context for accounting purposes. This makes even routing the
> call a
> challenge (how do you make sure that only certain users can get out
> to the
> PSTN whereas others stay internal to Asterisk). How do you get usable
> accounting records? If there are 5 calls from different users being
> passed
> to Asterisk they will all be accounted in the same way and it is not
> possible to bill them separately. If anyone has any advice on what
> I'm
> missing or how to get useful accounting records in Asterisk I would
> appreciate it.
We are working on a generic CDR module which will aggregate CDR
records for both Asterisk and openSER. It only accounts for calls
based on source and destination number, so if you need something
fancier than that (say path) then it won't work for you (as of yet,
path basis is coming) We'll be releasing it open source when it is
completed and we'll post to the list then.
Thanks
Miles
5 Nov
5 Nov
3:45 a.m.
New subject: [OpenSER-Users] MediaProxy 1.9.0 - Radius
Miles Scruggs writes:
I guess the question is there really any other
options?
yes, the internet way, i.e., don't try to charge per minute for
sip-to-sip calls and use accounting records for unreliable "information
only" purpose. calls that end up to/from pstn, you can reliable charge
at the gateway where both signaling and media is terminated.
-- juha
11:06 a.m.
New subject: [OpenSER-Users] MediaProxy 1.9.0 - Radius
On Nov 5, 2007, at 12:45 AM, Juha Heinanen wrote:
Miles Scruggs writes:
That works in theory, but still doesn't solve the case for network
failure of the end user when the end user was put on hold by an
upstream party. The carrier has no way to know the call ended (until
the call is taken off hold upstream) as they aren't expecting any
audio to be passed and we haven't sent a proper bye.
I think Ovidiu has the right solution, we'll see how we can implement
this.
Thanks
Miles
I guess the question is there really any other
options?
yes, the internet way, i.e., don't try to charge per minute for
sip-to-sip calls and use accounting records for unreliable
"information
only" purpose. calls that end up to/from pstn, you can reliable
charge
at the gateway where both signaling and media is terminated.
6280
days inactive
6283
days old
7 comments
6 participants
participants (6)
-
Brian Heath -
CSB -
Jeremy McNamara -
jh＠tutpro.com -
Miles Scruggs -
Ovidiu Sas