Hi, take a look at my post. Probably you will find a solution for your issue.
http://opensips-open-sip-server.1449251.n2.nabble.com/Rtpproxy-connection-td7581935.html
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On Nov 10, 2012, at 11:35 AM, zhi sun wrote:
hi guys,
does opensips support rtmp? i want to use flash rtmp phone (on web page) to
call opensips (and the FS behind).
if not, what is the correct point as solution?
IOpenSIPS does not handle media. No RTP nor RTMP. You can have OpenSIPS
Hi Denis,The client must send BYE to end the dialog. but in case of timeout you can generate the local BYE to end the dialog:create_dialog("B"); The string "B" activate the BYE on timeout behavior. In OpenSIPS 1.6 use "bye_on_timeout_flag".// BinanFrån: dpa denis7...@mail.ru Till:
Hello,
Your OpenSIPS starts, but you see those errors on the log file, right ?
Those errors are triggered because you have some dialogs in the database
that were ongoing on the old port, and since you switched to a new port,
it doesn't match the new listening sockets anymore. You can safely
Hello,
OpenSIPS does not currently support RTMP and if you want, you must use a
RTMP to SIP gateway.
Regards,
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 11/10/2012 12:35 PM, zhi sun wrote:
hi guys,
does opensips support rtmp? i want to use flash rtmp phone (on web
Hello,
Very nice tutorial, thank you.
Important to note that you can use other backends as well, such as
Cassandra in 1.8, or for the future release 1.9 you'll also be able to
use CouchBase or MongoDB in a similar fashion.
Best Regards,
Vlad Paiu
OpenSIPS Developer
Hi Duane,
Please replace the stop query by:
CALL update_radacct_record( \
'radius', \
'%S', \
'%{Acct-Delay-Time}', \
'', \
'%{X-RTP-Stat}', \
'%{Acct-Session-Id}', \
Hello
I am using “bye_on_timeout_flag” for each transaction in Opensips. The default
timeout – 10 s. “Real timeout” rewriting after loose_routing() when routing ACK
request on 200 ОК and different for clients. The max timeout in Opensips for
dialogs is s.
From:
Good morning.
I've been testing this again (Mediaproxy)
Playing with IPTABLES has not been a good idea because a rule to deny
traffic doesn't fire
/proc/sys/net/ipv4/netfilter/ip_conntrack_udp_timeout_stream rule so I was
mistaken Saul
Hello Community,
I'm currently using opensips + rtpproxy but I'm thinking of switching to
Mediaproxy instead RTPProxy due to performance issue.
In RTPProxy we are using External/Internal interfaces in order to bridge
RTP flows between 2 non contiguous network
I had a look inside Mediaproxy,
Thanks as always Bogdan. I guess the bulk approach should meet my
requirements for now.
Regards,
Nilanjan.
On Sat, Nov 10, 2012 at 1:07 AM, Bogdan-Andrei Iancu bog...@opensips.orgwrote:
**
Hi Nilanjan,
The LB load info is available only via the MI interface, as it is runtime
info and not
Hello.
2012/11/12 Pierre-Yves Marche pierrey.mar...@gmail.com:
Hello Community,
I'm currently using opensips + rtpproxy but I'm thinking of switching to
Mediaproxy instead RTPProxy due to performance issue.
I wonder what kind of performance issues have you encountered?
--
With best regards,
Hi,
We observed issue using RTPProxy under highload conditions 800 ongoing
calls where the jitter is increasing. (CPU usage is high also).
We think that Mediaproxy could be more efficient ( in term of CPU, Jitter)
as it does not forward RTP packet in usermode.
Regards,
Pierre-Yves
On Mon,
Your radius client is not sending it. You should fix the client, not the server.
Adrian
On Nov 12, 2012, at 4:17 PM, Duane Larson wrote:
That fixed it. Is there a reason why the latest download
cdrtool_9.0.0.tar.gz has
/cdrtool_9.0.0/CDRTool/setup/radius/OpenSIPS/sql.conf where it does
Hi,
It has to be removed. Connect-Info is not in the dictionary any more. In
version 1 freeradius skipped the field when it was not set or known,
unlike version 2. So that's why the sql broke, simply because the field
is not supplied.
I will push a patch to remove it from CDRTool as well,
Or you can try directly:
/check_user_blacklist( $fU, $fd ,$rU )/
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 11/12/2012 06:03 PM, Engineer voip wrote:
Hi,
I declared these parameters before an it's good
$avp(ru)=$rU;
$avp(fu)=$fU;
Hi,
Why should I take out these fields? These are separate fields and should
be set by radius and your radius client.
Best regards,
--
Tijmen de Mes
AG Projects
On 11/12/2012 04:35 PM, Duane Larson wrote:
Thanks Tijmen. So the OpenSIPS client shouldn't be sending the
Connect-Info like
Hello all,
I´ve checked on SST module and it does not mention anything about
refreshing the dialog timeout value. We need to handle this through UPDATE
messages.
We are using dialog module:
modparam(dialog, timeout_avp, $avp(session_expires))
When we get the INVITE, we do:
Hi,
I´ve created a module that uses libcurl to send requests to a HTTP server.
The problem is that the time to the function 'curl_easy_perform' be called is
increasing each time the method of the module is called.
I´ve made some profiling, and looks like 'all' opensips' processes are using
After some testing I found out the dialog timeout is correctly refreshed if
I set the avp before calling match_dialog(), but this is a problem, because
we can only refresh the timer if the UPDATE message belongs to a dialog.
Otherwise, we must reply with a 481.
I tried using loose_route() instead
Hi, Mariana!
The dialog timeout avp has effect only if it is set before match_dialog
and loose_route. But only the dialog matched by either loose_route,
either match_dialog, will be updated. If none is found, then nothing
will be changed. Therefore I can't really see a problem here - if the
Hi
I am running Opensips on Debian 6. I've got it installed and all is
well--except that it won't start at boot because it requires MySQL to run,
and for some reason, it is being launched BEFORE MySQLd during init.
What I'd like to do is modify the boot order of my process
Deciding that a session has timed out based on a single stream leg being
stopped is not a clever thing to do as one party may simply stop streaming for
legitimate reasons like voice audio detection or muting the input for listening
in into a conference depending on codec behavior. As mediaproxy
Hello everybody, i red the post and didn't work.
Here is my command line to run rtpproxy and my configuration file of
OPENSIPS, if some one could help me.
./rtpproxy -l PUBLICIP/192.168.1.220 -s udp:192.168.1.220:12333 -n tcp:
192.168.1.220:12333 -u user -m 35000 -M 35200
#
# $Id:
Please, post entire opensips log ( set debug to 6 ).
Why are you using same port for sock and notify_sock ??? Have you tried with
different ports?
Bye
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