9 Dec
2006
9 Dec
'06
6:13 p.m.
Hi,
I am currently testing a system which connects two "islands" ... each
island has its own SER (0.9.6) + rtp proxy ...
Actually there is NOT going to be NAT or firewall among them, but i
still do want to proxy all RTP streams (sorry ... requirements).
Probably even within the phones in the same island ...
I have a few questions ...
- i saw mentioned that chaining is possible ... no problem there
right? i need to send an extra parameter to the force_rtp_proxy and
that is it? no side-effects if, i.e, call between the phones in the
same island (thus, just one rtp proxy)?
- What would be the bare minimum config to force rtpproxy, without all
the NAT tests? as said, i have no such problem, so i would like a
minimal config: fix_nated_sdp and force_rtp_proxy? is that enough?
- For testing purposes in my little lab ... can i run two SERs on the
same box (that i know i can do :D ) ... which use the same rtpproxy on
the box, thus via the same unix socket? just for testing ... they put
me in a corner of an office with just a little table :)
Regards,
Cesc
9 Dec
9 Dec
7:30 p.m.
Hi Cesc,
Thanks for VON.
I have a few questions ...
- i saw mentioned that chaining is possible ... no problem there
right? i need to send an extra parameter to the force_rtp_proxy and
that is it? no side-effects if, i.e, call between the phones in the
same island (thus, just one rtp proxy)?
This is possible, but you need to turn of the checking :) (there is a
modparam that checkes if there in the sdp sasys a=nortpproxy or
something like that.
- What would be the bare minimum config to force rtpproxy, without all
the NAT tests? as said, i have no such problem, so i would like a
minimal config: fix_nated_sdp and force_rtp_proxy? is that enough?
route[]{
record_route();
if(method=="INVITE"){
force_rtp_proxy();
t_on_reply("1");
};
rewritehostport("bar.com:5060");
t_relay();
};
on_replyroute[1]{
if(status=="200"){
force_rtp_proxy();
};
};
I think this would be enough ;)
- For testing purposes in my little lab ... can i run two SERs on the
same box (that i know i can do :D ) ... which use the same rtpproxy on
the box, thus via the same unix socket? just for testing ... they put
me in a corner of an office with just a little table :)
I would use vmware for this, then you can have 2 "physical" machines on
your laptop ;) and you really need this since you want to send rtp
between 2 ip's..
- Atle
Regards,
Cesc
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
7:46 p.m.
On 12/9/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote:
Ok ... i think is a parameter in force_rtp_proxy, right?
in this ser.cfg ... the force_rtp_proxy() call will automatically
change whatever is needed in the SDP, won't it? so no nat stuff
checking functions ... cool :)
Hi Cesc,
Thanks for VON.
It was my pleasure to meet all you guys ... I am hooked now. In my
mind I have a background process trying to figure out how to go to the
next one (San Diego? :D )
I have a few questions ...
- i saw mentioned that chaining is possible ... no problem there
right? i need to send an extra parameter to the force_rtp_proxy and
that is it? no side-effects if, i.e, call between the phones in the
same island (thus, just one rtp proxy)?
This is possible, but you need to turn of the checking :) (there is a
modparam that checkes if there in the sdp sasys a=nortpproxy or
something like that.
- What would be the bare minimum config to force rtpproxy, without all
the NAT tests? as said, i have no such problem, so i would like a
minimal config: fix_nated_sdp and force_rtp_proxy? is that enough?
route[]{
record_route();
if(method=="INVITE"){
force_rtp_proxy();
t_on_reply("1");
};
rewritehostport("bar.com:5060");
t_relay();
};
on_replyroute[1]{
if(status=="200"){
force_rtp_proxy();
};
};
I think this would be enough ;)
I think i will do so ... I deleted vmware a while ago in favour of
having colinux on my windows laptop ... but i cannot run various
colinuxes (i think ... ) on the same box ...
A final question ... basically thinking out loud (and writing it down) :)
I read that rtpproxy won't start relaying until it got an rtp packet
from both sides ... is it true? could this not cause problems,
specially with chained rtpproxies, if say, i have one of the phones
not sending rtp packets (say, it starts muted ... muted means no rtp
packets)?
Cesc
- For testing purposes in my little lab ... can i run two SERs on the
same box (that i know i can do :D ) ... which use the same rtpproxy on
the box, thus via the same unix socket? just for testing ... they put
me in a corner of an office with just a little table :)
I would use vmware for this, then you can have 2 "physical" machines on
your laptop ;) and you really need this since you want to send rtp
between 2 ip's..
- Atle
Regards,
Cesc
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
11 Dec
11 Dec
11:08 a.m.
Cesc wrote:
On 12/9/06, Atle Samuelsen <clona(a)cyberhouse.no>
wrote:
Ok ... i think is a parameter in force_rtp_proxy, right?
yes, you need the "f" flag (guess it is still the same in ser)
http://www.openser.org/docs/modules/1.1.x/nathelper#AEN275
Hi Cesc,
Thanks for VON.
It was my pleasure to meet all you guys ... I am hooked now. In my
mind I have a background process trying to figure out how to go to the
next one (San Diego? :D )
I have a few questions ...
- i saw mentioned that chaining is possible ... no problem there
right? i need to send an extra parameter to the force_rtp_proxy and
that is it? no side-effects if, i.e, call between the phones in the
same island (thus, just one rtp proxy)?
This is possible, but you need to turn of the checking :) (there is a
modparam that checkes if there in the sdp sasys a=nortpproxy or
something like that.
A final question ... basically thinking out loud (and
writing it down) :)
I read that rtpproxy won't start relaying until it got an rtp packet
from both sides ... is it true? could this not cause problems,
specially with chained rtpproxies, if say, i have one of the phones
not sending rtp packets (say, it starts muted ... muted means no rtp
packets)?
AFAIR rtpproxy is asynchronous until the first RTP packet from each side
is received. Thus chaining should work.
regards
klaus
btw: rtpproxy also has some parameters:
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/rtpproxy/manpage.xml?rev=1.2&…
also main.c wil show you that there are some more undocumented parameters.
Cesc
--
Klaus Darilion
nic.at
- Atle
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers Regards,
Cesc
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
11:33 a.m.
Hi!
Thanks guys!
I got it to work yesterday morning ... I have audio going through a
chain of 2 rtpproxies ...
My first attempts failed ... i mistook the F parameter by the R one ... :(
I basically do as Atle showed on his email ... plus some
unforce_rtp_proxy here and there (i found it in some onsip.org config
file).
Cesc
On 12/11/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
Cesc wrote:
On 12/9/06, Atle Samuelsen
<clona(a)cyberhouse.no> wrote:
Ok ... i think is a parameter in force_rtp_proxy, right?
yes, you need the "f" flag (guess it is still the same in ser)
http://www.openser.org/docs/modules/1.1.x/nathelper#AEN275
Hi Cesc,
Thanks for VON.
It was my pleasure to meet all you guys ... I am hooked now. In my
mind I have a background process trying to figure out how to go to the
next one (San Diego? :D )
I have a few questions ...
- i saw mentioned that chaining is possible ... no problem there
right? i need to send an extra parameter to the force_rtp_proxy and
that is it? no side-effects if, i.e, call between the phones in the
same island (thus, just one rtp proxy)?
This is possible, but you need to turn of the checking :) (there is a
modparam that checkes if there in the sdp sasys a=nortpproxy or
something like that.
A final question ... basically thinking out loud
(and writing it down) :)
I read that rtpproxy won't start relaying until it got an rtp packet
from both sides ... is it true? could this not cause problems,
specially with chained rtpproxies, if say, i have one of the phones
not sending rtp packets (say, it starts muted ... muted means no rtp
packets)?
AFAIR rtpproxy is asynchronous until the first RTP packet from each side
is received. Thus chaining should work.
regards
klaus
btw: rtpproxy also has some parameters:
http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/rtpproxy/manpage.xml?rev=1.2&…
also main.c wil show you that there are some more undocumented parameters.
Cesc
--
Klaus Darilion
nic.at
- Atle
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers Regards,
Cesc
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
11:56 a.m.
Hi again Cesc,
* Cesc <cesc.santa(a)gmail.com> [061211 10:33]:
Hi!
Thanks guys!
I got it to work yesterday morning ... I have audio going through a
chain of 2 rtpproxies ...
My first attempts failed ... i mistook the F parameter by the R one ... :(
I basically do as Atle showed on his email ... plus some
unforce_rtp_proxy here and there (i found it in some onsip.org config
file).
Great that it works :) Hope you get your service working as it should.
- Atle
Cesc
On 12/11/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
>Cesc wrote:
>> On 12/9/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote:
>>> Hi Cesc,
>>>
>>> Thanks for VON.
>>
>> It was my pleasure to meet all you guys ... I am hooked now. In my
>> mind I have a background process trying to figure out how to go to the
>> next one (San Diego? :D )
>>
>>>
>>> > I have a few questions ...
>>> > - i saw mentioned that chaining is possible ... no problem there
>>> > right? i need to send an extra parameter to the force_rtp_proxy and
>>> > that is it? no side-effects if, i.e, call between the phones in the
>>> > same island (thus, just one rtp proxy)?
>>>
>>> This is possible, but you need to turn of the checking :) (there is a
>>> modparam that checkes if there in the sdp sasys a=nortpproxy or
>>> something like that.
>>>
>>
>> Ok ... i think is a parameter in force_rtp_proxy, right?
>
>yes, you need the "f" flag (guess it is still the same in ser)
>
>http://www.openser.org/docs/modules/1.1.x/nathelper#AEN275
>
>> A final question ... basically thinking out loud (and writing it down) :)
>> I read that rtpproxy won't start relaying until it got an rtp packet
>> from both sides ... is it true? could this not cause problems,
>> specially with chained rtpproxies, if say, i have one of the phones
>> not sending rtp packets (say, it starts muted ... muted means no rtp
>> packets)?
>
>
>AFAIR rtpproxy is asynchronous until the first RTP packet from each side
>is received. Thus chaining should work.
>
>regards
>klaus
>
>btw: rtpproxy also has some parameters:
>http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/rtpproxy/manpage.xml?rev=1.2&content-type=text/vnd.viewcvs-markup
>
>also main.c wil show you that there are some more undocumented parameters.
>
>
>>
>>
>> Cesc
>>
>>
>>>
>>> - Atle
>>>
>>> > Regards,
>>> >
>>> > Cesc
>>> > _______________________________________________
>>> > Serusers mailing list
>>> > Serusers(a)lists.iptel.org
>>> > http://lists.iptel.org/mailman/listinfo/serusers
>>>
>> _______________________________________________
>> Serusers mailing list
>> Serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>--
>Klaus Darilion
>nic.at
>
>
1:14 p.m.
Hi there,
Yeah ... well ... this was the easy step :)
Now i am to integrate it with an RSVP daemon that we run on the
machines ... don't ask me why, rsvp sucks ... but i need it for some
project of mine :)
Cesc
On 12/11/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote:
Hi again Cesc,
* Cesc <cesc.santa(a)gmail.com> [061211 10:33]:
Hi!
Thanks guys!
I got it to work yesterday morning ... I have audio going through a
chain of 2 rtpproxies ...
My first attempts failed ... i mistook the F parameter by the R one ... :(
I basically do as Atle showed on his email ... plus some
unforce_rtp_proxy here and there (i found it in some onsip.org config
file).
Great that it works :) Hope you get your service working as it should.
- Atle
Cesc
On 12/11/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
>Cesc wrote:
>> On 12/9/06, Atle Samuelsen <clona(a)cyberhouse.no> wrote:
>>> Hi Cesc,
>>>
>>> Thanks for VON.
>>
>> It was my pleasure to meet all you guys ... I am hooked now. In my
>> mind I have a background process trying to figure out how to go to the
>> next one (San Diego? :D )
>>
>>>
>>> > I have a few questions ...
>>> > - i saw mentioned that chaining is possible ... no problem there
>>> > right? i need to send an extra parameter to the force_rtp_proxy and
>>> > that is it? no side-effects if, i.e, call between the phones in the
>>> > same island (thus, just one rtp proxy)?
>>>
>>> This is possible, but you need to turn of the checking :) (there is a
>>> modparam that checkes if there in the sdp sasys a=nortpproxy or
>>> something like that.
>>>
>>
>> Ok ... i think is a parameter in force_rtp_proxy, right?
>
>yes, you need the "f" flag (guess it is still the same in ser)
>
>http://www.openser.org/docs/modules/1.1.x/nathelper#AEN275
>
>> A final question ... basically thinking out loud (and writing it down) :)
>> I read that rtpproxy won't start relaying until it got an rtp packet
>> from both sides ... is it true? could this not cause problems,
>> specially with chained rtpproxies, if say, i have one of the phones
>> not sending rtp packets (say, it starts muted ... muted means no rtp
>> packets)?
>
>
>AFAIR rtpproxy is asynchronous until the first RTP packet from each side
>is received. Thus chaining should work.
>
>regards
>klaus
>
>btw: rtpproxy also has some parameters:
>http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/rtpproxy/manpage.xml?rev=1.2&content-type=text/vnd.viewcvs-markup
>
>also main.c wil show you that there are some more undocumented parameters.
>
>
>>
>>
>> Cesc
>>
>>
>>>
>>> - Atle
>>>
>>> > Regards,
>>> >
>>> > Cesc
>>> > _______________________________________________
>>> > Serusers mailing list
>>> > Serusers(a)lists.iptel.org
>>> > http://lists.iptel.org/mailman/listinfo/serusers
>>>
>> _______________________________________________
>> Serusers mailing list
>> Serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>--
>Klaus Darilion
>nic.at
>
>
2:28 p.m.
New subject: [Serusers] any kind of statistics available
Hello,
is there any kind of statistic-support available for OpenSER?
I would like to have statistics about number of calls, time
of calls, active users, ...
Is there any tool available, which creates statistics. Pictures
and diagrams would be nice to have.
best regrads
Jörg
2:53 p.m.
New subject: [Users] Re: any kind of statistics available
Hello,
On 12/11/06 14:28, Wienand, Joerg wrote:
Hello,
is there any kind of statistic-support available for OpenSER?
I would like to have statistics about number of calls,
these kind of statistics are
exported by dialog module:
http://www.openser.org/docs/modules/1.2.x/dialog.html#AEN151
time
of calls, active users, ...
active (registered) users statistics are exported by usrloc module.
openserctl fifo get_statistics all - should return you all statistics
available in openser at that moment. This require FIFO interface to be
enabled. In development version you can get the statistics via XMLRPC as
well (see the new modules mi_xmlrpc and mi_fifo).
Is there any tool available, which creates statistics.
Pictures
and diagrams would be nice to have.
You can make a small script to pull periodically statistics via
FiFO/XMLRPC into a database/file and from there generate graphics at
your convenience.
Cheers,
Daniel
best regrads
Jörg
6560
days inactive
6562
days old
8 comments
5 participants
participants (5)
-
Atle Samuelsen -
Cesc -
Daniel-Constantin Mierla -
Klaus Darilion -
Wienand, Joerg