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
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@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
On 12/9/06, Atle Samuelsen clona@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?
- 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?
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 :)
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..
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
- Atle
Regards,
Cesc _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Cesc wrote:
On 12/9/06, Atle Samuelsen clona@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&a...
also main.c wil show you that there are some more undocumented parameters.
Cesc
- Atle
Regards,
Cesc _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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@pernau.at wrote:
Cesc wrote:
On 12/9/06, Atle Samuelsen clona@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&a...
also main.c wil show you that there are some more undocumented parameters.
Cesc
- Atle
Regards,
Cesc _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Klaus Darilion nic.at
Hi again Cesc,
* Cesc cesc.santa@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@pernau.at wrote:
Cesc wrote:
On 12/9/06, Atle Samuelsen clona@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&a...
also main.c wil show you that there are some more undocumented parameters.
Cesc
- Atle
Regards,
Cesc _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Klaus Darilion nic.at
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@cyberhouse.no wrote:
Hi again Cesc,
- Cesc cesc.santa@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@pernau.at wrote:
Cesc wrote:
On 12/9/06, Atle Samuelsen clona@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&a...
also main.c wil show you that there are some more undocumented parameters.
Cesc
- Atle
Regards,
Cesc _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Klaus Darilion nic.at
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
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
-
Atle Samuelsen -
Cesc -
Daniel-Constantin Mierla -
Klaus Darilion -
Wienand, Joerg