sr-users September 2012

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Re: [SR-Users] Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

thank you for helps cenk and pavel. I see that I had wrong information about kamailio.

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Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

juha writes: > this kind of questions have been asked many times on this list. search > archive for responses. you are right maybe but it is hard to search the archieve. And if it is easier to say "serach" than please show me way to search the arcieve quickly. In this link I m trying to search archieve month by month: http://lists.sip-router.org/pipermail/sr-users/ I m new and thanks your kind help...

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 14 Sep '12

14 Sep '12

Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or CANCEL. It'll just figure out what to do on its own. None of this has to do with dialog state, though. Just rtpproxy control. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: > >> Peter: Thanks for the tip! Really interesting. But I do not >> understand, why also this list contains the calls that were ended by >> sipp... Should I search for some mistake in my kamaillio config ? > > Perhaps you don't close them with unforce_rtp_proxy: > > if(method=="BYE" || method=="CANCEL"){ > unforce_rtp_proxy(); > } > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] Max-Forwards: and loops. Testing purpose.
by Daniel-Constantin Mierla 14 Sep '12

14 Sep '12

Hello, do not write private emails from mailing list discussions, if not sensitive data was asked explicitly. If there is no rule to breaking the loop, after some time the message gets very big. Cheers, Daniel On 9/14/12 7:03 AM, Anton Kvashenkin wrote: > Thanks for respond. I just have used forward() based on RURI. > > got > > 0(2558) ERROR: <core> [msg_translator.c:1835]: ERROR: > build_req_buf_from_sip_req: out of memory > 0(2558) ERROR: <core> [forward.c:601]: ERROR: forward_request: > building failed > 0(2558) WARNING: <core> [receive.c:211]: WARNING: receive_msg: error > while trying script > > With maxfwd it works like a charm. http://paste.ubuntu.com/1204093/ > > 2012/9/12 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > > > On 9/11/12 1:12 PM, Anton Kvashenkin wrote: > > Hi, List. > > I'm using sipsak to fire some predefined sip message with > Max-Forwards: header equal 0 to test mf_process_maxfwd_header > function. I'm doing this exclusively for learning purpose. As > far as we know, this exported function from maxfwd module > services only for adding Max-Forwards header, decrement or > detect Max-Forwards: 0 header to prevent loops. So, my > question guys, how can I create this loop in my lab, with two > proxies for example? What message should I craft? > > what kind of message are you looking to craft? > > Looping from one proxy to another can be achieved with > t_relay("_the other proxy address_") function from tm module. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla - http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - > http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - > http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - > http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

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Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

Does kamailio supports g729 codec? Or which codecs does kamailio supports? Where or how to learn the supported or unsupported codecs for kamailio? Thanks in advance...

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

That's a rather dated study. But it's better than a nonexistent reference point, true. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:According to this (http://transnexus.com/index.php/performance-test-results-for-openser-and-rt…) "For a server hosting both OpenSER and RTPproxy, each 1 GHz of CPU processing capacity can manage a maximum of 325 simultaneous calls." I have 2.4GHz for rtpproxy, but CPU/Mem/network is ok, so the bottleneck should be somewhere else probably.. On Thu, Sep 13, 2012 at 5:56 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > I'm not sure what a single instance of rtpproxy can handle, but most people > squeezing thousand of concurrent calls per box are probably doing it on > multicore boxes by binding multiple instances of rtpproxy with different > core affinities, and round-robining among them. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/ > > Mino Haluz <mino.haluz(a)gmail.com> wrote: > The results: > > - rtpproxy calls count 280 > - sipp calls count 2000 > - iptraf on proxy 4.8MB/s > - G711a codec > > So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), > rtpproxy calls count is really the right value. CPU usage is ok on > every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy > cannot serve more than 270-280 calls ? > > On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: >> Ok, so I put there unforce_rtp_proxy even though I'm using >> rtpproxy_manage. The tip with nc now really shows the calls count. >> >> But the dialog count is still higher and higher, so I have bug >> somewhere in the configuration. I'll check it. >> >> On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov >> <abalashov(a)evaristesys.com> wrote: >>> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE >>> or >>> CANCEL. It'll just figure out what to do on its own. >>> >>> None of this has to do with dialog state, though. Just rtpproxy control. >>> >>> >>> >>> >>> -- Alex >>> >>> -- >>> Sent from my Samsung mobile, and thus lacking in the refinement one might >>> expect from a proper keyboard. >>> >>> Alex Balashov - Principal >>> Evariste Systems LLC >>> 235 E Ponce de Leon Ave >>> Suite 106 >>> Decatur, GA 30030 >>> Tel: +1-678-954-0670 >>> Web: http://www.evaristesys.com/ >>> >>> Mino Haluz <mino.haluz(a)gmail.com> wrote: >>> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >>> >>> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> >>> wrote: >>>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>>> >>>>> Peter: Thanks for the tip! Really interesting. But I do not >>>>> understand, why also this list contains the calls that were ended by >>>>> sipp... Should I search for some mistake in my kamaillio config ? >>>> >>>> Perhaps you don't close them with unforce_rtp_proxy: >>>> >>>> if(method=="BYE" || method=="CANCEL"){ >>>> unforce_rtp_proxy(); >>>> } >>>> >>>> -- >>>> With best regards, Peter Lemenkov. >>>> >>>> _______________________________________________ >>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>>> sr-users(a)lists.sip-router.org >>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

I'm not sure what a single instance of rtpproxy can handle, but most people squeezing thousand of concurrent calls per box are probably doing it on multicore boxes by binding multiple instances of rtpproxy with different core affinities, and round-robining among them. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:The results: - rtpproxy calls count 280 - sipp calls count 2000 - iptraf on proxy 4.8MB/s - G711a codec So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), rtpproxy calls count is really the right value. CPU usage is ok on every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy cannot serve more than 270-280 calls ? On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: > Ok, so I put there unforce_rtp_proxy even though I'm using > rtpproxy_manage. The tip with nc now really shows the calls count. > > But the dialog count is still higher and higher, so I have bug > somewhere in the configuration. I'll check it. > > On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or >> CANCEL. It'll just figure out what to do on its own. >> >> None of this has to do with dialog state, though. Just rtpproxy control. >> >> >> >> >> -- Alex >> >> -- >> Sent from my Samsung mobile, and thus lacking in the refinement one might >> expect from a proper keyboard. >> >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Decatur, GA 30030 >> Tel: +1-678-954-0670 >> Web: http://www.evaristesys.com/ >> >> Mino Haluz <mino.haluz(a)gmail.com> wrote: >> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >> >> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: >>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>> >>>> Peter: Thanks for the tip! Really interesting. But I do not >>>> understand, why also this list contains the calls that were ended by >>>> sipp... Should I search for some mistake in my kamaillio config ? >>> >>> Perhaps you don't close them with unforce_rtp_proxy: >>> >>> if(method=="BYE" || method=="CANCEL"){ >>> unforce_rtp_proxy(); >>> } >>> >>> -- >>> With best regards, Peter Lemenkov. >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Really? Interesting, I had no idea. I thought the rtpproxy control protocol was binary and did not lend itself easily to interaction in this manner. Thanks for the tip. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > You can't get it from rtpproxy. You'd really have to use something like the > dialog or htable modules to keep call state and get that from Kamailio. On the contrary it's possible (using raw UDP reads/writes): work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 sessions created: 0 active sessions: 0 active streams: 0 work ~: Where * h1u203u03 is randomly chosen token, * 127.0.0.1 is the rtpproxy's control IP, * 22222 is the rtpproxy's control port, * "-u" means that we're using UDP * -w 1 is the timeout in seconds to wait before closing nc. I can't imagine that someone will use nc in performance testing but I think it looks like a good start. -- With best regards, Peter Lemenkov.

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Thanks for this info. Netcat is a really underappreciated tool. I can't count how many times I've unnecessarily written utilities to send stuff on UDP sockets, forgetting that nc can do it. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. Yep, this command protocol allows us to do tricks like this which is quite helpful for debugging. Unfortunately this particular command ( 'I' and its counterpart 'Ib' ) seems to be broken - it doesn't prepend received random cookie at the beginning of the answer. Just FYI - I collected some examples of control commands here (just grep for ser_proto:decode): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_proto_test.erl and here (which could be even more interesting since it contains an example of a real session - create session, lookup, destroy, get stats, etc): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_test.erl > Thanks for the tip. Anytime :) -- With best regards, Peter Lemenkov.

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

If the dialog module is working correctly, and the SIP flows are standards-compliant, the dialog module should automatically track all subsequent state changes and remove calls from tracking after a BYE is processed. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Mino Haluz <mino.haluz(a)gmail.com> wrote:Ok, I'm tagging dialogs with dlg_manage(), but even if the call ends, it still keeps info about this dialog in list "kamctl fifo dlg_list". Should I somehow close the dialog when the BYE transaction is ended ? Peter: Thanks for the tip! Really interesting. But I do not understand, why also this list contains the calls that were ended by sipp... Should I search for some mistake in my kamaillio config ? On Thu, Sep 13, 2012 at 3:57 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. > Thanks for the tip. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > Alex Balashov - Principal > Evariste Systems LLC > 235 E Ponce de Leon Ave > Suite 106 > Decatur, GA 30030 > Tel: +1-678-954-0670 > Web: http://www.evaristesys.com/ > > Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: >> You can't get it from rtpproxy. You'd really have to use something like >> the >> dialog or htable modules to keep call state and get that from Kamailio. > > On the contrary it's possible (using raw UDP reads/writes): > > work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 > sessions created: 0 > active sessions: 0 > active streams: 0 > work ~: > > Where > > * h1u203u03 is randomly chosen token, > * 127.0.0.1 is the rtpproxy's control IP, > * 22222 is the rtpproxy's control port, > * "-u" means that we're using UDP > * -w 1 is the timeout in seconds to wait before closing nc. > > I can't imagine that someone will use nc in performance testing but I > think it looks like a good start. > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

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sr-users September 2012