July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a
by Hoexum, Edwin 25 Jul '12

25 Jul '12

Gents, I am trying to do some load testing on a freeswitch for transcoding from SILK to G711a. The question is: Is the limit for RTP sessions limited to CPU power or are there also limits on the freeswitch application and can I tune freeswitch to do the job for 1500 sessions or more. At the moment freeswitch is the best transcoding device for SILK to G711a that I know of. Also compared to the big companies who deliver transcoding devices. SETUP: 3 X G7 HP DL 380 with 16 G mem 6 core 12 CPU's 1 for SIPP UAC (1Gb interface for media and Signaling) 1 for Freeswitch (trancoding function SILK to G711a) 1Gb interface for media and Signaling 1 for SIPP UAS (1Gb interface for media and Signaling) First tuning OS (with help from the wiki and some LINUX sysctl tuning) SIPP UAC <-> Freeswitch root@mnd-rc0001-srvot02:~# iperf -c 172.23.125.46 -u -d -b 400m ------------------------------------------------------------ Server listening on UDP port 5001 Receiving 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ ------------------------------------------------------------ Client connecting to 172.23.125.46, UDP port 5001 Sending 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ [ 4] local 172.23.125.42 port 46443 connected with 172.23.125.46 port 5001 [ 3] local 172.23.125.42 port 5001 connected with 172.23.125.46 port 39259 [ ID] Interval Transfer Bandwidth [ 4] 0.0-10.0 sec 483 MBytes 405 Mbits/sec [ 4] Sent 344796 datagrams [ 3] 0.0-10.0 sec 483 MBytes 406 Mbits/sec 0.001 ms 0/344828 (0%) [ 3] 0.0-10.0 sec 1 datagrams received out-of-order [ 4] Server Report: [ 4] 0.0-10.0 sec 483 MBytes 405 Mbits/sec 0.004 ms 261/344795 (0.076%) [ 4] 0.0-10.0 sec 1 datagrams received out-of-order root@mnd-rc0001-srvot02:~# Linux mnd-rc0001-srvot06 2.6.32-28-vserver #55~ppa1-Ubuntu SMP Fri Feb 4 21:25:09 UTC 2011 x86_64 GNU/Linux Ubuntu 10.04 LTS Welcome to Ubuntu! * Documentation: https://help.ubuntu.com/ Last login: Thu Jun 21 15:01:35 2012 from 172.21.88.65 root@mnd-rc0001-srvot06:~# iperf -s -u -i1 ------------------------------------------------------------ Server listening on UDP port 5001 Receiving 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ ------------------------------------------------------------ Client connecting to 172.23.125.42, UDP port 5001 Sending 1470 byte datagrams UDP buffer size: 64.0 KByte (default) ------------------------------------------------------------ [ 5] local 172.23.125.46 port 39259 connected with 172.23.125.42 port 5001 [ 3] local 172.23.125.46 port 5001 connected with 172.23.125.42 port 46443 [ ID] Interval Transfer Bandwidth [ 5] 0.0- 1.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 0.0- 1.0 sec 48.4 MBytes 406 Mbits/sec 0.004 ms 262/34754 (0.75%) [ 5] 1.0- 2.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 1.0- 2.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34476 (0%) [ 5] 2.0- 3.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 2.0- 3.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34480 (0%) [ 5] 3.0- 4.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 3.0- 4.0 sec 48.3 MBytes 406 Mbits/sec 0.004 ms 0/34483 (0%) [ 5] 4.0- 5.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 4.0- 5.0 sec 48.3 MBytes 406 Mbits/sec 0.005 ms 0/34482 (0%) [ 5] 5.0- 6.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 5.0- 6.0 sec 48.3 MBytes 405 Mbits/sec 0.004 ms 0/34474 (0%) [ 5] 6.0- 7.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 6.0- 7.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34477 (0%) [ 5] 7.0- 8.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 7.0- 8.0 sec 48.3 MBytes 405 Mbits/sec 0.004 ms 0/34480 (0%) [ 5] 8.0- 9.0 sec 48.3 MBytes 406 Mbits/sec [ 3] 8.0- 9.0 sec 48.3 MBytes 406 Mbits/sec 0.005 ms 0/34482 (0%) [ 5] 9.0-10.0 sec 48.3 MBytes 406 Mbits/sec [ 5] 0.0-10.0 sec 483 MBytes 406 Mbits/sec [ 5] Sent 344829 datagrams [ 3] 0.0-10.0 sec 483 MBytes 405 Mbits/sec 0.004 ms 261/344795 (0.076%) [ 3] 0.0-10.0 sec 1 datagrams received out-of-order [ 5] Server Report: [ 5] 0.0-10.0 sec 483 MBytes 406 Mbits/sec 0.000 ms 0/344828 (0%) [ 5] 0.0-10.0 sec 1 datagrams received out-of-order UDP traffic looks fine. SIPP (UAC) inject SILK RTP (PCAP ) and send this to Freeswitch. Freeswitch send this G711a to the SIPP( UAS). I doing some testing and until 700 sessions it look fine (Packed loss <0.1% and delay <40ms) If I do more sessions 10% of packet have no packet loss but delay of more the 100ms. SIPP uac side: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Call-rate(length) Port Total-time Total-calls Remote-host 50.0(0 ms)/1.000s 5060 230.53 s 2000 172.23.125.12:5060(UDP) Call limit reached (-m 2000), 0.000 s period 0 ms scheduler resolution 0 calls (limit 4500) Peak was 2000 calls, after 40 s 0 Running, 1654 Paused, 0 Woken up 0 dead call msg (discarded) 6459 out-of-call msg (discarded) 1 open sockets Messages Retrans Timeout Unexpected-Msg Pause [ 2000ms] 2000 0 INVITE ----------> 2000 0 0 100 <---------- 2000 0 0 0 183 <---------- 0 0 0 0 400 <---------- 0 0 0 0 401 <---------- 0 0 0 0 403 <---------- 0 0 0 0 404 <---------- 0 0 0 0 405 <---------- 0 0 0 0 406 <---------- 0 0 0 0 408 <---------- 0 0 0 0 415 <---------- 0 0 0 0 433 <---------- 0 0 0 0 480 <---------- 0 0 0 0 481 <---------- 0 0 0 0 484 <---------- 0 0 0 0 485 <---------- 0 0 0 0 486 <---------- 0 0 0 0 487 <---------- 0 0 0 0 488 <---------- 0 0 0 0 489 <---------- 0 0 0 0 500 <---------- 0 0 0 0 502 <---------- 0 0 0 0 503 <---------- 0 0 0 0 180 <---------- 2000 0 0 0 200 <---------- E-RTD1 2000 0 0 0 ACK ----------> 2000 0 Pause [ 500ms] 2000 0 [ NOP ] BYE <---------- 2000 0 0 0 200 ----------> 2000 0 ------------------------------ Test Terminated -------------------------------- ----------------------------- Statistics Screen ------- [1-9]: Change Screen -- Start Time | 2012-06-28 20:17:17:312 1340907437.312927 Last Reset Time | 2012-06-28 20:21:07:857 1340907667.857562 Current Time | 2012-06-28 20:21:07:857 1340907667.857769 -------------------------+---------------------------+-------------------------- Counter Name | Periodic value | Cumulative value -------------------------+---------------------------+-------------------------- Elapsed Time | 00:00:00:000 | 00:03:50:544 Call Rate | 0.000 cps | 8.675 cps -------------------------+---------------------------+-------------------------- Incoming call created | 0 | 0 OutGoing call created | 0 | 2000 Total Call created | | 2000 Current Call | 0 | SIPP UAS side: ------------------------------ Scenario Screen -------- [1-9]: Change Screen -- Port Total-time Total-calls Transport 5060 65.10 s 2000 UDP 0 new calls during 1.001 s period 1 ms scheduler resolution 2000 calls Peak was 2000 calls, after 44 s 0 Running, 2002 Paused, 4 Woken up 0 dead call msg (discarded) 3 open sockets 2021 Total echo RTP pckts 1st stream 5.327 last period RTP rate (kB/s) 0 Total echo RTP pckts 2nd stream 0.000 last period RTP rate (kB/s) Messages Retrans Timeout Unexpected-Msg ----------> INVITE 2000 0 0 0 <---------- 100 2000 0 [ 500ms] Pause 2000 0 <---------- 180 2000 0 <---------- 200 2000 0 0 ----------> ACK 2000 0 0 0 [ 3:08] Pause 2000 0 <---------- BYE 0 0 0 ----------> 200 0 0 0 0 ------------------------------ Sipp Server Mode ------------------------------- ############################################################################## MPSTAT -A 5 on freeswitch: 8:29:50 CPU %usr %nice %sys %iowait %irq %soft %steal %guest %idle 18:29:55 all 24.84 0.00 25.63 0.00 0.02 7.46 0.00 0.00 42.05 18:29:55 0 43.52 0.00 26.72 0.00 0.00 4.05 0.00 0.00 25.71 18:29:55 1 27.88 0.00 39.18 0.00 0.00 3.12 0.00 0.00 29.82 18:29:55 2 27.70 0.00 33.40 0.00 0.00 4.17 0.00 0.00 34.72 18:29:55 3 25.57 0.00 35.98 0.00 0.00 3.03 0.00 0.00 35.42 18:29:55 4 27.36 0.00 27.56 0.00 0.00 2.56 0.00 0.00 42.52 18:29:55 5 25.45 0.00 24.44 0.00 0.00 2.22 0.00 0.00 47.88 18:29:55 6 24.77 0.00 31.28 0.00 0.00 2.98 0.00 0.00 40.97 18:29:55 7 23.05 0.00 23.81 0.00 0.00 2.67 0.00 0.00 50.48 18:29:55 8 22.22 0.00 23.03 0.00 0.20 2.02 0.00 0.00 52.53 18:29:55 9 4.84 0.00 5.71 0.00 0.00 62.86 0.00 0.00 26.59 18:29:55 10 23.36 0.00 17.01 0.00 0.00 2.46 0.00 0.00 57.17 18:29:55 11 20.21 0.00 15.11 0.00 0.00 2.98 0.00 0.00 61.70 18:29:50 CPU intr/s 18:29:55 all 30433.80 18:29:55 0 0.00 18:29:55 1 0.00 18:29:55 2 0.00 18:29:55 3 0.00 18:29:55 4 0.00 18:29:55 5 0.00 18:29:55 6 0.00 18:29:55 7 0.00 18:29:55 8 0.00 18:29:55 9 0.00 18:29:55 10 0.00 18:29:55 11 0.00 18:29:50 CPU 0/s 1/s 8/s 18:29:55 0 0.00 0.00 0.00 18:29:55 1 0.00 0.00 0.00 18:29:55 2 0.00 0.00 0.00 18:29:55 3 0.00 0.00 0.00 18:29:55 4 0.00 0.00 0.00 18:29:55 5 0.00 0.00 0.00 18:29:55 6 0.00 0.00 0.00 18:29:55 7 0.00 0.00 0.00 18:29:55 8 0.00 0.00 0.00 18:29:55 9 0.00 0.00 0.00 18:29:55 10 0.00 0.00 0.00 18:29:55 11 0.00 0.00 0.00 I tested it with G711a to G711a and got the same result. I tested also some RTP parameters like <param name="rtp-autoflush-during-bridge" value="true"/> or false <param name="rtp-autoflush" value="true"/> or false The lot of the module are not activated like CDR stuff. mvg, Edwin ________________________________ This message is confidential and may be privileged. Any review, retransmission, dissemination or other use of, taking any action with reference to this information by persons other then the intended recipient is prohibited. If you receive this message in error, please notify the sender by reply e-mail and delete this message from all computers. Please note that e-mails are susceptible to change. The sender will not accept liability for the improper or incomplete transmission of the information contained in this message. Spaar het milieu door deze e-mail niet te printen/Please consider the environment before printing this email.

9 18

Testing iSAC
by Martyn Davies 25 Jul '12

25 Jul '12

I've now got: global_codec_prefs=isac@16000h,isac,PCMU,PCMA,GSM,G722 and I can connect in 16 and 32 modes (inbound to the freeswitch). I haven't got outbound iSAC working from the FreeSwitch yet, but I'm guessing that the client doesn't like the CNG SDP option (13). I've added this to the dialplan: <action application="export" data="suppress_cng=true"/> and restarted, but still see the 13 option going out. How to disable this? Regards, Martyn On 25 July 2012 08:45, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: No audio after upgrade to F17 (Chris Mylonas) > 2. Re: Testing iSAC (curriegrad2004) > 3. mod_spandsp compilation error (Evgeniy Movlyan) > 4. Re: mod_spandsp compilation error (Peter Olsson) > 5. Re: mod_spandsp compilation error (Dmitry R) > > > ---------- Forwarded message ---------- > From: Chris Mylonas <chris(a)opencsta.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 25 Jul 2012 09:23:47 +1000 > Subject: Re: [Freeswitch-users] No audio after upgrade to F17 > Further to Paul's comments > > In a terminal > tail -f /var/log/messages > > In another terminal > tail -f /path/to/your/freeswitch/log/freeswitch.log > > In another terminal, start freeswitch > > > If it is an SELinux problem, which is probably suspected, there will be a kernel message stating something along the lines of a context is not allowed to do something. > I had similar issues with openvpn and SELinux - this blog post may be useful in the process of keeping SELinux running and customising it's profile for running freeswitch. > http://www.mrvoip.com.au/blog/selinux-openvpn > > Alternatively, look at ways of disabling SELinux - on other redhat related distros, it's a matter of editing a setting in a file to "disabled" and rebooting so the kernel doesn't load SELinux stuff. Another alternative is to change SELinux's mode from "enforcing" to "permissive" > > > Cheers > Chris > > > > > > On 25/07/2012, at 3:56 AM, Paul Cupis wrote: > > On 24/07/12 17:21, Todd Bailey wrote: > > On Mon, 2012-07-23 at 12:20 -0700, Todd wrote: > > When I installed F17, I copied the FS installation directly > > > Booting back to F14, every this works as expected, so it's not the > > spa3102 or config files. > > > Thinking there might be a incompatibility issues, I recompiled FS under > > F17, no change in operational behavior. > > > Ideas where else to look ? > > > Have a look at the "new features" in F15, F16 or F17, you'll probably > find some security function (SElinux or similar) which may be blocking > FreeSWITCH RTP traffic. > > Regards, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: curriegrad2004 <curriegrad2004(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 24 Jul 2012 20:45:34 -0700 > Subject: Re: [Freeswitch-users] Testing iSAC > > You might want to report this issue in the JIRA as a feature request. > > Iirc it doesn't support that yet. > > On Jul 24, 2012 6:26 AM, "Martyn Davies" <martyn(a)magiccow.co.uk> wrote: >> >> I have an iSAC client that I'm testing against FreeSwitch. >> >> Currently I have my vars.xml with: >> >> global_codec_prefs=isac,PCMU,PCMA,GSM,G722 >> >> Which works, but only if the client specifies 32kHz mode. I want to test 16kHz mode, but FreeSwitch returns 'not acceptable here', and I can see that when Freeswitch offers iSAC in the SDP it says only isac/32000. >> >> So how to enable 16kHz mode for isac? >> >> I tried explicitly putting isac@320000h30i, isac@16000h30i into vars.xml, but this simply stops FreeSwitch from accepting anything. >> >> Hints gladly received. >> >> Regards, >> Martyn >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ---------- Forwarded message ---------- > From: Evgeniy Movlyan <evgeniy(a)bestnet.kharkov.ua> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Wed, 25 Jul 2012 10:12:18 +0300 > Subject: [Freeswitch-users] mod_spandsp compilation error > Hi to all, > i made make current command today, and got this: > > making all mod_spandsp > make[5]: Entering directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o > In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, > from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error: expected specifier-qualifier-list before ‘lab_params_t’ > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: error: expected specifier-qualifier-list before ‘lab_params_t’ > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > make[4]: *** [mod_spandsp-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Any ideas? > > -- > Evgeniy Movlyan, > BestNet Ltd. > > > > > ---------- Forwarded message ---------- > From: Peter Olsson <peter.olsson(a)visionutveckling.se> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 25 Jul 2012 07:42:30 +0000 > Subject: Re: [Freeswitch-users] mod_spandsp compilation error > 'make spandsp-reconf'. If it doesn't help, file a Jira. > > /Peter > > 25 jul 2012 kl. 09:19 skrev "Evgeniy Movlyan" <evgeniy(a)bestnet.kharkov.ua>: > > > Hi to all, > > i made make current command today, and got this: > > > > making all mod_spandsp > > make[5]: Entering directory > > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > > Creating mod_spandsp_la-mod_spandsp.lo > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > > -I../../../../src/include -I../../../../libs/xmlrpc-c > > -I/usr/local/src/freeswitch/libs/curl/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC > > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > > -I/usr/local/src/freeswitch/libs/curl/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > > -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic > > -Wdeclaration-after-statement > > -I/usr/local/src/freeswitch/libs/spandsp/src > > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > > -I/usr/local/src/freeswitch/libs/spandsp/src > > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > > .libs/mod_spandsp_la-mod_spandsp.o > > In file included from > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, > > from > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, > > from mod_spandsp.h:50, > > from mod_spandsp.c:39: > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: > > error: expected specifier-qualifier-list before ‘lab_params_t’ > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: > > error: expected specifier-qualifier-list before ‘lab_params_t’ > > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > > make[5]: Leaving directory > > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > > make[4]: *** [mod_spandsp-all] Error 1 > > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > > make[3]: *** [all-recursive] Error 1 > > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > > make[2]: *** [all-recursive] Error 1 > > make[2]: Leaving directory `/usr/local/src/freeswitch' > > make[1]: *** [all] Error 2 > > make[1]: Leaving directory `/usr/local/src/freeswitch' > > make: *** [current] Error 2 > > > > Any ideas? > > > > -- > > Evgeniy Movlyan, > > BestNet Ltd. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > !DSPAM:500f9ab132761590495690! > > > > > > > ---------- Forwarded message ---------- > From: "Dmitry R" <rdmitry0911(a)gmail.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 25 Jul 2012 11:34:28 +0400 > Subject: Re: [Freeswitch-users] mod_spandsp compilation error > I got the same problem. Solved after git pull && ./bootstrap.sh && > ./configure && make > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Evgeniy > Movlyan > Sent: Wednesday, July 25, 2012 11:12 AM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] mod_spandsp compilation error > > Hi to all, > i made make current command today, and got this: > > making all mod_spandsp > make[5]: Entering directory > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > -I../../../../src/include -I../../../../libs/xmlrpc-c > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > .libs/mod_spandsp_la-mod_spandsp.o > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, > from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: > error: expected specifier-qualifier-list before 'lab_params_t' > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: > error: expected specifier-qualifier-list before 'lab_params_t' > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[5]: Leaving directory > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > make[4]: *** [mod_spandsp-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Any ideas? > > -- > Evgeniy Movlyan, > BestNet Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

1 0

Testing iSAC
by Martyn Davies 25 Jul '12

25 Jul '12

I have an iSAC client that I'm testing against FreeSwitch. Currently I have my vars.xml with: global_codec_prefs=isac,PCMU,PCMA,GSM,G722 Which works, but only if the client specifies 32kHz mode. I want to test 16kHz mode, but FreeSwitch returns 'not acceptable here', and I can see that when Freeswitch offers iSAC in the SDP it says only isac/32000. So how to enable 16kHz mode for isac? I tried explicitly putting isac@320000h30i, isac@16000h30i into vars.xml, but this simply stops FreeSwitch from accepting anything. Hints gladly received. Regards, Martyn

2 1

No audio after upgrade to F17
by Todd 24 Jul '12

24 Jul '12

Well this is odd, after I installed F17, I don't get audio to or from the pstn line (via a spa3102 ata) once the call is answered. I do get ring tones, and Fs generated voice messages, and can leave or retrieve vm. I've tried a couple approaches so. When I installed F17, I copied the FS installation directly Booting back to F14, every this works as expected, so it's not the spa3102 or config files. Thinking there might be a incompatibility issues, I recompiled FS under F17, no change in operational behavior. Ideas where else to look ?

6 5

Bypass media and NAT Traversal with Aastra
by Phil Quesinberry 24 Jul '12

24 Jul '12

Hi, Is this a bug? (just kidding) As the number of extensions we serve increases, we're trying to avoid saturating our local bandwidth by proxying all of the media between endpoints. This seems to work fine with the Linksys devices but not with Aastra in all scenarios. The Aastras are configured with the NAT IP field populated appropriately and Rport (RFC 3581) enabled. The FS server is behind a NAT firewall with SIP and RTP port ranges forwarded to it. There are endpoints here on the same network as the switch, and there are remote endpoints behind their own NAT firewalls. Everything works fine in Proxy Media mode but in bypass mode we see the following behavior: Scenario 1: Remote endpoint places a call to PSTN, FS negotiates bypass media between remote endpoint and PSTN gateway. Call succeeds. Scenario 2: Remote endpoint places a call to another remote endpoint on same LAN. Call succeeds. Scenario 3: Local endpoint places a call to PSTN gateway. Call succeeds. Scenario 4: Remote endpoint places a call to local endpoint (or vice versa). Call fails with no audio with Aastra devices on both ends, but if one of the endpoints is a LinkSys, the call succeeds. I realize that I should be able to solve this problem with a local SBC like OpenSIPS/RTPproxy, but in the meantime I was hoping that there might be a setting that I was overlooking with the Aastra. When I look at the SIP traffic for the call, FS appears to be negotiating the addresses/ports properly so I suspect that this problem is related to Aastra's inability to deal with symmetric NAT. The Linksys devices seem to be able to deal with it with FS' help, presumably by doing some UDP hole-punching. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

2 1

error 403 registering at 1und1.de
by Christian Tausch 24 Jul '12

24 Jul '12

Hi list, i ran into a problem configuring the german voip-provider 1und1.de as a gateway. I have not completed my configuration yet, but in a first test the registration fails: The response to the second REGISTER message is "403: Contact User und Anrufernummer verschieden" (en: 403 Authentication User Name does not match account name). The visible difference between the REGISTER of freeswitch and the (working) SIP-client twinkle is indeed the user-string in the contact-header. Freeswitch: (gw + name) Contact: <sip:gw+1und1.de@<public-ip>:5080;transport=udp;gw=1und1.de> Twinkle: (extension / login-user) Contact: <sip:4921XXXXXXXXX@<public-ip>;transport=udp>;expires=3600 If this is the cause of the error message, is this expected behavior of a provider? And more important is it possible to configure freeswitch to set a specific contact-user? Thank you for your time. Christian Tausch -- Attachment (conf/directory/default/1und1.de.xml): <include> <user id="1und1.de"> <gateways> <gateway name="1und1.de"> <param name="realm" value="1und1.de"/> <param name="username" value="4921XXXXXXXXX"/> <param name="password" value="password"/> <param name="auth-username" value="4921XXXXXXXXX"/> <param name="proxy" value="1und1.de"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="context" value="public"/> </gateway> </gateways> <params> <param name="password" value="password"/> </params> </user> </include>

2 1

AOC for SIP
by Frank Rosengart 24 Jul '12

24 Jul '12

Hi, I'm wondering if anyone has implemented some kind of AOC (advice of charge) for SIP channels? This is either a frequent INFO message during the call or an extra to the BYE message. This includes pricing calculation based on call duration and destination (can be handled externally) and sending out the AOC message (either in the Snom-way or the Patton-dialect). Thanks! Frank http://wiki.snom.com/wiki/index.php/Advice_of_charge_%28AOC%29_in_SIP

1 0

dialplan commands on playback
by Alexandre Fiori 24 Jul '12

24 Jul '12

is there any way to tell fs to execute a command at, say, 75% of a playback? like, call or schedule a detect_speech (or read) halfway through - af

3 9

Important Information About Not Reporting Bugs on the Mailing List
by Michael Collins 24 Jul '12

24 Jul '12

Hello all, We've had yet another thread where someone wasn't sure if there was a but so they asked on the mailing list. I've written up some simple guidelines and put them on the world-famous Bugs wiki page: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Important_Note.2C_Please_Read In short: not only should no one report bugs to the mailing list, no one should report symptoms to the mailing list and ask, "Is this a bug? Should I file a Jira?" If you have collected enough information to send an email to the list and ask whether or not to file a Jira then there's really no point in asking - just file the Jira. If you would like to discuss the symptoms on the mailing list then open the Jira *first* and then link to it in the message you send to the list. It's much better to open a Jira and find out later that it's not really a bug than it is to report a possible bug on the mailing list. Even if something is flagged as "not a bug" or "works as designed" there is still value in having the Jira ticket in the database for historical and research purposes. For those who have heard this discussion before, please be kind to the noobs and first-timers who may not appreciate the issues involved. Just point them to the bugs page. If someone repeatedly reports bugs to the mailing list then we will deliver the message with less delicacy. Thanks again for all your help. Using Jira will help keep Tony's life from become any more complicated than it already is. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

4 3

mod_avmd
by BF 24 Jul '12

24 Jul '12

Hello, I'm trying to use it from a Lua script. My understanding is that mod_avmd detects beep from voicemail systems and is CPU intensive, please correct me if I'm wrong. The Lua example at FreeSWITCH Wiki is local human_detected = false; local voicemail_detected = false; function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then human_detected = true; return "break"; end if type == "event" and voicemail_detected == false then voicemail_detected = true; return "break"; end end session:setInputCallback("onInput"); session:execute("avmd","start"); In order to implement this example, the script must wait for the beep to be detected or not to process the case accordingly. What is no beep is detected? How can I prevent called party to hear only silence while potential beep detection is being executed? Thank you Bernard

4 10