August 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

record and silence detection
by mbo 15 Aug '14

15 Aug '14

Hello, I found out that a call is hungup, when record is stopped by silence detection, however when the time limit is reached, it continues with the next step in the dialplan. Is it somehow configureable that it always continues with the next step. ... <action application="playback" data="/var/sounds/beep.gsm"/> <action application="record" data="/tmp/data.wav 30 50 10“/> <action application="playback" data="/var/sounds/beep.gsm"/> <action application="record" data="/tmp/data.wav 30 50 10“/> ... I'm using Version 1.4.2+git~20140108T200418Z~d8fc8469b4~64bit Thanks Markus

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Fwd: Freeswitch Dialer using SIPml5
by sami sakly 15 Aug '14

15 Aug '14

---------- Forwarded message ---------- From: sami sakly <it.samisakly(a)gmail.com> Date: 2014-08-13 22:56 GMT+02:00 Subject: Freeswitch Dialer using SIPml5 To: freeswitch-dev(a)lists.freeswitch.org As it appears from the thread title I'm new to Freeswitch and Voip solution. So please excuse my ignorance if I didn't get you from the first time. I'm building a Dialer as solution for small compaign based on freeswitch. The main idea is that when agent are present : I'm not sure they call fs server or it calls them and starting dialing customers from a database and whenever a client pick up the phone : it's automatically bridged to one of the free agent (already parked and waiting). What I have now ? : *Freeswitch & webrtc2sip working on my server. *SIPml5 Client issuing calls and working I tried also to write some php script to run from a client host , that calls 2 sip client and do the bridging. What I want to do now ? : I'm looking for a way to apply my business model automatically : 1- Calling agent and parking 2- Dialling customers and whenever it's picked up bridge to a free agent My questions : will it be possible to do this with only dialplans or I need to have some external script ? Where exactly I can use fs events ? I will need them on the php app to create control on dialling client parameters and for bridging. That seems a lot but I hope I will not be disappointed =) Thank you,

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This Wiki is LOCKED!
by Mujahid Ali 15 Aug '14

15 Aug '14

why i see this message on https://wiki.freeswitch.org/wiki/Main_Page even every page and git also didn't work This Wiki is LOCKED! Migration to http://confluence.freeswitch.org is under way! Please visit Confluence for the latest updates!

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Freeswitch rpm building make and make install
by Mujahid Ali 15 Aug '14

15 Aug '14

making all mod_gsmopen make[4]: Entering directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src/mod/endpoints/mod_gsmopen' make[5]: Entering directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src/mod/endpoints/mod_gsmopen' Compiling gsmopen_protocol.cpp... fatal: Not a git repository (or any of the parent directories): .git fatal: Not a git repository (or any of the parent directories): .git g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-4.0.2/libtiff -DGSMOPEN_C_VER="" -DMODGSMOPEN_C_VER="" -I/root/rpmbuild/BUILD/freeswitch-1.2.23/src/include -I/root/rpmbuild/BUILD/freeswitch-1.2.23/src/include -I/root/rpmbuild/BUILD/freeswitch-1.2.23/libs/libteletone/src -I/root/rpmbuild/BUILD/freeswitch-1.2.23/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -D_GNU_SOURCE -DHAVE_CONFIG_H -c -o gsmopen_protocol.o gsmopen_protocol.cpp fatal: Not a git repository (or any of the parent directories): .git fatal: Not a git repository (or any of the parent directories): .git In file included from gsmopen_protocol.cpp:37: gsmopen.h:119:26: error: ctb-0.16/ctb.h: No such file or directory In file included from gsmopen_protocol.cpp:37: gsmopen.h:477: error: 'ctb' has not been declared gsmopen.h:477: error: ISO C++ forbids declaration of 'SerialPort' with no type gsmopen.h:477: error: expected ';' before '*' token gsmopen.h:479: error: 'ctb' has not been declared gsmopen.h:479: error: ISO C++ forbids declaration of 'SerialPort' with no type gsmopen.h:479: error: expected ';' before '*' token gsmopen_protocol.cpp: In function 'int gsmopen_serial_init(private_t*, int)': gsmopen_protocol.cpp:117: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp:117: error: expected type-specifier before 'ctb' gsmopen_protocol.cpp:117: error: expected ';' before 'ctb' gsmopen_protocol.cpp:119: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp:119: error: 'ctb' has not been declared gsmopen_protocol.cpp: In function 'int gsmopen_serial_config_AT(private_t*)': gsmopen_protocol.cpp:192: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp: In function 'int gsmopen_serial_read_AT(private_t*, int, int, int, const char*, int)': gsmopen_protocol.cpp:638: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp:786: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp:821: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp:1628: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int gsmopen_serial_write_AT(private_t*, const char*)': gsmopen_protocol.cpp:1872: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp:1879: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp:1923: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp:1930: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp: In function 'int gsmopen_serial_write_AT_nocr(private_t*, const char*)': gsmopen_protocol.cpp:1984: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp:1991: error: 'struct private_object' has no member named 'serialPort_serial_control' gsmopen_protocol.cpp: In function 'int gsmopen_serial_write_AT_noack(private_t*, const char*)': gsmopen_protocol.cpp:2042: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int gsmopen_serial_write_AT_ack(private_t*, const char*)': gsmopen_protocol.cpp:2082: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int gsmopen_serial_write_AT_ack_nocr_longtime(private_t*, const char*)': gsmopen_protocol.cpp:2124: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int gsmopen_serial_write_AT_expect1(private_t*, const char*, const char*, int, int)': gsmopen_protocol.cpp:2166: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int ucs2_to_utf8(private_t*, char*, char*, size_t)': gsmopen_protocol.cpp:2370: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp:2397: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int utf8_to_iso_8859_1(private_t*, char*, size_t, char*, size_t)': gsmopen_protocol.cpp:2439: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int iso_8859_1_to_utf8(private_t*, char*, char*, size_t)': gsmopen_protocol.cpp:2494: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int utf8_to_ucs2(private_t*, char*, size_t, char*, size_t)': gsmopen_protocol.cpp:2542: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp:2551: warning: comparison between signed and unsigned integer expressions gsmopen_protocol.cpp: In function 'int gsmopen_serial_init_audio_port(private_t*, int)': gsmopen_protocol.cpp:3156: error: 'struct private_object' has no member named 'serialPort_serial_audio' gsmopen_protocol.cpp:3156: error: expected type-specifier before 'ctb' gsmopen_protocol.cpp:3156: error: expected ';' before 'ctb' gsmopen_protocol.cpp:3159: error: 'struct private_object' has no member named 'serialPort_serial_audio' gsmopen_protocol.cpp:3159: error: 'ctb' has not been declared gsmopen_protocol.cpp: In function 'int serial_audio_shutdown(private_t*)': gsmopen_protocol.cpp:3191: error: 'struct private_object' has no member named 'serialPort_serial_audio' make[5]: *** [gsmopen_protocol.o] Error 1 make[5]: Leaving directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src/mod/endpoints/mod_gsmopen' make[4]: *** [all] Error 1 make[4]: Leaving directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src/mod/endpoints/mod_gsmopen' make[3]: *** [mod_gsmopen-all] Error 1 make[3]: Leaving directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src/mod' make[3]: Entering directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src' make[3]: Nothing to be done for `all-am'. make[3]: Leaving directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/root/rpmbuild/BUILD/freeswitch-1.2.23/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/root/rpmbuild/BUILD/freeswitch-1.2.23' make: *** [all] Error 2 error: Bad exit status from /var/tmp/rpm-tmp.H9lKTQ (%build) how i can solve it ?

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FreeSWITCH Friday BugHunt Reminder!
by Ken Rice 15 Aug '14

15 Aug '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday BugHunt Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken http://ift.tt/vGSsGm http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Fwd: SRTP problem on the FS
by Herold Park 15 Aug '14

15 Aug '14

---------- Forwarded message ---------- From: Herold Park <heroldpark(a)gmail.com> Date: 2014-08-15 16:09 GMT+09:00 Subject: SRTP problem on the FS To: "freeswitch-users-request(a)lists.freeswitch.org" < freeswitch-users-request(a)lists.freeswitch.org> Hello, everyone! I'm having trouble with SRTP calls between two internal extensions. Whenever I configure the phones to require SRTP calls has wrong. If I select "allow SRTP" so that encryption is not mandatory. I get only encryption on phone to FS in a call. Here's an example of what's happening: Extension 1003 --- SRTP --> FreeSWITCH -- RTP --> 1004 Extension 1003 <-- RTP ---- FreeSWITCH -- SRTP <-- 1004 Phones are LG-ERICSSON IP-8830E model. These phone is supported Secure RTP. These phones are configured to support SRTP, which is AES_CM_128_HMAC_SHA1_80. FreeSWITCH version is FreeSWITCH-1.2.23. conf/default.xml is configed. [image: 본문 이미지 1] I've setup the sip_secure_media=true variable as instructed from https://wiki.freeswitch.org/wiki/Secure_RTP but it's still not working by SRTP. Any help is greatly appreciated! Kind regards, -- 세상은 아름답다. -- 세상은 아름답다.

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Re: [Freeswitch-users] Skypopen failure ?
by Shabbir abbasi 15 Aug '14

15 Aug '14

> 1) you install the skype client downloaded by install.pl in a regular desktop linux > 2) you load skypopen.ko and make the /dev/ device as per start-skype-clients script > 3) you start the skype instance on desktop > 4) you connect to the skype net with username and password and set the > "login at startup" checkbox > 5) you compile the client.c you found in configs dir of the mod_skypopen > source, as per the first line of the client.c file > 6) you exec the client.c program you just compiled > 7) you give the skype client authorization to be connected from skypopen > application and check the "no ask again" checkbox > 8) you exec the client.c program again to be sure it connect no problem > 9) you save a copy of the entire ~/.Skype directory > 10) you use that copy as a substitute of > /usr/local/freeswitch/skypopen/skype_clients_configuration_dir/skype101 to skypeN (eg you copy it N times to skype101 skype102 etc) > 11) you modify the startup_skype_clients.sh script so it simply launches the skype client, without username and password on stdin and without --pipelogin option I have tried all your suggestions but still not working i have deleted /root/.Skype dir rmmod snd_pcm_oss rmmod snd_mixer_oss rmmod snd_seq_oss mknod /dev/dsp c 14 3 insmod /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko lsmod | grep skyp skypopen 13271 0 launched skype on desktop centos 6.5 skype 2.0.0.72 but unable to login error -> server connect failed new skype version is able to login

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Re: [Freeswitch-users] Network Card
by Anthony Minessale 14 Aug '14

14 Aug '14

Ok it isn't obscene. Thread closed. On Aug 14, 2014 7:27 PM, "aademattia(a)comcast.net" <aademattia(a)comcast.net> wrote: > Why is 1000 cps obscene? > Denovo lab uses nokia stack and we get over 1k cps in a server that cost > 2k. > > > > Sent from my HTC > > ----- Reply message ----- > From: "Anthony Minessale" <anthony.minessale(a)gmail.com> > To: "Freeswitch-users" <freeswitch-users(a)lists.freeswitch.org> > Subject: [Freeswitch-users] Network Card > Date: Thu, Aug 14, 2014 6:16 PM > > We don't give advice on load related questions outside what you have > already seen in this thread. > > 1000cps is obscene. If you are actually getting paid for 1000cps you can > surely afford to invest in hardware and professional help. > On Aug 14, 2014 3:01 PM, "Andre" <andretodd(a)verizon.net> wrote: > >> Just testing for now, no production traffic. Trying to find the breaking >> point. >> >> >> >> The reason for the questions are because I’m testing FS and after a >> certain point I am getting very high PDD, more than 6 seconds. Per the >> CDR’s it takes about 6 seconds from start time in CDR Inbound to CDR >> outbound. >> >> I have tried 1 to 10 profiles to see if that makes a difference but it >> doesn’t and 1 to 4 consoles running one profile each still no difference. >> >> >> >> Any idea why the CDR’s should show it would take 6 seconds to get from >> start of inbound to start of outbound? >> >> >> >> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: >> freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Ken Rice >> *Sent:* Thursday, August 14, 2014 2:06 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Network Card >> >> >> >> 1000 CPS the NIC is not the bottle neck this is actually relatively small >> amount of TCP/UDP traffic for just the SIP message. >> >> Now the RTP @ 1000 calls on the other hand assuming G711 at 20ms ptime. >> And assuming 1 leg calls originating from a dialer would generate >> 80megabits of RTP (at the IP layer even more on the wire), and approx 50K >> PPS. Assuming 2 leg’d calls bridging media 160megabit and 100K PPS of just >> RTP... >> >> But then we have to extrapolate peak concurrent from 1000 CPS... Assuming >> dialer traffic again, 75% ASR and 20 second ACD, and lets assume the 25% >> are just hard failed for the calculation (no early media on these) that >> gives us 750 new calls/sec and over the course of 20 seconds we accumulate >> 15,000 concurrent calls.. >> >> So lets take that and extrapolate bandwidth and PPS requirements for >> single leg’d calls of 1.43Gig/sec (on the wire) and 750K PPS so that’s >> going to exceed the bandwidth of a GIG-E connection so you’d have to break >> this into 2 RTP endpoints in sofia, and make sure you switches/routers >> could handle the PPS loads. >> >> Double that to 2 leg calls and you have 2.86G/sec of RTP (again on the >> wire including UDP, IP, ethernet framing over heads) and pushing 1.5million >> PPS. >> >> Here again this dictates at least 3 gig-e NICs on your FS box. Is this >> possible? I’m sure it is with the right hardware configuration... But lets >> step back a minute and really consider what is going on... >> >> You are now putting approx over $2000/hr of traffic on a very expensive >> box that can and will at some point fail. Why are you doing such a thing? >> You’re better off limiting rates to sub 1000 CPS, (say 500CPS) deploying 3 >> boxes to handle the load in n+1 configuration, and if you lose a box you >> arent losing $2000+ in revenue and cranky customers while you >> repair/replace the now dead machine. >> >> Lets face it, you can purchase a colo a nice machine for say 200 to >> 300$/month (assuming bandwidth costs are the same as above so that’s a wash >> instead of trying to cram it all on box that ends up costing you 1000/month >> by the time you pay for the hardware and colocation expenses for it. >> >> >> >> >> >> On 8/14/14 11:56 AM, "Russell Treleaven" <rtreleaven(a)bunnykick.ca> wrote: >> >> 1000 calls per second... no idea but I really doubt the network interface >> is going to be the bottleneck. >> Not sure why you make a distinction here between sessions and connected. >> To be clear you mean 5000 bridge calls correct? >> Since media is not an issue I think you can just start testing with sipp >> and bwm-ng to see what is what. >> >> 1000 calls per second is a very big number. >> Let us know how you make out. >> >> Sincerely, >> >> Russell Treleaven >> >> >> >> >> >> On Wed, Aug 13, 2014 at 5:40 PM, Andre <andretodd(a)verizon.net> wrote: >> >> 1000 calls per second and 10000 sessions with 4000 connected. What would >> be recommend? >> >> *From:* freeswitch-users-bounces(a)lists.freeswitch.org [ >> mailto:freeswitch-users-bounces@lists.freeswitch.org >> <freeswitch-users-bounces(a)lists.freeswitch.org>] *On Behalf Of *Russell >> Treleaven >> *Sent:* Wednesday, August 13, 2014 5:25 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Network Card >> >> >> >> If the media is bypassing you then you only need bandwidth for the >> signalling. >> >> >> >> Depends primarily on the calls per second. >> >> >> >> bwm-ng will help you to see whats going on. >> >> >> >> >> >> >> >> >> >> On Wed, Aug 13, 2014 at 5:20 PM, Andre <andretodd(a)verizon.net> wrote: >> >> >> >> Hi, is there anything special with the network card to get 10,000 >> sessions in media bypass mode with FreeSwitch? Will a Gig E be enough >> bandwidth? >> >> What would it look like if the network card couldn’t keep up and how can >> I verify the network card is the bottleneck? >> Thanks >> Andre >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www..freeswitchsolutions.com <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 >> >> 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 >> >> >> >> _________________________________________________________________________ >> 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 >> >> 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 >> >> >> ------------------------------ >> >> _________________________________________________________________________ >> 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 >> >> 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 >> >> >> -- >> Ken >> >> >> >> *http://www.FreeSWITCH.org >> <http://www.FreeSWITCH.org>http://www.ClueCon.com <http://www.ClueCon.com> >> http://www.OSTAG.org <http://www.OSTAG.org>*irc.freenode.net #freeswitch >> Twitter: @FreeSWITCH >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www..freeswitchsolutions.com <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 >> >> 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 >> >> > _________________________________________________________________________ > 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 > > 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 > >

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Re: [Freeswitch-users] Network Card
by aademattia＠comcast.net 14 Aug '14

14 Aug '14

Why is 1000 cps obscene? Denovo lab uses nokia stack and we get over 1k cps in a server that cost 2k. Sent from my HTC ----- Reply message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: "Freeswitch-users" <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] Network Card Date: Thu, Aug 14, 2014 6:16 PM We don't give advice on load related questions outside what you have already seen in this thread. 1000cps is obscene. If you are actually getting paid for 1000cps you can surely afford to invest in hardware and professional help. On Aug 14, 2014 3:01 PM, "Andre" <andretodd(a)verizon.net> wrote: Just testing for now, no production traffic. Trying to find the breaking point. The reason for the questions are because I’m testing FS and after a certain point I am getting very high PDD, more than 6 seconds. Per the CDR’s it takes about 6 seconds from start time in CDR Inbound to CDR outbound. I have tried 1 to 10 profiles to see if that makes a difference but it doesn’t and 1 to 4 consoles running one profile each still no difference. Any idea why the CDR’s should show it would take 6 seconds to get from start of inbound to start of outbound? From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ken Rice Sent: Thursday, August 14, 2014 2:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Network Card 1000 CPS the NIC is not the bottle neck this is actually relatively small amount of TCP/UDP traffic for just the SIP message. Now the RTP @ 1000 calls on the other hand assuming G711 at 20ms ptime. And assuming 1 leg calls originating from a dialer would generate 80megabits of RTP (at the IP layer even more on the wire), and approx 50K PPS. Assuming 2 leg’d calls bridging media 160megabit and 100K PPS of just RTP... But then we have to extrapolate peak concurrent from 1000 CPS... Assuming dialer traffic again, 75% ASR and 20 second ACD, and lets assume the 25% are just hard failed for the calculation (no early media on these) that gives us 750 new calls/sec and over the course of 20 seconds we accumulate 15,000 concurrent calls.. So lets take that and extrapolate bandwidth and PPS requirements for single leg’d calls of 1.43Gig/sec (on the wire) and 750K PPS so that’s going to exceed the bandwidth of a GIG-E connection so you’d have to break this into 2 RTP endpoints in sofia, and make sure you switches/routers could handle the PPS loads. Double that to 2 leg calls and you have 2.86G/sec of RTP (again on the wire including UDP, IP, ethernet framing over heads) and pushing 1.5million PPS. Here again this dictates at least 3 gig-e NICs on your FS box. Is this possible? I’m sure it is with the right hardware configuration... But lets step back a minute and really consider what is going on... You are now putting approx over $2000/hr of traffic on a very expensive box that can and will at some point fail. Why are you doing such a thing? You’re better off limiting rates to sub 1000 CPS, (say 500CPS) deploying 3 boxes to handle the load in n+1 configuration, and if you lose a box you arent losing $2000+ in revenue and cranky customers while you repair/replace the now dead machine. Lets face it, you can purchase a colo a nice machine for say 200 to 300$/month (assuming bandwidth costs are the same as above so that’s a wash instead of trying to cram it all on box that ends up costing you 1000/month by the time you pay for the hardware and colocation expenses for it. On 8/14/14 11:56 AM, "Russell Treleaven" <rtreleaven(a)bunnykick.ca> wrote: 1000 calls per second... no idea but I really doubt the network interface is going to be the bottleneck. Not sure why you make a distinction here between sessions and connected. To be clear you mean 5000 bridge calls correct? Since media is not an issue I think you can just start testing with sipp and bwm-ng to see what is what. 1000 calls per second is a very big number. Let us know how you make out. Sincerely, Russell Treleaven On Wed, Aug 13, 2014 at 5:40 PM, Andre <andretodd(a)verizon.net> wrote: 1000 calls per second and 10000 sessions with 4000 connected. What would be recommend? From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Russell Treleaven Sent: Wednesday, August 13, 2014 5:25 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Network Card If the media is bypassing you then you only need bandwidth for the signalling. Depends primarily on the calls per second. bwm-ng will help you to see whats going on. On Wed, Aug 13, 2014 at 5:20 PM, Andre <andretodd(a)verizon.net> wrote: Hi, is there anything special with the network card to get 10,000 sessions in media bypass mode with FreeSwitch? Will a Gig E be enough bandwidth? What would it look like if the network card couldn’t keep up and how can I verify the network card is the bottleneck? Thanks Andre _________________________________________________________________________ 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 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 _________________________________________________________________________ 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 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 _________________________________________________________________________ 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 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH _________________________________________________________________________ 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 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

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How does DNS SRV and SIP OPTIONS logic combine for a gateway?
by Victor Chukalovskiy 14 Aug '14

14 Aug '14

Good day, Anyone know how exactly FS handles a Sofia gateway configured with the combination below? * proxy resolved through DNS SRV; returns 5 records of the same weight * registration disabled * SIP OPTIONS enabled Is it going to send OPTIONS to 5 IPs in round-robin? What if 2 out 5 fail OPTIONS but the rest remain active? Is FS going to intelligently round-robin calls between remaining 3 IPs or it will mark the entire gateway as "down"? Thank you, -Victor

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