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Re: [Freeswitch-users] CPS per mod_sofia profile limited?
by Anthony Minessale 31 Jan '10

31 Jan '10

There is a tradeoff between cps and audio quality and timing accuracy. Try -vm -nocal if you want to mimic 1.0.4 we don't discuss performance here. On Jan 31, 2010 8:56 AM, "Robin Vleij" <robin(a)swip.net> wrote: On 1/31/10 2:57 AM, Michael Jerris wrote: Hi Mike! > For clarification, is it correct that your getting worse numbers for > sustainable cps on SVN the... Yes, 1.04 offered a lot better performance in my test scenario, using the same call scripts. I agree that the scenario isn't really realistic, I should test with at least 30 second calls. > numbers you would get with bypass_media=true instead of proxy_media=true Thought about the same, and funnily enough that didn't help. But in the long list of things I tried I don't remember if I tried it on 1.04, without media proxying. I expect a much higher performance just bypassing the media. I'll try again tomorrow on 1.04 and post the results. > and with neither setting set as well. Also, make sure your logging level > is low and try putting... I check disk performance using iostat and they're on 2% usage. It's a 10k rpm sas disk we're writing to, with ram cache on the raid controller. Also -nosql didn't have any influence at all on performance. > that length of call matters very little. In proxy media or normal mode, > the performance of the ... Exactly. I'll adjust the lengt of the call to 30 or 60 seconds and then try to see how many calls we can have. That's in the end what matters really and not cps on it's own. Have to create a 60 second pcap first. :) > calls). Some other tips. While this extension may be trivial, what else > is there in your dialpl... Nothing, it's a single entry dialplan and that entry is thus on the top. > cause a significant impact. Do you have any of the presence features All presence features disabled. > enabled? These do significantly impact call handling performance even if > your sipp scenarios do... Yep. :) Thanks for the input so far! Maybe after I get some final results I put it on the wiki for future reference to load tests? /robin _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@list...

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CPS per mod_sofia profile limited?
by Robin Vleij 31 Jan '10

31 Jan '10

Hi guys, Doing a bit of testing / benchmarking with FS 1.04 (and 1.05 SVN). Found out some interesting things, I think. The setup is like this: SIPP Client -> FS -> SIPP Server The dialplan is as simple as it gets: <extension name="test-out"> <condition field="network_addr" expression="^130\.244\.190\.150$"/> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="{sip_contact_user=transit}sofia/gateway/${distributor(gwg1)}/$1|sofia/gateway/${distributor(gwg2)}/$1"/> </condition> </extension> For the rest it's running CSV cdr's, commented out all modules I'm not using, etc etc all that I could find on the wiki and the Interwebs. Hardware / OS: from the shelve quadcore Xeon, debian 64-bit, 12GB memory. SIPP is running a 500ms RTP pcap and the other side echos back. I had a few test setups then: 1: FS SVN, 1 sofia profile where the gateways were configured and the server_IP:5060 was used. 2. FS SVN, 2 sofia profiles where the gateways where in an seperate profile (server_IP:5070) and the "customer facing" side was the original profile. 3. FS 1.04, same as above 4. FS 1.04, 4 sofia profiles, distributor to spread load over 2 incoming and 2 outgoing profiles. Now the interesting thing was that under 1 I could go up, almost without any CPU load, to 50cps. As soon as I went over this, calls where handled slower and "ongoing" calls would pile up untill it became really slow. CPU load went to 100% on the FS process (both user and system time). Lots of interupts and context switches. No throughput anymore untill I lower and wait till the "buffer" is empty and FS is keeping up again. Under 2, I was able to increase the CPS to about 100 with the same effect. 3 then went much better, I was able to increase CPS to about 200 cps and response times in SIPP went up slighty untill it just hits some kind of limit and calls are handled slower. 4 is pretty cool. Here I can run 2 sipp clients both doing 150cps to the gateways that are spread on the distribution module, so I spread traffic over 2 profiles. With 300 cps in total, FS is keeping up and I have 30% idle CPU. However, increasing to over 300cps gives problems again, even though I have idle CPU left! All in all, I have a feeling that a single sip profile can't run more than a certain limit untill it gets into some problem. Depending on if I'm running SVN or 1.04 that limit seems to be 50cps or a bit higher. After that limit it starts piling up "ongoing" calls, by taking time to handle them and when that limit gets too high it's too late. All in all really fine, I just set the system wide limit to a little under that "threshold". But when I'm running just UNDER the threshold it's not CPU that's a problem. Theoretically I should be able to run (based on the CPU usage at 300cps) about 400cps. When running at 300 I get SOME failed calls and I see "switch_core_state_machine.c:525 a9a60636-0cea-11df-85a1-09c991f2afc5 Timeout waiting for next instruction in CS_NEW!" in the console. I didn't find much on how people do high cps setups and it feels a bit like a "friday afternoon solution" to run multiple sofia profiles on the same machine in order to max out the system. Maybe I'm missing something and I know it's not an exact science this, but I'm not sure "all is OK" because I'm not slowly getting to a 100% cpu (or disk / network) usage, I hit some kind of limit after which stuff goes wrong. Anyone any input! /Robin

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NAT keep alive
by Airsignal 31 Jan '10

31 Jan '10

Good Evening: I am trying to get my switch to send keep alives to the ata's in the field. <param name="nat-options-ping" value="true"/> seems to be meant for this. where should it go? I can little documentation discussing this... Thanks!

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Inbound sip invite from external gateway
by juan camilo ospina quintero 31 Jan '10

31 Jan '10

hi to all im already do the integration with. Freeswitch sends invite messages to sailfin, in sailfin there is a sip servlet that acts as a proxy, this means it receives the invite from extension1000 and send the invite back to freeswitch at extension 1001, but i get the freeswitch messages go to sailfin, but i dont get freeswitch to understand sailfin messages. there is my configuration for sending messages and for receiving messages In /freeswitch/conf/dialplan/default.xml <extension name="redirect"> <condition field="network_addr" expression="^192\.168\.2\.9$"/> <condition field="destination_number" expression="^1(\d+)$"> <action application="bridge" data="sofia/external/$0@127.0.0.1:5070"/> </condition> </extension> this works fine, it redirects the messages to sailfin in 127.0.0.1 In /freeswitch/conf/dialplan/public.xml <extension name="receive"> <condition field="network_addr" expression="^127\.0\.0\.1$"/> <condition field="destination_number" expression="^1(\d+)$"> <action application="bridge" data="sofia/internal/$0@192.168.2.9:5060"/> </condition> </extension> this doesnt work, i also use <action application="transfer" data="$1001 XML default/> instead <action application="bridge" data="sofia/internal/$0@192.168.2.9:5060"/> but still doesnt work, the invite that sailfin sends appears in the freeswitch console, but the 1001 extension doesnt get it _________________________________________________________________ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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Re: [Freeswitch-users] Multiple DTMF on group_confirm_key
by Anthony Minessale 31 Jan '10

31 Jan '10

If the both answer you can hangup on the one who entered the wrong or no info. The winner is whichever one exits the script first and is not hungup. On Jan 29, 2010 6:43 AM, "lakshmanan ganapathy" <lakindia89(a)gmail.com> wrote: I tested by executing a script. It works great. But a small doubt. Assume that I made a parallel dial using bridge application. Normally, when one party answer the call, other party end will be hanged up. But if I use group_confirm_key=exec and group_confirm_file=perl script.pl, both the end can answer, and call bridged with the person who finished the script first. What I've to do if I need to execute the script only for the person who answer's the call first? On Thu, Jan 28, 2010 at 10:44 PM, Anthony Minessale < anthony.minessale(a)gmail.com> wrote: > > you ... _______________________________________________ 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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Multiple DTMF on group_confirm_key
by lakshmanan ganapathy 31 Jan '10

31 Jan '10

Hi all, I've experimented with group confirm key and group confirm file. It works great. However, I was unable to give multiple DTMF digits to get the confirmation. I've set group_confirm_key=1234, I thought it will ask the 4 digits from the user. But it simply taken 1 and when the user presses 1, the call got bridged. Is there any way to specify multiple dtmf to be confirmed??

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Strategies for reliably detecting nat on B-leg?
by Bill W 31 Jan '10

31 Jan '10

Hey All, I know FreeSWITCH has the agressive-nat-detection parameter for sofia configs which will detect NAT on an incoming call. So we know if the A-leg is natted. The question is, are there any reliable ways to detect nat at the destination before bridging that call? One could assume that if the destination is a gateway, then it would be okay to bypass-media. Also, one could check the sofia registry and assume that if an endpoint is registered to FreeSWITCH that it is NATted and therefore no point in trying to bypass media. But neither of these options seems 100% reliable. Thoughts? Suggestions? Thanks! Bill W.

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problem with hard phone
by jwssr 31 Jan '10

31 Jan '10

Iam having probles with my hard phone...van access ip0020...which is not completing the 'B' leg on incoming calls to it and on outgoing calls....immedialtely issuing a 488... and fs log states ... switch_core_codec.c:537 Codec G722 Exists but not at the desired implementation. 16000hz 20ms 2010-01-30 11:49:07.736157 [ERR] sofia_glue.c:2126 Can't load codec? ... I have multiple soft phones (sflphones extensions 1000-1019), a dlink wifi dph-540 (extension 4001 ip=192.168.1.190), and this van access ip0020 hard phone (extension 2001 ip=192.168.1.180).....fs running on 192.168.1.102. all phones sans '2001" work perfectly..to/fro vitelity and local on lan. Im almost certain that the problem lies in the port number assigned to the uri. wireshark shows that call was cancel because port could not be reached and port number is missing on ua shown by fs cli. Call-ID: af4cfd49b52bf8b3b3ec8db3e8a309e5(a)192.168.1.190 User: 4001(a)192.168.1.102 Contact: 4001 <sip:4001@192.168.1.190:5060> ... ... Call-ID: 4c1ed9ae64405367(a)192.168.1.180 User: 2001(a)192.168.1.102 Contact: "user" <sip:2001@192.168.1.180> ... notice missing port no. here Call-ID: 9be86b04-720f-4f7c-af52-b4f67dccf76b User: 1000(a)192.168.1.102 Contact: "jon" <sip:1000@192.168.1.101:5060> .... wireshark..... 292 14.903431 192.168.1.102 192.168.1.180 ICMP Destination unreachable (Port unreachable) 295 14.927845 192.168.1.102 192.168.1.180 SIP Request: BYE sip:2001@192.168.1.180 I have output from siptrace...but did not want to clutter up forum with too much detail but can provide. I also have screenshots of html config of phone... I would appreciate some help please thanks -- View this message in context: http://old.nabble.com/problem-with-hard-phone-tp27386032p27386032.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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freeswitch with T.38
by Sharad 30 Jan '10

30 Jan '10

Is paid freeswitch available with T.38 support. ? regards Sharad -- View this message in context: http://n2.nabble.com/freeswitch-with-T-38-tp4485495p4485495.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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External Profile Problem
by juan camilo ospina quintero 30 Jan '10

30 Jan '10

Hi, im trying to establish a simple conference using freeswitch and sailfin, sailfin is and application server that works with SipSevlets. the all thing works as follow. two softphone register with freeswitch, extension 1000 and 1001 1000 sends and invite to 1001, this invite goes to sailfin, i use this <extension name="redirect"> <condition field="network_addr" expression="^192\.168\.2\.9$"/> <condition field="destination_number" expression="^1(\d+)$"> <action application="bridge" data="sofia/external/$0@127.0.0.1:5070"/> </condition> </extension> this is in /freeswitch/conf/dialplan/default.xml this far all goes well, the servlet receives the invite, and sends back the invite to freeswitch, i put this <extension name="receive"> <condition field="network_addr" expression="^192\.168\.2\.9$"/> <condition field="destination_number" expression="^1(\d+)$"> <action application="transfer" data="$0 XML default"/> </condition> </extension> in /freeswitch/conf/dialplan/public.xml, but freeswitch returns this ------------------------------------------------------------------------ 2010-01-26 20:14:29.512927 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000(a)192.168.2.9 [5177e93a-0ae1-11df-afc9-db39c681a2f1] 2010-01-26 20:14:29.512927 [NOTICE] sofia.c:3527 Hangup sofia/external/1000(a)192.168.2.9 [CS_NEW] [MANDATORY_IE_MISSING] send 632 bytes to udp/[192.168.2.9]:5070 at 01:14:29.517927: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.153.1:5070;branch=z9hG4bKdaacdd64d693615c451ab9db43f9c71c2626;received=192.168.2.9 From: <sip:1000@192.168.2.9>;tag=g4xfbi12-3 To: <sip:1001@192.168.2.9:5080>;tag=4r91165pvcycB Call-Id: 192.168.153.1_3_3990383226484831353 Cseq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=96;text="MANDATORY_IE_MISSING" Content-Length: 0 ------------------------------------------------------------------------ 2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1086 Session 9 (sofia/external/1000(a)192.168.2.9) Ended 2010-01-26 20:14:29.525646 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000(a)192.168.2.9 [CS_DESTROY] i dont understand why i doesnt work if in public.xml, i tell to transfer the call if extension starts with 1 and the caller ip address is 192.168.2.9 please if someone can help me _________________________________________________________________ Windows Live: Keep your friends up to date with what you do online. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so…

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