- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch - Performance issues
by Shaun Stokes 07 Sep '15

07 Sep '15

Hi, We're experiencing performance issues with FreeSwitch, our target is 500 concurrent sessions, but at the moment this starts to bottleneck around 30. Host system: Windows Server 2012 Hyper-V Host AMD Opteron 4386 (2 processors) - 16 cores total 128GB DDR3 2TB RAID 5 (700MB/s tested read and write throughput) FreeSwitch Virtual Machine: FreeSwitch 1.4.15 Ubuntu 12.04 LTS 16 Virtual cores (high priority) 2GB RAM (would assign more but FreeSwitch never seems to use much) 500GB HD (on VHDX) After around 30 concurrent sessions we begin to see CPU spikes almost every time a new call comes in, as the sessions increase the size and frequency of the CPU spikes also increase. The system seems to be able to sit comfortably with over 100 concurrent sessions and 80% idle CPU, providing we don't have any new calls hitting the platform. The spikes are causing audio (RTP) to stutter or in some cases drop completely for a few seconds. The FreeSwitch spikes are occurring on all 16 cores, we have been monitoring the system using htop and mpstat. This is an example of when we receive an inbound call while we have 34 concurrent sessions: 14:26:04 CPU %usr %nice %sys %iowait %irq %soft %steal %guest %idle 14:26:05 all 1.67 0.00 27.34 0.00 0.25 0.00 0.00 0.00 70.74 14:26:05 0 1.00 0.00 28.00 0.00 0.00 0.00 0.00 0.00 71.00 14:26:05 1 0.99 0.00 26.73 0.00 0.00 0.00 0.00 0.00 72.28 14:26:05 2 0.99 0.00 26.73 0.00 0.00 0.00 0.00 0.00 72.28 14:26:05 3 3.92 0.00 26.47 0.00 0.00 0.00 0.00 0.00 69.61 14:26:05 4 0.99 0.00 25.74 0.00 1.98 0.00 0.00 0.00 71.29 14:26:05 5 0.99 0.00 26.73 0.00 0.99 0.00 0.00 0.00 71.29 14:26:05 6 0.00 0.00 26.47 0.00 0.98 0.00 0.00 0.00 72.55 14:26:05 7 0.00 0.00 26.73 0.00 0.00 0.00 0.00 0.00 73.27 14:26:05 8 3.96 0.00 27.72 0.00 0.00 0.00 0.00 0.00 68.32 14:26:05 9 2.00 0.00 27.00 0.00 0.00 0.00 0.00 0.00 71.00 14:26:05 10 6.00 0.00 32.00 0.00 0.00 0.00 0.00 0.00 62.00 14:26:05 11 1.96 0.00 27.45 0.00 0.00 0.00 0.00 0.00 70.59 14:26:05 12 0.99 0.00 26.73 0.00 0.00 0.00 0.00 0.00 72.28 14:26:05 13 1.00 0.00 28.00 0.00 0.00 0.00 0.00 0.00 71.00 14:26:05 14 0.00 0.00 27.00 0.00 0.00 0.00 0.00 0.00 73.00 This is when the system is not receiving an inbound call but is sitting comfortably at 34 concurrent sessions: 14:25:57 CPU %usr %nice %sys %iowait %irq %soft %steal %guest %idle 14:25:58 all 0.87 0.00 0.62 0.00 0.12 0.00 0.00 0.00 98.39 14:25:58 0 0.99 0.00 0.99 0.00 0.00 0.00 0.00 0.00 98.02 14:25:58 1 0.99 0.00 0.99 0.00 0.00 0.00 0.00 0.00 98.02 14:25:58 2 0.98 0.00 0.98 0.00 0.00 0.00 0.00 0.00 98.04 14:25:58 3 0.00 0.00 0.99 0.00 0.00 0.00 0.00 0.00 99.01 14:25:58 4 0.00 0.00 0.00 0.00 1.98 0.00 0.00 0.00 98.02 14:25:58 5 0.00 0.00 0.00 0.00 0.99 0.00 0.00 0.00 99.01 14:25:58 6 0.00 0.00 0.99 0.00 0.00 0.00 0.00 0.00 99.01 14:25:58 7 0.00 0.00 0.99 0.00 0.00 0.00 0.00 0.00 99.01 14:25:58 8 5.00 0.00 1.00 0.00 0.00 0.00 0.00 0.00 94.00 14:25:58 9 1.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 14:25:58 10 1.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 99.00 14:25:58 11 0.99 0.00 0.99 0.00 0.00 0.00 0.00 0.00 98.02 14:25:58 12 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 14:25:58 13 1.98 0.00 0.99 0.00 0.00 0.00 0.00 0.00 97.03 14:25:58 14 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 14:25:58 15 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 100.00 We're using FreeSwitch in multi-tenant mode, we have tried dedicating a single VM for one tenancy but still experience the issue. My assumption has been that this will be due to the dialplan, I've optimized this slightly by writing LUA scripts to handle some of the inbound calls this appears to take away some of the load but we're still using the internal dialplans for each tenancy (tenancy dialplans have an average of 300 entries). We are using the following arguments when running FreeSwitch: -rp -nc -nonat I've seen another post which says we should operate FreeSwitch in High Performance mode using -hp but can't find anything information about this, is this still a valid argument to use with FreeSwitch? Has anyone experienced similar performance issues before or have any advice? Hope someone may be able help. Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

13 26

mod_rtmp codec
by Paul Cuttler 07 Sep '15

07 Sep '15

It looks like mod_rtmp has the codec hard-coded to Speex. I was wondering if there was issues with getting other codecs to work. We'd really like to use PCMA/U. I've tried changing a couple of lines of code where the codec is set and it works okay in the incoming (play) direction, but is terribly distorted or just absent in the outgoing (publish) direction. Could someone please advise whether it's possible to use other codecs, and where I'm best to get started. thanks, Paul

2 2

Bridging to external SIP gateway doesn't seems to be called...
by Michael Nielsen 05 Sep '15

05 Sep '15

I'm running FreeSWITCH 1.4 on Debian 7. I've hooked up my FreeSWITCH to an external SIP gateway. The following scenarios work: FS -> FS SIP GW -> FS But the following scenario does not work: FS -> SIP GW I only have one single dial plan in my setup: <include> <extension name="test-dialplan"> <condition field="destination_number" expression="^\+(\d{3,20})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name}"/> <action application="bridge" data="user/${dialed_extension}@ ${domain_name},sofia/gateway/sip-outbound/+${dialed_extension}"/> </condition> </extension> </include> My users are named the same as real GSM numbers (country code and number). I would like my FS to call internally if the user exists, if not, then route it to my SIP gateway for connecting to the GSM world. As mentioned, everything works except outgoing calls to the SIP gateway. What goes wrong?

1 2

Video gets garbled/corrupt after a few seconds
by Ankhit Vivekananda 04 Sep '15

04 Sep '15

Hello, I am new to using FreeSwitch. Recently, we started implementing FreeSwitch to place video conference calls between VoIP cameras. The ca,eras call into a conference room on FreeSwitch. The audio is fine during the call, but the video gets garbled after a few seconds. The call is based on SIP protocol. Are there any settings that can be configured for video to function properly. Presently set video parameters in the autoload_configs/conference.conf.xml file are: <param name="conference-flags" value="rfc-4579|livearray-sync|minimize-video-encoding"/> <param name="video-mode" value="mux"/> <param name="video-layout-name" value="2x2"/> <param name="video-layout-name" value="group:grid"/> <param name="video-canvas-size" value="1280x720"/> <param name="video-canvas-bgcolor" value="#333333"/> <param name="video-layout-bgcolor" value="#000000"/> <param name="video-codec-bandwidth" value="1mb"/> <param name="video-fps" value="15"/> Appreciate any help! Thanks, Ankhit

4 8

ACL Routing with multiple conditions.
by Tim King 04 Sep '15

04 Sep '15

I'm certain I am not the first one to try this, but I have been digging through forums and googling for 2 days now without success. I am trying to route toll free traffic based on the source ip's. I am using ACL's to match on the source IP. If i only use one of them it works, btu when I try to add the second no calls ever make it to the second condition. Here are the ways I have tried <extension name="tollfree_to_trunk1"> <condition field="${acl(${network_addr} trunk1)}" expression="true"/> <condition field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/external/$2@1.1.1.1:5060"/> </condition> </extension> <extension name="tollfree_to_trunk2"> <condition field="${acl(${network_addr} trunk2)}" expression="true"/> <condition field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="sofia/external/$2@2.2.2.2:5060"/> </condition> </extension> The second is using the regex function. <extension name="tollfree_to_trunks"> <condition regex="all"> <regex field="destination_number" expression="^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$"> <regex field="${acl(${network_addr} trunk1)}" expression="true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="set" data="sourceip=${network_addr}"/> <action application="bridge" data="sofia/external/$2@1.1.1.1:5060"/> <regex field="${acl(${network_addr} trunk2)}" expression="true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="set" data="sourceip=${network_addr}"/> <action application="bridge" data="sofia/external/$2@2.2.2.2:5060"/> </condition> </extension> Again in summary I am just trying to send toll free traffic to different gateways based on an acl match of the ip the call originates from. any help would be greatly appreciated.

1 0

ASTPP Version 2.2 is Released
by ASTPP Opensource VOIP Billing 04 Sep '15

04 Sep '15

Hi All, Hope you all are enjoying using Version 2.1 which was just released few days back. We highly appreciate your suggestion/feedback on it. Based on that and bugs found in it, we have made some quick updates and few improvements including bug fixes. Its been included in this latest version 2.2 *The list goes here:* 1. Resolved Paypal enable/disable issues #0000060 2. Corrected validations for state field in accounts module. #0000051 3. Resolved batch update issues in rates module 4. Improved footer layout 5. Improved error messages 6. Resolved dashboard graph issue for display decimal value 7. Resolved global DID number translation not accepting blank value issue 8. SIP Devices and Opensips Devices issues fixed 9. Resolved duplicate email address issues 10. Resolved button related issues 11. Accounts DID module validation issues 12. Resolved dialplan errors when number starting with + 13. Improved installation script 14. Resolved SIP device issues in customer edit 15. Resolved DID destination saving issue in reseller portal *To install v2.2:* # wget --no-check-certificate https://goo.gl/ETmAJl -O install.sh # chmod +x install.sh # ./install.sh *Upgrade steps from v2.1 to v2.2:* # wget --no-check-certificate https://goo.gl/RIuIej -O install.sh # chmod +x install.sh # ./install.sh *Want to get more out of ASTPP?* Empower your ASTPP with FM Addon: http://www.astpp.org/freeswitch-monitoring-addon In case if you find any issues, then please feel free to report in our bug tracker below: http://bugs.astpp.org/ Thank you everyone for your continuous support in ASTPP journey. Please donate at our paypal address donation(a)astpp.org to keep the project running smoothly. -- Regards, ASTPP Team iNextrix Technologies Pvt. Ltd.

1 0

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 04 Sep '15

04 Sep '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

408 error code
by Fernando Flórez 04 Sep '15

04 Sep '15

Hello, We are experiencing an issue with one user account. He is receiving a 408 error code. There are almost 50 other users on that box and the issue is only occurs with that account. We have tried switchinf phones, tried different networks and reconfiguring his account from scratch and no go. Any hints on how to investigate whats happening? Thanks,

3 4

Mobile SIP client supporting seamless 3G to WiFi switchover?
by Victor Chukalovskiy 03 Sep '15

03 Sep '15

Good day, anyone knows if there is a SIP client out there capable of switching between 3G and WiFi during a call? ... seamless or nearly seamless

3 3

RFC 3263 and failing over
by Joshua Gigg 03 Sep '15

03 Sep '15

I've setup some SRV records to test failover between a few servers. $ dig _sip._udp.testing-sip.giggsey.net SRV ;; ANSWER SECTION: _sip._udp.testing-sip.giggsey.net. 371 IN SRV 10 0 5060 testing-sip-real.giggsey.net. _sip._udp.testing-sip.giggsey.net. 371 IN SRV 0 0 5080 testing-sip-fake.giggsey.net. testing-sip-fake is a FreeSWITCH box, configured to hangup every inbound call with 503: NETWORK_OUT_OF_ORDER. testing-sip-real will accept traffic from this server, and actually connect to something. When I make a call via FreeSWITCH (latest master), it sends the INVITE to testing-sip-fake, which responds correctly. FreeSWITCH does not ACK the 503, but sends the INVITE to testing-sip-real. testing-sip-real responds with a 200, but FreeSWITCH sends the ACK to testing-sip-fake. Trace from sngrep below (apologies for any formatting issues): http://i.imgur.com/aGOBHpp.png │ 193.104.89.40:5080 178.62.61.192:5080 193.104.89.57:5060 │ ──────────┬───────── ──────────┬───────── ──────────┬───────── │ ▒ 17:06:29.054095 │ INVITE (SDP) │ │ │ ▒ +0.016957 │ ──────────────────────────> │ │ │ ▒ 17:06:29.071052 │ 100 Trying │ │ │ ▒ +0.004131 │ <────────────────────────── │ │ │ ▒ 17:06:29.075183 │ 503 Service Unavailable │ │ │ ▒ +0.005387 │ <────────────────────────── │ │ │ ▒ 17:06:29.080570 │ INVITE (SDP) │ │ ▒ +0.003829 │ ────────────────────────────────────────────────────────> │ │ ▒ 17:06:29.084399 │ 100 Giving a try │ │ ▒ +0.491453 │ <──────────────────────────────────────────────────────── │ │ ▒ 17:06:29.575852 │ 503 Service Unavailable │ │ │ ▒ +1.000255 │ <────────────────────────── │ │ │ ▒ 17:06:30.576107 │ 503 Service Unavailable │ │ │ ▒ +2.000364 │ <<<──────────────────────── │ │ │ ▒ 17:06:32.576471 │ 503 Service Unavailable │ │ │ ▒ +2.662937 │ <<<──────────────────────── │ │ │ ▒ 17:06:35.239408 │ 183 Session Progress (SDP) │ │ ▒ +1.155243 │ <──────────────────────────────────────────────────────── │ │ ▒ 17:06:36.394651 │ 183 Session Progress (SDP) │ │ ▒ +0.093683 │ <<<────────────────────────────────────────────────────── │ │ ▒ 17:06:36.488334 │ 180 Ringing (SDP) │ │ ▒ +0.088306 │ <──────────────────────────────────────────────────────── │ │ ▒ 17:06:36.576640 │ 503 Service Unavailable │ │ │ ▒ +1.936790 │ <────────────────────────── │ │ │ ▒ 17:06:38.513430 │ 200 OK (SDP) │ │ ▒ +0.010268 │ <──────────────────────────────────────────────────────── │ │ │ 17:06:38.523698 │ ACK │ │ │ │ +0.489731 │ ──────────────────────────> │ │ │ │ 17:06:39.013429 │ 200 OK (SDP) │ │ │ +0.000296 │ <──────────────────────────────────────────────────────── │ │ │ 17:06:39.013725 │ ACK │ │ │ │ +0.999795 │ ──────────────────────────> │ │ │ │ 17:06:40.013520 │ 200 OK (SDP) │ │ │ +0.000282 │ <──────────────────────────────────────────────────────── │ │ │ 17:06:40.013802 │ ACK │ │ │ │ +0.562804 │ ──────────────────────────> │ │ │ │ 17:06:40.576606 │ 503 Service Unavailable │ │ │ │ +1.436391 │ <────────────────────────── │ │ │ │ 17:06:42.012997 │ 200 OK (SDP) │ │ │ +0.000149 │ <──────────────────────────────────────────────────────── │ │ │ 17:06:42.013146 │ ACK │ │ │ │ +2.564912 │ ──────────────────────────> │ │ │ │ 17:06:44.578058 │ 503 Service Unavailable │ │ │ │ +0.685651 │ <────────────────────────── │ │ │ │ 17:06:45.263709 │ BYE │ │ │ │ +0.016618 │ ──────────────────────────> │ │ │ │ 17:06:45.280327 │ 481 Call Does Not Exist │ │ │ │ +0.730549 │ <────────────────────────── │ │ │ │ Any ideas on where to start to debug? -- Joshua Gigg

3 6