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Audio delays from conferences under heavy CPU load on Windows
by Markus von Arx 05 Feb '14

05 Feb '14

Hi I've got a nasty problem with FreeSwitch conferences: after some time, I observe that audio that goes through FreeSwitch conferences gets more and more delayed (by the FreeSwitch conferences). Sometimes it's only 2 seconds, but I have also observerd audio delays up to 5:30 minutes. Audio is never missing, only delayed, and even transmitted correctly after many minutes of delays. More details: - FreeSwitch 1.2.17, 64bit - OS: Windows Server 2012 64bit and Windows 7 64bit (observed on both platforms) - audio codec: G.711 alaw only - external channels: only SIP channels (mod_sofia), RTP with 20ms packets, G.711 alaw, all SIP endpoints in the local LAN on the same Ethernet switch. - dialplan: very very simple; only two actions: answer and connect every incoming/outgoing SIP channel directly to its private conference (mod_conference); and I set and export the following variables on all channels: rtp_disable_vad_in=true, rtp_disable_vad_out=true, send_silence_when_idle=-1, bridge_generate_comfort_noise=-1, suppress_cng=true - my own application: does some logic over mod_event_socket (on localhost); basically connects/disconnects these conferences using loopback channels (mod_loopback) -> simple example: two SIP channels from SIP users 1001 and 1002, dialplan creates two conferences 1001c and 1002c, my application connectes 1001c and 1002c with a loopback channel, audio from SIP user 1001 travels through SIP channel to FreeSwitch conference 1001c, then through loopback channel to conference 1002c, then through SIP channel to SIP user 1002. The audio delay I'm talking about is that what user 1001 speaks into his phone arrives at user 1002 N seconds later instead of almost immediately. - mod_conference config: rate=8000, interval=20, max-members=99, energy-level=0, *-sound="", comfort-noise=0, conference-flags=audio-always - mod_sofia: rtp-timer-name=soft, suppress-cng=true, vad=none - timer: soft timer (no other available on Windows) Usage scenarios: - failing scenario 1 (low load): 34 SIP channels, 34 conferences, 97 loopback channels, 228 FreeSwitch channel objects (34 + 2*97) -> ~24% CPU load -> audio delays only sometimes, a bit unpredictable, mostly none, but sometimes in the region of 2-3 seconds - failing scenario 2 (medium load): ... -> ~51% CPU load -> audio delays start slowly, in the region of 2-10 seconds, grow slowly with time while keeping channels and conferences alive - failing scenario 3 (high load): 66 SIP channels, 66 conferences, 385 loopback channels, 836 FreeSwitch channel objects (66 + 2*385) -> ~74% CPU load -> audio delays start almost immediately: 18 seconds delay after 15 minutes, 41 seconds delay after 3 hours, 5:30 minutes delay after 19 hours. => My guess: it could be that under heavy CPU load, the (soft) timers that are responsible for the RTP audio streams don't get fired reliably anymore but sometimes (often) too late.Theoretically this should not cause an ever increasing delay, as the next timer event after the delayed one should be queued a bit earlier. But still somehow the delay happens and it looks like the mixed RTP packets from the conferences are sent later and later and the delay is never recovered. => Any ideas? Is this a known issue? Maybe some combination of Windows, soft timers, high CPU load, loopback channels, disabled CNG/VAD, long running conferences? Maybe a FreeSwitch (timer) bug? Any help would be appreciated! Thanks and best regards, Markus

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Freeswitch not sending 401 on register
by Stephen Thwaites 05 Feb '14

05 Feb '14

Hello, Registrations on my FS follow this pattern. Register Unauthorised 401 Register with Authorization 200 OK I have recently added another phone in another domain on the same profile and the first Register attempt follows the above pattern but thereafter (every 10 mins) the phone sends a register with the authorization already included, the 401 is not sent. I.e. Register with Authorization 200 OK I haven't modified any ACL, apply-register-acl is commented out in the internal.xml profile. I have studied the ACL page in the wiki, but cannot clarify this behavior. I also cannot find any settings in the sip phone either. Any insights would be appreciated? Small sip trace here, First Register is without 401 and the second with 401! Many thanks! Regards, Steve. ------------------------------------------------------------------------ recv 827 bytes from udp/[80.101.42.120]:58860 at 10:37:08.546282: ------------------------------------------------------------------------ REGISTER sip:voiptest.callstera.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.13:25260;branch=z9hG4bK45463f76b31dc556;rport From: "1002" <sip:1002@voiptest.callstera.com>;tag=e03fcb9ff95bdf71 To: <sip:1002@voiptest.callstera.com> Contact: <sip:1002@80.101.42.120:58860;transport=udp> Supported: path X-Grandstream-PBX: true Authorization: Digest username="1002", realm="voiptest.callstera.com", algorithm=MD5, uri="sip:voiptest.callstera.com", qop=auth, nc=00000010, cnonce="42dc84a07d688734", nonce="809cda5a-8cbe-11e3-8d0c-5dbb90e62c86", response="1c8d0736df9903992397426089525d48" Call-ID: fe877c770cc8d17d(a)192.168.1.13 CSeq: 10011 REGISTER Expires: 60 User-Agent: Grandstream GXP1200 1.2.5.3 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 627 bytes to udp/[80.101.42.120]:58860 at 10:37:08.548713: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.13:25260;branch=z9hG4bK45463f76b31dc556;rport=58860;received=80.101.42.120 From: "1002" <sip:1002@voiptest.callstera.com>;tag=e03fcb9ff95bdf71 To: <sip:1002@voiptest.callstera.com>;tag=gH8etK6gQ81HB Call-ID: fe877c770cc8d17d(a)192.168.1.13 CSeq: 10011 REGISTER Contact: <sip:1002@80.101.42.120:58860;transport=udp>;expires=60 Date: Mon, 03 Feb 2014 10:37:08 GMT User-Agent: Callstera VOIP PBX v1.20 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 511 bytes from udp/[80.101.42.120]:41476 at 10:37:13.918646: ------------------------------------------------------------------------ REGISTER sip:callstera.callstera.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.57:5060;branch=z9hG4bK3aa0c704c1fb893893ca86fc64ba2658;rport From: "Marc" <sip:1000@callstera.callstera.com>;tag=1895422338 To: "Marc" <sip:1000@callstera.callstera.com> Call-ID: 3438668376@192_168_1_57 CSeq: 17927 REGISTER Contact: <sip:1000@192.168.1.57:5060> Max-Forwards: 70 User-Agent: N510 IP PRO/42.075.00.000.000 Expires: 300 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ------------------------------------------------------------------------ 2014-02-03 11:37:13.959442 [WARNING] sofia_reg.c:1634 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1000(a)callstera.callstera.com] from ip 80.101.42.120 send 677 bytes to udp/[80.101.42.120]:41476 at 10:37:13.969082: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.57:5060;branch=z9hG4bK3aa0c704c1fb893893ca86fc64ba2658;rport=41476;received=80.101.42.120 From: "Marc" <sip:1000@callstera.callstera.com>;tag=1895422338 To: "Marc" <sip:1000@callstera.callstera.com>;tag=Ht17UeQmmHr4p Call-ID: 3438668376@192_168_1_57 CSeq: 17927 REGISTER User-Agent: Callstera VOIP PBX v1.20 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="callstera.callstera.com", nonce="25084f5c-8cbf-11e3-8d11-5dbb90e62c86", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 790 bytes from udp/[80.101.42.120]:41476 at 10:37:14.017384: ------------------------------------------------------------------------ REGISTER sip:callstera.callstera.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.57:5060;branch=z9hG4bK4e0ee57b1a907eea7e0c800286896269;rport From: "Marc" <sip:1000@callstera.callstera.com>;tag=1895422338 To: "Marc" <sip:1000@callstera.callstera.com> Call-ID: 3438668376@192_168_1_57 CSeq: 17928 REGISTER Contact: <sip:1000@192.168.1.57:5060> Authorization: Digest username="1000", realm="callstera.callstera.com", qop=auth, algorithm=MD5, uri="sip:callstera.callstera.com", nonce="25084f5c-8cbf-11e3-8d11-5dbb90e62c86", nc=00000001, cnonce="9a8446414e2c5ca357b9b53bc645064e", response="c52337b2e6f18904d0e2a4ec14b44de9" Max-Forwards: 70 User-Agent: N510 IP PRO/42.075.00.000.000 Expires: 300 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 ------------------------------------------------------------------------ send 624 bytes to udp/[80.101.42.120]:41476 at 10:37:14.019683: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.57:5060;branch=z9hG4bK4e0ee57b1a907eea7e0c800286896269;rport=41476;received=80.101.42.120 From: "Marc" <sip:1000@callstera.callstera.com>;tag=1895422338 To: "Marc" <sip:1000@callstera.callstera.com>;tag=j3t0X97QHteQj Call-ID: 3438668376@192_168_1_57 CSeq: 17928 REGISTER Contact: <sip:1000@192.168.1.57:5060>;expires=300 Date: Mon, 03 Feb 2014 10:37:14 GMT User-Agent: Callstera VOIP PBX v1.20 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0

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codec configuration ignored by freeswitch
by Giuseppe Cardone 05 Feb '14

05 Feb '14

Hello, I have tried to configure a limited set of two codecs for outbound calls. I first tried with the parameters described in the documentation in a sip_profile and then in a dialplan and then in vars.xml. However, each time the set of codecs seems to be overwritten and the original list offered by the calling UA is applied. I am new to FS and suspect I am missing something quite simple, can anyone lend a hand in suggesting what would cause the described behaviour ? Here is the essential part of the log showing the absolute_codec_string first applied: EXECUTE sofia/internal/390742850020@217.72.102.128:5070 set(absolute_codec_string=PCMA,G729) 2014-01-29 17:17:40.290655 [DEBUG] mod_dptools.c:1402 sofia/internal/390742850020@217.72.102.128:5070 SET [absolute_codec_string]=[PCMA,G729] EXECUTE sofia/internal/390742850020@217.72.102.128:5070 bridge(sofia/gateway/unidata_dev/+390144750000) 2014-01-29 17:17:40.290655 [DEBUG] switch_channel.c:1198 sofia/internal/390742850020@217.72.102.128:5070 EXPORTING[export_vars] [RFC2822_DATE]=[Wed, 29 Jan 2014 17:17:40 +0100] to event 2014-01-29 17:17:40.290655 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables 2014-01-29 17:17:40.290655 [NOTICE] switch_channel.c:1052 New Channel sofia/external/+390144750000 [e00a6dde-8900-11e3-8b08-0b2b96c9cf60] 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:5231 (sofia/external/+390144750000) State Change CS_NEW -> CS_INIT 2014-01-29 17:17:40.290655 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/+390144750000 [BREAK] and then overwritten: 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:5301 [zrtp_passthru] Setting a-leg inherit_codec=true 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:5304 [zrtp_passthru] Setting b-leg absolute_codec_string='G722@8000h@20i@64000b,PCMU@8000h@20i@64000b,PCMA@8000h@20i@64000b,GSM@8000h@20i@13200b' 2014-01-29 17:17:40.290655 [DEBUG] switch_core_state_machine.c:415 (sofia/external/+390144750000) Running State Change CS_INIT 2014-01-29 17:17:40.290655 [DEBUG] switch_core_state_machine.c:454 (sofia/external/+390144750000) State INIT 2014-01-29 17:17:40.290655 [DEBUG] mod_sofia.c:87 sofia/external/+390144750000 SOFIA INIT 2014-01-29 17:17:40.290655 [DEBUG] sofia_glue.c:2739 Local SDP: Thanks in advance for your help. Regards, Giuseppe

2 3

copy variables from a-leg to b-leg
by mbo 05 Feb '14

05 Feb '14

When I bridge a call and I want to copy all variables from the a-leg to b-leg of a call I can use the „export“ function, and all variables are copied automatically. To implement call queueing I use another approach. On an incoming call I set some channel variables and then park the call. After that, I use originate to call (one or more) targets and in case a target gets answered I use uuid_bridge to connect both call legs together. The point is that the „export“ function does not work in that scenario. Is there another way to copy channel variables from one call leg to another? Thanks Markus

3 2

Asterisk / Proxy / Freeswitch reinvites race condition
by dotnetdub 04 Feb '14

04 Feb '14

Hi All, We are facing an issue where we have a customer using asterisk which insists on sending connected line updates via SIP UPDATES and REINVITES. I have tried to mitigate this on the customer Asterisk by using all the config instructions you would expect to work but there are still some scenarios where asterisk insists on sending REINVITES with caller ID update to the trunk. I've also tried a number of things on the freeswitch side to try and persuade Asterisk not to send UPDATES or REINVITES without success... SIP Dialog always starts off well but quickly descends into thousands upon thousands of UPDATES and INVITES Freeswitch sends OK to every UPDATE and an OK to the first REINVITE but then starts sending 500 . Asterisk does ACK these errors but immediately sends another UPDATE (again FS sends ok) and INVITE which again we send back a 500 - you can see where I'm going here. This can end up with 1000s of UPDATE,INVITE,500 going on in a SIP dialog and customer reports it is affecting the call audio while this is happening.. I still am unsure of why the audio is affected, the asterisk invites always have SDP so maybe asterisk is opening/closing the port very quickly or something... I've attached a pastie of the first 48 packets of the SIP Trace. This particular call has 2067 SIP packets , mostly INVITE, 500 Olle (OEJ) comments on the Kamailio mailing list: 'If freeswitch believes it already has an open INVITE transaction it should not respond with 500, it should respond with 491 request pending. In that case Asterisk will back off and retry. Please check with the FreeSwitch people and file a bug report so that they can fix this issue. That's the long term solution, all the rest is just quick and dirty fixes. Seems like if this problem is still around, no one filed a bug report.' Should I file a bug report or what are freeswitch users / dev thoughts on this? Maybe there is something I can do in config or is this something that I must address with Asterisk... With older versions that some customers use I was able to stop reinvites completely. With version 11 this functionality is still documented but doesn't work so well... traces are http://pastebin.freeswitch.org/21924 truncated after 48 packets. Many Thanks, Brian

3 6

Using park_timeout with uuid_park
by Siri MM 04 Feb '14

04 Feb '14

Hi, I have a lua script which parks a leg using uuid_park I would like to restrict the amount of time that the leg is parked. In order to achieve this, I tried using park_timeout as follows: cmd = "uuid_setvar "..uuid.." park_timeout 10"; api:executeString(cmd); api:execute("uuid_park", uuid); However, the parked call doesn't seem to timeout. Am I missing something? Thanks!

2 9

mod_directory
by fs 04 Feb '14

04 Feb '14

I am trying to compile the lastest git to include mod_directory but I cannot get it to compile. I have uncommented 'applications/mod_directory' in '/build/modules.conf.in' but it still does not compile. Is there something else I need to compile this. fs(a)voice2net.ca Darcy Primrose 613-704-1248 voice2net.ca Prescott, ON K0E 1T0

2 3

setting VAD & TALK / NOTALK Events on ongoing call
by Gerald Weber 03 Feb '14

03 Feb '14

Hi, is it possible to enable VAD to get TALK / NOTALK events for an ongoing call, i.e. switch_channel_set_variable(channel, "rtp_enable_vad_in", "true"); switch_channel_set_variable(channel, "rtp_enable_vad_out", "true"); switch_channel_set_variable(channel, "fire_talk_events", "true"); switch_channel_set_variable(channel, "fire_not_talk_events", "true"); Should this work ? Is there another way instead of settings channel vars ? Thanks in advance. gw

1 0

Re: [Freeswitch-users] Play IVR Before Answer
by Shahzad Bhatti 03 Feb '14

03 Feb '14

Anthony Minessale and Mario G thanks for your reply! Anthony Minessale, as u say that provider does not let me to do that but i have no provider in between as i am a soft-phone eyeBeam or xlite that directly registered to FreeSwitch and i try using pre_answer also with playback but these wouldn't work for me and if i answer the call and then playback then it works my syntax for playback is as <action application="playback" data="{timeout=20000}/usr/local/sounds/1.wav" /> Regards Shahzad Bhatti > ---------- Forwarded message ---------- > From: Mario G <mario_fs(a)mgtech.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 1 Feb 2014 09:29:17 -0800 > Subject: Re: [Freeswitch-users] Play IVR Before Answer > Good point, I forgot to mention that. An example: when I was using > Voip-your-life custom early media worked, after about 18 months, V-Y-L > "upgraded" their system and custom early media stopped working. After > days/nights on the phone with them I got them to put me back on their old > system. I also was using Callcentric which always supported custom early > media and I still use them, I dropped V-Y-L. Contact your ITSP and let them > know. > Mario G > > On Jan 31, 2014, at 7:36 AM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > > If you see it saying its playing early media and you can't hear it, that > means the provider does not let you do it. > Just adding playback as the first thing in your dialplan would > automatically invoke playback in early media so if that doesn't work you > are out of luck. > > > On Fri, Jan 31, 2014 at 5:41 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com>wrote: > >> i observe on my phone that this is possible as when i on my mobile when >> bill is not paid and outgoing is blocked then when try to call some number >> then Message is played and call hangup. just like above Scenario i want to >> play message using IVR or may with some other application to tell the >> caller what is his current credit on leg A and then bridge call to leg B >> without answer. >> >> any help will be appreciated >> >> you can get the log of cli here http://pastebin.freeswitch.org/21912 >> and example of dialplan here http://pastebin.freeswitch.org/21913 >> >> Regards >> Shahzad Bhatti >> >> >>> ---------- Forwarded message ---------- >>> From: Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> >>> To: freeswitch-users(a)lists.freeswitch.org >>> Cc: >>> Date: Wed, 29 Jan 2014 18:13:34 +0500 >>> Subject: Re: [Freeswitch-users] Play IVR Before Answer >>> Thanks Mario G for your reply i worked through it using ringback and >>> play file using local stream and find that freeswitch send early media but >>> unable to listen on leg A so i remove the line of answer and play a sample >>> ivr for test and also remove bridge the call to leg B for testing and found >>> that during early media state ivr is playing and log show that it plays but >>> on leg A i am unable to listen anything from that ivr but a ring tone and >>> when the ivr completed the channel state is returned to hangup >>> so i think i am near but missing some thing in configuration so anyone >>> can reply that what i do to listen ivr before answer through early media >>> here is my new log >>> http://pastebin.freeswitch.org/21912 >>> and here is my dialplan for the above call >>> http://pastebin.freeswitch.org/21913 >>> Regards >>> Shahzad Bhatti >>> >>> >>> ---------- Forwarded message ---------- >>>> From: Mario G <mario_fs(a)mgtech.com> >>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>> Cc: >>>> Date: Tue, 28 Jan 2014 13:08:24 -0800 >>>> Subject: Re: [Freeswitch-users] Play IVR Before Answer >>>> I don't know how to do IVR this way but... this may help you get started: >>>> What you want is to dynamically create early media. Our calls use early >>>> media to say hello "name" to the person and change instructions and the >>>> music in early media. This is done by setting the "ringback" variable. >>>> Before the bridge I call a LUA program that looks up the incoming number >>>> and creates the early media string and sets the "ringback" variable before >>>> returning (it does many other things as well). If all you want to do when >>>> someone calls in is to say "Hi Mario, you have a credit of 34.64", it can >>>> be done with a little work. This is NOT IVR though. >>>> >>>> Before messing with IVR note this IMPORTANT FACT: Most (all?) ITSPs >>>> limit how long they will ring (send early media) before doing something >>>> else, in my case the max I can set ringing to is 60 seconds before they >>>> send to VM, etc on their end. That means I am limited to early media of 60 >>>> seconds which my LUA program dynamically calculates and ensures we don't go >>>> past 55 seconds. >>>> Good luck >>>> Mario G >>>> >>>> On Jan 28, 2014, at 8:26 AM, Shahzad Bhatti <shahzad.bhatti(a)g-r-v.com> >>>> wrote: >>>> >>>> Hello i want to play an ivr using phrase to say the caller credit, i do >>>>> this after answer and it is working i want to know that is it also possible >>>>> to play IVR before answer using execute_on_preanswer like >>>>> >>>> *<action application="set" data="execute_on_preanswer=phrase >>>> saycredit,34:64:97" />* >>>> here is my xml_curl xml for the call >>>> http://pastebin.freeswitch.org/21909 >>>> i want to know is it possible and if possible then what is the correct >>>> way i want to play IVR on legA but before answering the leg >>>> >>>> Regards >>>> Shahzad Bhatti >>>> >>>>

2 1

Trouble getting Caller ID to Change
by Chris Tunbridge 03 Feb '14

03 Feb '14

I am trying to get the caller id to change on the channel when it comes on to label it depending on the DID it came in on, my goal is to prefix the caller id, say i have a DID that is in a Phone book, and i want to prefix "PB:" to the caller id (without the quotes) in order to know which DID this call came in on before answering the call. Version: FreeSWITCH Version 1.5.8b+git~20140202T113337Z~9059fb91cc~64bit (git 9059fb9 2014-02-02 11:33:37Z 64bit) INFO output: http://pastebin.freeswitch.com/21926 lua script (with attempts) to change this attribute: http://pastebin.freeswitch.com/21928 get_variable (and set_variable) are references to the session:setVariable and session:getVariable commands. In my case, i can clearly see if i output the value of caller_id_name, that its changed, if i turn ignore_display_updates to false (or not set it at all) i can see in my dial plan that its updating the caller id, and erasing the Prefix i added to it. (well ignoring my set CID is probably a better way of putting it). Regardless of what variables are set on the incoming call/channel the extension shows the original caller id. I apologize if I'm doing something incredibly stupid, or missing something small, but I've spend a solid 8 hours trying to work through this. Calls are handled via a lua script on the inbound per instructions found here: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration#dialplan Which is why I'm using an actions table to do things. This is using lua and MySQL (odbc) to handle a large number of look ups on the fly. If you need anything more from me, let me know.

1 0