- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch does not free memory
by Eduardo Alonso 07 May '18

07 May '18

Hi list: Some time ago I openned an issue: https://freeswitch.org/jira/browse/FS-11000, related to a problem with the memory.This problem seems to be related to the RTP read packets proccess. When a call reached the 3 hours of conversation aprox. FS begin to catch memory constantly. Someone sufffered this behaviour? If I restart FS the memory si freeing. You can to obtain more info in the Jira issue. Thanks in advance. -- *Eduardo Alonso Gil* Operations Technical Lead @ Quobis <http://www.quobis.com/> | e: eduardo.alonso(a)quobis.com | t: +34902 999 465 *NEW ADDRESS: **P.I A Granxa-**Rua D-Paralela 1-"Casa de Pedra"-**36400 O Porriño*

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How to playbak a file in a conference?
by Babak Yakhchali 05 May '18

05 May '18

Hi I'm using the following lua script to add some users taken as argument to a conference: session:execute('answer'); session:setVariable('api_hangup_hook','conference paging kick all'); --kick out all page devices on hangup session:setVariable('conference_auto_outcall_timeout','5'); --?? session:setVariable('conference_auto_outcall_flags','none') --you can mute page devices session:setVariable('conference_auto_outcall_announce','tone_stream://%(500,200,800);loops=3') for i,v in ipairs(argv) do session:execute('conference_set_auto_outcall','user/'..v) end session:execute('conference','paging@default'); this is working fine and all members are connected to conference. now I need to start playing a file into conference. currently I'm using caller-controls part of conference.conf.xml to play the file when a caller presses the 0 key: ... <caller-controls> <group name="default"> <control action="execute_application" digits="0" data="playback applications/2/test.mp3"/> ... </group> </caller-controls> .... now when each member presses 0 the file is played but just to the member which pressed the 0 key. how can I make it to play the file for all conference members?

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FSBlue / FreeSWITCH 1.8
by Markus Bönke 04 May '18

04 May '18

Hello, I just saw that the freeswitch team is offering now a hosting service FSBlue (https://freeswitch.com/index.php/fs-blue/ <https://freeswitch.com/index.php/fs-blue/>). In the description it is mentioned that the manages instances are based on Freeswitch 1.8. So is there now a commercial, closed source branch for Freeswitch 1.8 and a community edition based on the 1.6 branch? Thanks and regards Markus

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WebRTC using rtp_sdes_suites=AES_CM_128_HMAC_SHA1_80
by Jerry Chinn 04 May '18

04 May '18

Good Day, Running FS 1.6.17 on CentOS 7.4 We are running WebRTC and are required to use AEAD_AES_256_GCM_8 or AEAD_AES_128_GCM_8 for security. I have eliminated all of the options in the vars file except rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8. Calls are successfully completing, however, in debug we are seeing AES_CM_128_HMAC_SHA1_80 as the sdes suite for srtp:dtls. 2018-05-04 22:38:30.429310 [INFO] switch_rtp.c:3185 Changing audio DTLS state from HANDSHAKE to SETUP 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3094 audio Fingerprint Verified. 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3908 Activating audio Secure RTP SEND 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3886 Activating audio Secure RTP RECV 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3134 Changing audio DTLS state from SETUP to READY 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2018-05-04 22:38:30.450549 [DEBUG] switch_rtp.c:1885 rtcp_stats_init: audio ssrc[3910337773] base_seq[2433] Any ideas on how or where to change this to the desired encryption protocol? Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com<http://www.thenavisway.com/> Facebook<https://www.facebook.com/theNAVISway/> | Twitter<https://twitter.com/NAVISway> | LinkedIn<https://www.linkedin.com/company/navisway> | Blog<https://www.thenavisway.com/blog>

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In proxy_media mode, RTCP is not proxied or activated
by Stefano Bertuola 03 May '18

03 May '18

Hi all. On February last year there was a discussion related to proxy_media mode and RTCP and I understood this is not actually supported. I am actually facing the same issue and I would like to understand if there is any way to get it working; since RTP is using an even port and RTCP the following one (RTP port + 1), is there any way to get both ports proxied? Br. Stefano

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Re: [Freeswitch-users] youtube broadcast from freeswitch audio quality
by Charlie DeTar 03 May '18

03 May '18

Any chance you could share pertinent details of the freeswitch config you use when broadcasting ClueCon to youtube? I'd love to compare differences with the mod_conference config I posted below to see if there's something obvious we're missing. Also, apart from conference.conf.xml, are there other confs that would be involved with determining details of the audio encoding for record-to-rtmp? Recording to a local mp4 file works flawlessly (just as broadcasting rtmp to UStream does). Bandwidth shouldn't be an issue, this is indeed running on MIT's fairly beefy networking pipes. It has been some time; we're still struggling to find an adequate multiparty video broadcast solution. We've ditched nginx-rtmp-module (it's been abandoned by its author) and are now attempting to use youtube's broadcast infrastructure instead. Because our multiparty video system has been so buggy, most of our users just use a single-point broadcast for real-time parts. But we'd really like to get multiparty working more smoothly. The project: https://unhangout.media.mit.edu best, Charlie On Wed, May 2, 2018, at 12:55 PM, Anthony Minessale wrote: > mod_rtmp is not related to recording over rtmp its a legacy endpoint > for flash audio embedded in a browser.> > We broadcast ClueCon weekly on our youtube page every week and we have > no audio issues but we do make sure we have at least 4-5mb stream for > 1080p quality.> https://www.youtube.com/watch?v=gJqkld_C0dk > > Maybe try recording to an mp4 and compare. > > BTW, I haven't heard from you in 2 years! Are you still doing the > MIT thing? Ping me offline maybe.> > > > On Tue, May 1, 2018 at 9:10 PM, Charlie DeTar <cfd-fs(a)fohn.org> wrote:>> Hi, >> >> I'm trying to broadcast multiparty audio and video through youtube, >> using this stack:>> >> Browser (Chrome) -> Custom webapp using Verto -> mod_conference -> >> record to rtmp -> youtube live>> >> However, I'm getting poor audio quality on youtube in the live >> broadcast. Here's an example (this video is screen-capped from the >> live player, because youtube seems to clean up the recorded file >> after the fact. I've tested with multiple computers/browsers that >> this is how it sounds to everyone during the live broadcast.):>> https://youtu.be/bO8t78YXxjc >> >> That is using the "Low Latency" setting in youtube. The >> conference.conf.xml "interval" parameter seems to also have an >> effect; I have it set to 10 here as that seemed to get fewer pops >> than the default 20.>> >> Using youtube's "Ultra-low latency" setting, things are even worse: >> https://youtu.be/ShuitSWYy5A >> >> The "Normal latency" setting is better, but still has more frequent >> popping than I'd like:>> https://youtu.be/cyZQTWDctEs >> >> The same configuration when pointed at UStream endpoint instead of >> YouTube produces fine audio quality (the following example is >> screencapped from a live ustream player then uploaded to youtube):>> https://www.youtube.com/watch?v=Ng9boh_5_ik >> >> I've been trying to tweak all of the audio parameters I can find to >> discern some magic combination that makes youtube happy -- but I >> haven't been able to find something that gets rid of the popping. >> The fact that UStream likes the audio just fine tells me that there's >> some incompatibility between freeswitch's stream and youtube, and not >> just brokenness on the webrtc or mux setup. I've tried 48kHZ, >> 44.1kHz, and lower; stereo and mono; various interval settings; and a >> lot of other parameters that appear to be unrelated (does >> rtmp.conf.xml get involved in recording a conference at all, or is >> that just for incoming rtmp calls?).>> >> >> Any ideas on what I might need to change to make freeswitch-to- >> youtube live audio smooth?>> >> >> The following is my conference.conf.xml in its entirety. Happy to >> share other configs if helpful.>> >>  >> >> <configuration name="conference.conf" description="Audio Conference">>>  >> <advertise> >> <room name="3001@$${domain}" status="FreeSWITCH"/> >> </advertise> >> >> >> >> <caller-controls> >> <group name="default"> >> <control action="deaf mute" digits="**"/> >> <control action="mute" digits="0"/> >> <control action="vmute" digits="*0"/> >> <control action="vmute snap" digits="*1"/> >> <control action="vmute snapoff" digits="*2"/> >> <control action="energy up" digits="9"/> >> <control action="energy equ" digits="8"/> >> <control action="energy dn" digits="7"/> >> <control action="vol talk up" digits="3"/> >> <control action="vol talk zero" digits="2"/> >> <control action="vol talk dn" digits="1"/> >> <control action="vol listen up" digits="6"/> >> <control action="vol listen zero" digits="5"/> >> <control action="vol listen dn" digits="4"/> >> <control action="hangup" digits="#"/> >> </group> >> </caller-controls> >> >> >> <profiles> >> <profile name="plenary-conference-profile"> >> <param name="domain" value="$${domain}"/> >> <param name="rate" value="44100"/> >> <param name="channels" value="2"/> >> <param name="interval" value="10"/> >> <param name="interval" value="10"/> >> <param name="energy-level" value="0"/> >> >>  >>  >> <param name="muted-sound" value="conference/conf-muted.wav"/> >> <param name="unmuted-sound" value="conference/conf- >> unmuted.wav"/>>>  >> >>  >> <param name="moh-sound" value="silence"/> >> <param name="comfort-noise" value="0"/> >> >> <param name="caller-id-name" value="$${outbound_caller_name}"/>>> <param name="caller-id-number" value="$${outbound_caller_id}"/>>> <param name="conference-flags" value="livearray-sync|minimize-video-encoding|livearray-json- >> status"/>>> >>  >> <param name="video-mode" value="mux"/> >> <param name="video-layout-name" value="overlaps-with-floor"/> >> <param name="video-auto-floor-msec" value='10' /> >>  >> <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="5mb"/> >> <param name="video-fps" value="30" /> >> >> <param name='auto-record' >> value='rtmp://a.rtmp.youtube.com/live2/<... youtube secret >> streaming key ...>'/>>> <param name='min-required-recording-participants' value='1'/> >> </profile> >> </profiles> >> </configuration> >> >> __________________________________________________________________- >> _______>> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 > > > > -- > Anthony Minessale II > Founder, FreeSWITCH. > http://freeswitch.com > > > https://youtu.be/l_hOxzCt6X4 > https://www.youtube.com/watch?v=oAxXgyx5jUw > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > ___________________________________________________________________- > ________> Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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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youtube broadcast from freeswitch audio quality
by Charlie DeTar 02 May '18

02 May '18

Hi, I'm trying to broadcast multiparty audio and video through youtube, using this stack: Browser (Chrome) -> Custom webapp using Verto -> mod_conference -> record to rtmp -> youtube live However, I'm getting poor audio quality on youtube in the live broadcast. Here's an example (this video is screen-capped from the live player, because youtube seems to clean up the recorded file after the fact. I've tested with multiple computers/browsers that this is how it sounds to everyone during the live broadcast.): https://youtu.be/bO8t78YXxjc That is using the "Low Latency" setting in youtube. The conference.conf.xml "interval" parameter seems to also have an effect; I have it set to 10 here as that seemed to get fewer pops than the default 20. Using youtube's "Ultra-low latency" setting, things are even worse: https://youtu.be/ShuitSWYy5A The "Normal latency" setting is better, but still has more frequent popping than I'd like: https://youtu.be/cyZQTWDctEs The same configuration when pointed at UStream endpoint instead of YouTube produces fine audio quality (the following example is screencapped from a live ustream player then uploaded to youtube): https://www.youtube.com/watch?v=Ng9boh_5_ik I've been trying to tweak all of the audio parameters I can find to discern some magic combination that makes youtube happy -- but I haven't been able to find something that gets rid of the popping. The fact that UStream likes the audio just fine tells me that there's some incompatibility between freeswitch's stream and youtube, and not just brokenness on the webrtc or mux setup. I've tried 48kHZ, 44.1kHz, and lower; stereo and mono; various interval settings; and a lot of other parameters that appear to be unrelated (does rtmp.conf.xml get involved in recording a conference at all, or is that just for incoming rtmp calls?). Any ideas on what I might need to change to make freeswitch-to-youtube live audio smooth? The following is my conference.conf.xml in its entirety. Happy to share other configs if helpful.   <configuration name="conference.conf" description="Audio Conference">  <advertise> <room name="3001@$${domain}" status="FreeSWITCH"/> </advertise>   <caller-controls> <group name="default"> <control action="deaf mute" digits="**"/> <control action="mute" digits="0"/> <control action="vmute" digits="*0"/> <control action="vmute snap" digits="*1"/> <control action="vmute snapoff" digits="*2"/> <control action="energy up" digits="9"/> <control action="energy equ" digits="8"/> <control action="energy dn" digits="7"/> <control action="vol talk up" digits="3"/> <control action="vol talk zero" digits="2"/> <control action="vol talk dn" digits="1"/> <control action="vol listen up" digits="6"/> <control action="vol listen zero" digits="5"/> <control action="vol listen dn" digits="4"/> <control action="hangup" digits="#"/> </group> </caller-controls>  <profiles> <profile name="plenary-conference-profile"> <param name="domain" value="$${domain}"/> <param name="rate" value="44100"/> <param name="channels" value="2"/> <param name="interval" value="10"/> <param name="interval" value="10"/> <param name="energy-level" value="0"/>   <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/>   <param name="moh-sound" value="silence"/> <param name="comfort-noise" value="0"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="conference-flags" value="livearray-sync|minimize-video-encoding|livearray-json-status"/>  <param name="video-mode" value="mux"/> <param name="video-layout-name" value="overlaps-with-floor"/> <param name="video-auto-floor-msec" value='10' />  <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="5mb"/> <param name="video-fps" value="30" /> <param name='auto-record' value='rtmp://a.rtmp.youtube.com/live2/<... youtube secret streaming key ...>'/> <param name='min-required-recording-participants' value='1'/> </profile> </profiles> </configuration>

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FreeSWITCH/UniMRCP: Using multiple mrcp profiles in a single dialplan: how to switch/destroy session
by . 02 May '18

02 May '18

Hi, I am trying to use two separate MRCP profiles in the context of the same dialplan. The configurations work fine as long as a given dialplan only uses _one_ of the profiles during a given call. However I currently I have a dialplan where I would like to use profile A. (connecting to MRCP server 1) to field one question, and profile B. (connecting to MRCP server 2) to field another question. I issue the following command for profile A. and question 1: - %[current_grammar=s0]/usr/local/freeswitch/scripts/test-ivr/prompts/prompt.wav detect:unimrcp:profile-a {start-input-timers=false,no- input-timeout=3000,speech-incomplete-timeout=3500, recognition-timeout=5000,define-grammar=true,save- waveform=true,speech-language=es-MX}builtin:speech/grammar1 and the next one for profile B. and question 2: - %[current_grammar=s1]/usr/local/freeswitch/scripts/test-ivr/prompts/prompt.wav detect:unimrcp:profile-b {start-input-timers=false,no- input-timeout=3000,speech-incomplete-timeout=1500, recognition-timeout=5000,define-grammar=true,save- waveform=true}builtin:grammar/grammar2?language=es-MX Again, these both work fine if I only use one profile for a given dialplan application or call. However, in the current case, I can see in my FreeSWITCH logs, in my ESL logs, and in the respective UniMRCP server logs, that the second request is apparently somehow reusing the session and profile details from the first request. In my log I see the correct command is sent, but FreeSWITCH/the UniMRCP client still sends the second request using the connection details from the first profile, which then fails because grammar2 is not available via profile A [because it is handled by the server sitting behind profile B]. If I reverse the order, I see that both requests again arrive at the MRCP server attached to profile B. I dug around the confluence wiki and found some references to parameters being sticky: - https://freeswitch.org/confluence/display/FREESWITCH/ mod_dptools%3A+play_and_detect_speech#mod_dptools: play_and_detect_speech-Lua <https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+play_an…> and also tried manually setting "play_and_detect_speech_close_asr=true", but nothing seems to close the session. Any hints would be greatly appreciated; alternatively if this impossible without digging around in the mod_ source that would be good to know too. Best regards, Joe

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Re: [Freeswitch-users] FSBlue / FreeSWITCH 1.8
by Darren Williams 30 Apr '18

30 Apr '18

Look at it on the positive side, back in the day when working in the enterprise we would not consider rolling an upgrade until the fix had been in the field a good few months. Strangely, these Freeswitch Advantage people become the 'testers' for the rest of the community, by the time the community gets these fixes they should be solid and well tested.

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MoH/IVR & Play_and_get_digits or Playback
by Ismael Blanco 30 Apr '18

30 Apr '18

Hello, I'm working on a scenario where inbound calls to a DID are bridged to a termination provider. At the same time, I'm looking to inject a whisper to the b-leg so that agents can have knowledge on where the call is coming from. On a normal scenario where there is no IVR on MoH (hosted from the b-leg) everything works as expected. However, when there is MoH or an IVR, the caller only hears ringing and not the actual media from the above. I've tested a couple of different scenarios, each without success. Both using a nolocal:execute_on_answer lua script: scenario a) session:execute( "play_and_get_digits", "1 1 10 2000 '' "..playbackstring ); scenario b) session:execute("set","playback_terminators=any") session:execute( "playback", playbackstring ) Any ideas on how the audio can be maintained for the caller would be really appreciated. Thanks in advance, Ismael

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