August 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Specifying G722.1 bitrate
by Andrew White 25 Aug '20

25 Aug '20

Hi all, Our server has had someone send us the following SDP: v=0 o=- 1598394304 1598394304 IN IP4 1.2.3.4 s=- c=IN IP4 1.2.3.4 t=0 0 m=audio 14198 RTP/AVP 8 0 96 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 G7221/16000 a=fmtp:96 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=silenceSupp:off - - - - We have the following configuration in our vars.xml file: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@16000h,G7221@32000h,OPUS,G722,PCMU,PCMA”/> However the debug log shows the following output: e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G7221:107:16000:20:32000:1] e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G7221:115:32000:20:48000:1] …snip… e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:96:16000:20:24000:1]/[G7221:107:16000:20:32000:1] e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:96:16000:20:24000:1]/[G7221:115:32000:20:48000:1] e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:96:16000:20:24000:1]/[opus:116:48000:20:0:1] e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:96:16000:20:24000:1]/[G722:9:8000:20:64000:1] e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:96:16000:20:24000:1]/[PCMU:0:8000:20:64000:1] e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:5478 Audio Codec Compare [G7221:96:16000:20:24000:1]/[PCMA:8:8000:20:64000:1] …snip… e95e4aaa-e721-11ea-ba06-4735eb6f008b 2020-08-26 08:25:42.327098 [DEBUG] switch_core_media.c:3781 Set Codec sofia/mrf/nobody(a)4.5.6.7 <mailto:sofia/mrf/nobody@4.5.6.7>:5060 G722/8000 20 ms 160 samples 64000 bits 1 channels It seems we’re trying to match G722.1 16-24k (from the sender) with G722.1 16-32k (which we have configured). I cannot however find a way to configure G722.1 in FreeSWITCH to accept that bitrate range. I did try setting “G7221@24000h” in the vars.xml file, however this didn’t seem to have any effect It’s entirely possible I’ve also misunderstood the issue here, so any direction would be very much appreciated. Thanks in advance! Andrew

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Can mod_python be adapted tap media?
by David P 24 Aug '20

24 Aug '20

Hi Brian, We need a media bug that allows us to process inbound audio and video using python. What's the best way to do this? Should we diff mod_python with mod_cv to see what C fns to add to the former in order to receive video streams? And diff with something else to find what audio handlers to add to mod_python? In ClueCon session 2, Nickolay seemed to indicate a need for this also. He streams audio out of FS using unimrcp but mentions that that can't be load-balanced. The other alternative, rtmp, imposes several seconds of latency at the start. It seems that load-balancing media connections into a farm of FSs, and doing media handling in a module, would be better overall. Yet there seems no guidance about how to implement it. That's what a media bug does. record_session is an example of such functionality. /b On Mon, Aug 10, 2020 at 1:24 AM David P <davidswalkabout(a)gmail.com> wrote: Where can I find guidance about creating a media bug for audio+video? Adapt mod_cv? I'd like the inbound streams to be handled by a scriptable env like python. Or could streams be exposed to mod_python?

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MOH not playing to A Number
by Hamid Hashmi 19 Aug '20

19 Aug '20

Dear Users I am trying to configure MOH but facing issue when B number holds the call and MOH does not play to A Number instead a "ringing" is started playing at A number and un-holding the call didn't resume media while vice versa is working fine. Moreover, if I disable 'proxy_hold' and 'renegotiate_after_hold' in sip profile then both scenarios work fine but as per requirement, I need both options enabled. Regards Hamid R. Hashmi

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verto screen sharing chrome extension not available
by Assaf Dahary 18 Aug '20

18 Aug '20

Hi, I've compiled from source a fresh Freeswitch Version 1.10.3 with verto-communicator for video conference application. The video conference works well except the screen sharing when using the Chrome browser. When clicking on the Screenshare icon (in call) a popup warning is coming up with this link to install the screen-capturing extension: https://chrome.google.com/webstore/detail/screen-capturing/ajhifddimkapgcif… The problem is that this link results with 404 (The requested URL was not found on this server). Is there any other chrome extension compatible with the verto screen sharing function? *This screen sharing extension is available in the chrome web store (among others) but not compatible with the verto system: https://chrome.google.com/webstore/detail/webrtc-desktop-sharing/nkemblooio… Regards Assaf

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Freeswitch not replying ACK to some 200 OK messages
by Alejandro Recarey 17 Aug '20

17 Aug '20

Hello everyone! I’m having an issue where freeswitch sometimes will not send ACK back to a 200 OK message. These are outbound calls made by mod_callcenter, and every 64 seconds a RE-INVITE is issued. My UA’s answer with a 200 OK, and most of the times, Freeswitch then sends an ACK. However, sometimes the ACK is not sent, and this causes the UA to drop the call 30 seconds later. At the moment I’m checking on ways to increase the session timers for these calls, as all of my UA are WSS connections and thus TCP based, I’m not worried about NAT. But sending so many critical REINVITES per call means that if something is missed like is happening, the call will drop. Has anybody else encountered this issue? I’m running on 1.10. I don’t see anything special in the Freeswitch logs. Curiously, the header shows `Session-Expires: 120;refresher=uac` but FS is refreshing the session every 64 seconds. I know its ClueCon now, congrats everyone! I hope someone has encountered this before and can point me in the right direction. Thanks again!

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callcenter_track and execute_on_originate
by Alejandro Recarey 17 Aug '20

17 Aug '20

Hi everyone! We have implemented callcenter_track in our deployment to get around issues where our agents would still receive calls from mod_callcenter when they were busy. This has worked fine *except* when the agent is receiving or sending a call that is in ringing status (180 sent and no 200 received yet). I tracked it down to firing callcenter_track with “execute_on_answer”, so instead I’m executing callcenter_track on “execute_on_ring” and “execute_on_originate”. However, even executing “callcenter_track” with “execute_on_originate/ring” the agent still receives a new call when the current one is ringing. Is this expected behavior for “callcenter_track” functionality? Is there a way to count ringing channels as occupied? Thanks for your help!

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MCU Video Performance Test Results
by Alexander Haugg 17 Aug '20

17 Aug '20

Hi together, at the moment I’m generating performance test results for video conferencing. The hardware is an AMD 8 core with 4 GHz an 16 GB Ram OS is a Debian 10 Freeswitch is the current stable version. The load is generated from a second FreeSwitch which playback some video recordings called via the conferece_set_auto_outcall DP tool from the performance test FreeSwitch. The ulimit config from the page “https://freeswitch.org/confluence/display/FREESWITCH/Performance+Testing+and+Configurations” is set. The conference profile looks like: <profile name="video-mcu-stereo-720-15fps"> <param name="domain" value="$${domain}"/> <param name="rate" value="48000"/> <param name="channels" value="1"/> <param name="interval" value="20"/> <param name="energy-level" value="200"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="alone-sound" value="conference/conf-alone.wav"/> <param name="moh-sound" value="$${hold_music}"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="false"/> <param name="conference-flags" value="minimize-video-encoding|video-required-for-canvas|video-mute-exit-canvas|mute-detect"/> <param name="video-auto-floor-msec" value="1000"/> <param name="video-mode" value="mux"/> <param name="video-canvas-count" value="1"/> <param name="video-quality" value="1"/> <param name="video-layout-name" value="3x3"/> <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="3mb"/> <param name="video-fps" value="15"/> <param name="suppress-events" value="stop-talking,start-talking" /> </profile> For the first conference with 10 video member the FreeSwitch needs 210% (of 800%) CPU. If I’m adding a second conference with 10 video member the FreeSwitch needs 350% CPU. Mem is minimum. Is there a possibility to get more performance? What’s your experience? If I’m closed the performance tests, I will send my results to the list!

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Help setting up external profile for registrations
by Jungle Boogie 16 Aug '20

16 Aug '20

Hi, Problem I want to solve: Allow internal registrations to FreeSWITCH behind NAT AND for external authenticated registrations to FreeSWITCH. I think this means using two different profiles. I'm using the default one and one I've called "outside". There's a page that touches on the second half but I need a bit more help with it. https://freeswitch.org/confluence/display/FREESWITCH/External+Profile There's basically three changes I've made: created a new sip_profile; created a new directory called outside.xml; created a new dialplan called outside.xml sip_profile changes for external5090.xml (like suggested on the page): 1. Deleted gateway registrations 2. updated sip-port to 5090 3. Updated the context to a new one "outside" directory/outside.xml: 1. Use domain sip.example.com 2. Use group name outside 3. Create a user and put it at directory/outside/ with the user_context of outside dialplan/outside.xml: 1. Use context name ouside The problem: User 2602 can register to sip.example.com:5090 and dial 3500 user 2600 can register on the LAN and call 3500. Both have bidirectional audio. If user 2602 calls user 2600, the phone immediately disconnects on answer on 2600 end, and no audio is heard. A 481 Call Does Not Exist is generated and I don't know why. SIP trace here: https://pastiebin.com/5f38e341e3fd8 Another thing I noticed is sofia status says sip.example.com is an alias for internal. Is that right? https://pastiebin.com/5f38e3b0276d3 Thanks for any pointers, Sean

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multiple extensions on incoming
by John Tuxies 13 Aug '20

13 Aug '20

I am having my system with the DIDs assigned on each extension for the incoming calls. My /dialplan/public has the following structure: *<include> <extension name="public_did"> <condition field="destination_number" expression="^(2109065738)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1020 XML default"/> </condition> </extension></include>* I would like this DID to reach I would like to have the one DID to reach 3 extensions simultaneously 1020-22 * <include> <extension name="public_did"> <condition field="destination_number" expression="^(2109065738)$"> <action application="set" data="domain_name=$${domain}"/> <action application="transfer" data="1023 XML default"/> </condition> </extension></include> * and a 2nd DID to call an extension(1023) and after 10secs to call another extension(1024) and after 10 secs make an outgoing call to a particular mobile number(69400700123) Thanks in advance,

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Retrieval of Caller-ID in Valet parking
by Chandramouli P 13 Aug '20

13 Aug '20

Hello, Please find my below environment: Operating System: Debian (v. 9.13) FreeSWITCH Version: 1.10.3 (64bit) Softphones that I am using: MicroSIP, 3CX, Grandstream I hosted our server in AWS and was able to make and receive calls successfully. Along with "valet_park_in" and "valet_park_out", also enabled the "CID Lookup" module. I am facing two issues in Valet parking and below is the scenario: User A calls User B. User B transfers User A call to *5900. Now User B is free, and User A is in valet parking at 5901 extension. Here, It is announcing the parked extension i.e. 5901 to User A, instead of User B. It should announce to User B. Right? *How to solve this issue?* Now, User C calls *5901 and is able to speak to User A. In this scenario, User C should be able to see the actual caller ID of User A. But, he is seeing only *5901. *How to get the original caller ID of User A and display to User C?* Any help would be appreciated. Thank you. Best Regards, Chandramouli.

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