June 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS6947 - Tuning Opus bandwidth
by Emrah 26 Oct '16

26 Oct '16

Hi list, I re-read FS6947 and don't understand how this problematic was addressed and the issue fixed. The scope is simple. There should be a setting in the dialplan that allows downsampling of Opus for applications that do not require the 48khz / 2 channels framework. I.e.: terminating to the PSTN with Opus to take advantage of low bandwidth and great PLC. There seems to be a lot of confusion around bandwidth in general there. It doesn't matter if the internal clock of the device is always sampling at 48khz / 2ch. There are settings that can facilitate a lower bandwidth consumption for particular use cases, and it seems the reason it is not being implemented in FS is just a matter of being confused about the intent of the 48khz 2ch base. Please revisit this issue. FS should allow tuning of Opus audio / network bandwidth in the dialplan. It would optimize greatly lots of use cases. If I'm calling the PSTN, I'd rather have my client downsample and stream at a lower bandwidth, even if my audio capture would still be at 48khz / 2ch as per the RFC, and save on bandwidth, than transcode the full 48khz spectrum into PCM on my FS and minimize processing power on the client's side. Jira here: https://freeswitch.org/jira/browse/FS-6947 Emrah

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Capturing the Current Call Audio
by Leonardo Lima Ribeiro 13 Oct '16

13 Oct '16

Hi! I’m tying to capture the current call audio but I have no idea on how to do that. My main achievement is to get the current call audio and them stream this audio bytes to an API via HTTP Post. I was doing some tries with mod_shout and mod_vlc, and I was successful in the streaming part - I tested some wavs and mp3 files. But the mainly part is that I need the current call audio, and it has no file in the system to stream right? I was thinking if there’s a command that I can get the audio bytes from the current call but I really don’t know, I could not find any option like that. Could you please help me? Thank you, Leonardo Ribeiro

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Kernel Hyperthreading ON or OFF?
by Raimund Sacherer 06 Oct '16

06 Oct '16

Hi, we have a new Freeswitch cluster with Dell R410. It has 2 CPUs (Xeon E5620, QuadCore. it's 8 cores withouth HT and 16 cores with HT. I googled with hyperthreading and freeswitch but some say turn it on it will give you a boost, others say turn it off, you will gain performance. Does anyone know if it's better to leave it on or turn it off? And some small explanation? I would be very gratefull for your help, best regards Raimund --

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freeswitch 1.6 without video support
by Grant Bagdasarian 19 Aug '16

19 Aug '16

Hello, Is it possible to compile and install fs 1.6 without support for video? Or is the 1.4 version the same as 1.6 with bug fixes and security updates? Regards, Grant

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Testing for call hangup in dialplan xml
by Cody Rosenbloom 29 Jul '16

29 Jul '16

Hi, Is there a way to check if call has been hung up in dialplan xml. I want to run lua script to process voicemail after caller has hungup the call. I tried checking if state is equal to CS_HANGUP but it does not work. Current dialplan xml is at http://pastebin.com/5PMKSrdh Thanks Cody

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Presence issue. How packet should look?
by Igor Olhovskiy 26 Jul '16

26 Jul '16

Hi! I’m trying to subscribe to Freeswitch with this packet SUBSCRIBE sip:*55@master.rufan.at SIP/2.0 Record-Route: <sip:10.0.20.70:5060;ftag=jz8i26dy0f;lr=on> Via: SIP/2.0/UDP 212.232.26.232:5060 ;branch=z9hG4bKbb4c.eafe0e8f9ec4b28fb6e4056a64c702ed.0 Via: SIP/2.0/UDP 192.168.88.59:37600 ;received=176.37.91.15;branch=z9hG4bK-rzfuku5tc99o;rport=37600 Route: <sip:10.0.20.70:5060;ftag=jz8i26dy0f;lr=on> From: <sip:607@master.rufan.at>;tag=jz8i26dy0f To: <sip:*55@master.rufan.at;user=phone>;tag=qBuxIeW1MNWF Call-ID: 313435373534383931393431313632-94bfcwdyrzz2 CSeq: 610 SUBSCRIBE Max-Forwards: 69 User-Agent: snom760/8.7.5.35 Contact: <sip:607@176.37.91.15:37600;line=ukipitb6>;reg-id=1 Event: dialog Accept: application/dialog-info+xml Expires: 80 Content-Length: 0 Path: <sip:212.232.26.232:5060;lr;received=sip:176.37.91.15:37600> X-AUTH-IP: 176.37.91.15 Actually I get Accepted (202) But with NOTIFY from freeswitch I got this NOTIFY sip:607@176.37.91.15:37600;line=ukipitb6 SIP/2.0 Via: SIP/2.0/UDP 10.0.20.71;rport;branch=z9hG4bKS5ZKag4N6XXFK Route: <sip:10.0.20.70:5060>;ftag=jz8i26dy0f;lr=on Record-Route: <sip:10.0.20.70:5060>;ftag=jz8i26dy0f;lr=on Max-Forwards: 70 From: <sip:*55@master.rufan.at;user=phone>;tag=qBuxIeW1MNWF To: <sip:607@master.rufan.at>;tag=jz8i26dy0f Call-ID: 313435373534383931393431313632-94bfcwdyrzz2 CSeq: 297327351 NOTIFY Contact: <sip:*55@10.0.20.71:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Event: dialog Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=80 Content-Type: application/dialog-info+xml Content-Length: 152 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="8" state="full" entity="sip:*55@master.rufan.at"> </dialog-info> Problem is, XML data is empty, cause actually there is no registrations for *55 extension. Why it’s not sending «Unavail» or smth. like this? What is the correct packet for SUBSCRIBE? Or internal profile? Internal profile is nonce-ttl [60] auth-calls [true] inbound-reg-force-matching-username [true] auth-all-packets [false] ext-rtp-ip [10.0.20.71] ext-sip-ip [10.0.20.71] rtp-timeout-sec [300] rtp-hold-timeout-sec [1800] tls-verify-policy [all] multiple-registrations [contact] enable-timer [false] dbname [share_presence] send-presence-on-register [true] inbound-codec-negotiation [greedy] NDLB-force-rport [safe] challenge-realm [auto_to] outbound-proxy [10.0.20.70] track-calls [true] nat-options-ping [true] liberal-dtmf [true] all-reg-options-ping [true] force-publish-expires [true] unregister-on-options-fail [true] user-agent-string [FreeSWITCH] log-auth-failures [true] forward-unsolicited-mwi-notify [false] context [public] rfc2833-pt [101] sip-port [5060] dialplan [XML] dtmf-duration [2000] inbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] outbound-codec-prefs [G7221@32000h,G7221@16000h,G722,PCMU,PCMA,OPUS,SILK] rtp-timer-name [soft] rtp-ip [10.0.20.71] sip-ip [10.0.20.71] hold-music [local_stream://default] apply-nat-acl [nat.auto] aggressive-nat-detection [true] apply-inbound-acl [domains] local-network-acl [localnet.auto] record-path [/usr/local/freeswitch/recordings] record-template [${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}] manage-presence [true] presence-probe-on-register [true] manage-shared-appearance [true] tls [false] -- Best regards, Igor

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Docker DTMF related issue
by Danny Gershman 10 Jul '16

10 Jul '16

I'm having an issue with my DTMF to a Docker container running FS 1.6.9. My sofia profile is set up to use RFC2833. However from a softphone and another FS server it's not detecting the digits. If I switch to INFO method, it works fine. Also is there a quick was to do a uuid_send_dtmf using SIP INFO to work around this issue temporarily? I saw something about a channel variable dtmf_type, which didn't seem to work when I did a uuid_setvar. Thanks, Danny Gershman

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Update RTP Address On Re-Invite
by Colin Morelli 08 Jul '16

08 Jul '16

I have mobile applications running behind NAT. When their reachability changes (and the device's local route updates), I want to automatically switch the RTP stream to the new address/port combination. I've tried using the RTP ALWAYS_AUTO_ADJUST bug, but that results in *very* choppy audio. I also don't really need it, as the only time I care to perform RTP auto-adjust is after an invite session. For this case I can safely assume that the client *will* send a re-invite when its address changes. Is there any way to perform RTP audio adjust only on a re-invite? Using STUN doesn't seem right for two reasons: 1) on its own it can't solve this problem (even if it determines the external IP, the port is still wrong). 2) it results in issues on FS "Invalid STUN/ICE packet received 20 bytes" Would appreciate any help. Best, Colin

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Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function
by Andrew Keil 08 Jul '16

08 Jul '16

To FreeSWITCH Users, See below for a sample template for a Lua Service Script running inside FreeSWITCH. The issue I have is fairly straightforward. I need a function to run when hangup is detected (ie. at the end of the call) however I understand this must not delay ending the script. This function is CleanUp(). Then I would like the service to end. The problem I am having is if the caller hangs up during the playback of "intro.wav" (as shown inside the MainService() function below), then the code jumps to the myHangupHook which calls CleanUp() perfectly, the issue is once CleanUp() is complete I would like the Lua script to end there and then (ie. at the bottom of CleanUp()). What actually happens is it returns to MainService() and continues to try and play "info.wav", unless I either check for session:ready() everywhere or add a goto as shown below under each streamFile() function call. My aim is to reduce extra code and to make the Lua script simpler and easier to read. Also I would like to try and avoid goto statements, which I know can be done with if (session:ready()) etc.... So is there a way to stop a Lua script running inside FreeSWITCH cleanly? I have tried the os.exit() this is barred from use by FreeSWITCH. I have also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time! I could look further into the Lua additions done by the FreeSWITCH team in the source code, however if someone has already solved this then that would be the best solution. FYI: Obviously the script below is simple, however I am sure that you understand if the script was complicated having to use "if (session:ready()) then ...." or "if (not session:ready()) then goto HANGUPEXIT end" makes the code ugly. Thanks in advance, Andrew Keil Visytel Pty Ltd ------------------------------------------------------------------------------ Sample Lua Service ----------------------------------------------------------------------- -- Lua template for FreeSWITCH service -- By: Andrew Keil (Visytel Pty Ltd) -- Email: support(a)visytel.com -- Setup script wide variables here function PreAnswer() freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); -- Add your pre answer code from here -- End of your pre answer code freeswitch.consoleLog("INFO", "PRE ANSWER SECTION COMPLETE\n"); end function AnswerCaller() session:answer() session:sleep(1000) end function MainService() freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n"); if (session:ready()) then -- Note (1): If you wish to end the call then simply use: goto ENDSERVICE -- Note (2): To terminate the service sooner when HANGUP is detected use: if (not session:ready()) then goto HANGUPEXIT end -- Add your main service code from here (caller would have been answered) session:streamFile("intro.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("info.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("outro.wav") if (not session:ready()) then goto HANGUPEXIT end -- End of your main service code end ::ENDSERVICE:: if (session:ready()) then -- End of service so hangup session:hangup() -- Should automatically jump to CleanUp() via hangup handler if caller still online at this stage end goto END ::HANGUPEXIT:: freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP DETECTED)\n"); ::END:: freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION COMPLETE\n"); end function CleanUp() freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); -- Add your cleanup code from here (caller would have been disconnected) -- End of your cleanup code freeswitch.consoleLog("INFO", "CLEANUP SECTION COMPLETE\n"); end function myHangupHook(s, status, arg) session:hangup() CleanUp() -- Run CleanUp function now since the caller has disconnected end -- Setup Hangup event handler here v_hangup = "HANGUP" session:setHangupHook("myHangupHook", "v_hangup") -- Call service functions in order PreAnswer() AnswerCaller() MainService() -- End of Lua service

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Security vs compatibility / NAT etc
by Rick Jarvis 06 Jul '16

06 Jul '16

I’d be interested to hear what different people use to provide some level of security for remote end-users such as homeworkers, and to get round NAT issues. We currently use OpenVPN, as this is built into the firmware of Yealink handsets (it’s a great feature, I’m not sure why more handset manufacturers don’t do this?!). The pros are that not only is it secure, but it also removes any problems with NAT for the RTP streams. The downsides are that it is complicated (and downright frustrating sometimes) to set up, and there are additional things to consider such as the server configuration and overheads. TLS/SSL with SRTP is another option, but my understanding of this is that it can cause NAT problems, with FreeSWITCH trying to initiate control channels back to the phone for inbound calls. In fact, I’ve always had problems with getting phones to work when behind NAT anyway, even without SSL/TLS. STUN can be used to ascertain the IP, but how do you handle situations where multiple handsets are behind NAT - you can’t open all RTP ports to all handsets at once?!! Would be very interested to hear thoughts and methods on these points. Thanks R

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