September 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Opus problem
by Volkan Oransoy 26 Sep '17

26 Sep '17

Hi All, I am trying to transcode between OPUS and G711. I encountered a strange problem which I couldn’t find any clue about the cause. After the channel has been answered with opus, I hear a strange sound (highly distorted version of my voice) for 10 seconds and it starts to operate properly with the message "[DEBUG] switch_core_media_bug.c:1198 Removing BUG from…” . I can reproduce same problem with echo application. Here is my logs and opus.conf.xml files. Did anybody experienced this issue? Thanks. /Volkan My FreeSWITH Version: FreeSWITCH Version 1.6.16+git~20170407T164913Z~3da6bd0108~64bit (git 3da6bd0 2017-04-07 16:49:13Z 64bit) Opus.xml config: <configuration name="opus.conf"> <settings> <param name="use-vbr" value="1"/> <param name="use-dtx" value="1"/> <param name="complexity" value="10"/> <param name="maxaveragebitrate" value="14400"/> <param name="maxplaybackrate" value="8000"/> <param name="packet-loss-percent" value="15"/> <param name="keep-fec-enabled" value="1"/> <param name="use-jb-lookahead" value="0"/> <param name="advertise-useinbandfec" value="1"/> <param name="asymmetric-sample-rates" value="1"/> </settings> </configuration> Freeswitch Log: 2017-09-25 22:01:06.277462 [NOTICE] switch_channel.c:1104 New Channel sofia/internalopus/2223(a)sip.example.com [1048c869-8d71-41fd-b0b7-f8fbd01a0d93] 2017-09-25 22:01:06.277462 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223(a)sip.example.com) Running State Change CS_NEW (Cur 2 Tot 3196652) 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:9828 sofia/internalopus/2223(a)sip.example.com receiving invite from 192.168.100.2:5060 version: 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223(a)sip.example.com entering state [received][100] 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7058 Remote SDP: v=0 o=- 8070049943 45777 IN IP4 192.168.100.2 s=dvjtiwu c=IN IP4 192.168.100.2 t=0 0 m=audio 10036 RTP/AVP 103 8 0 9 101 a=rtpmap:103 opus/48000/2 a=fmtp:103 maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=nortpproxy:yes . . . . . 2017-09-25 22:01:06.337394 [INFO] mod_dialplan_xml.c:637 Processing 2223 <2223>->*71 in context default Dialplan: sofia/internalopus/2223(a)sip.example.com parsing [default->echo] continue=false Dialplan: sofia/internalopus/2223(a)sip.example.com Regex (PASS) [echo] destination_number(*71) =~ /^\*71$/ break=on-false Dialplan: sofia/internalopus/2223(a)sip.example.com Action answer() Dialplan: sofia/internalopus/2223(a)sip.example.com Action echo() 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:286 (sofia/internalopus/2223(a)sip.example.com) State Change CS_ROUTING -> CS_EXECUTE 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:643 (sofia/internalopus/2223(a)sip.example.com) State ROUTING going to sleep 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223(a)sip.example.com) Running State Change CS_EXECUTE (Cur 2 Tot 3196652) 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:650 (sofia/internalopus/2223(a)sip.example.com) State EXECUTE 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:198 sofia/internalopus/2223(a)sip.example.com SOFIA EXECUTE 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:328 sofia/internalopus/2223(a)sip.example.com Standard EXECUTE EXECUTE sofia/internalopus/2223(a)sip.example.com answer() 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4345 Set telephone-event payload to 101@8000 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:3049 Set Codec sofia/internalopus/2223(a)sip.example.com opus/8000 20 ms 960 samples 0 bits 1 channels 2017-09-25 22:01:06.357410 [DEBUG] switch_core_codec.c:111 sofia/internalopus/2223(a)sip.example.com Original read codec set to opus:116 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4747 sofia/internalopus/2223(a)sip.example.com Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6803 AUDIO RTP [sofia/internalopus/2223(a)sip.example.com] 192.168.100.12 port 23594 -> 192.168.100.2 port 10036 codec: 103 ms: 20 2017-09-25 22:01:06.357410 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 960 bytes per 20ms 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7109 sofia/internalopus/2223(a)sip.example.com Set 2833 dtmf send payload to 101 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7116 sofia/internalopus/2223(a)sip.example.com Set 2833 dtmf receive payload to 101 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7139 sofia/internalopus/2223(a)sip.example.com Set rtp dtmf delay to 40 2017-09-25 22:01:06.357410 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internalopus/2223(a)sip.example.com! 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3473 (sofia/internalopus/2223(a)sip.example.com) Callstate Change RINGING -> EARLY 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6786 Audio params are unchanged for sofia/internalopus/2223(a)sip.example.com. 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:850 Local SDP sofia/internalopus/2223(a)sip.example.com: v=0 o=FreeSWITCH 1506342472 1506342473 IN IP4 192.168.100.12 s=FreeSWITCH c=IN IP4 192.168.100.12 t=0 0 m=audio 23594 RTP/AVP 103 101 a=rtpmap:103 opus/48000/2 a=fmtp:103 useinbandfec=1; usedtx=1; maxaveragebitrate=14400; maxplaybackrate=8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2017-09-25 22:01:06.357410 [NOTICE] mod_dptools.c:1312 Channel [sofia/internalopus/2223(a)sip.example.com] has been answered 2017-09-25 22:01:06.357410 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223(a)sip.example.com entering state [completed][200] 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/internalopus/2223(a)sip.example.com 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3772 (sofia/internalopus/2223(a)sip.example.com) Callstate Change EARLY -> ACTIVE EXECUTE sofia/internalopus/2223(a)sip.example.com echo() 2017-09-25 22:01:06.737463 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223(a)sip.example.com entering state [ready][200] 2017-09-25 22:01:06.837459 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed. 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:448 Setting BUG Codec opus:116 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps] 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz] 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:780 Engaging Read Buffer at 320 bytes vs 15 [...CREEPY VOICE STARTS HERE AND ENDS WITH THE BELOW MESSAGE...] 2017-09-25 22:01:16.017463 [DEBUG] switch_core_media_bug.c:1198 Removing BUG from sofia/internalopus/2223(a)sip.example.com <mailto:sofia/internalopus/2223@sip.example.com>

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Multiple FS Servers - Sofia Shared Database - Behind Kamailio
by Geoff Mina 25 Sep '17

25 Sep '17

Greetings, I am hoping this is a relatively simple question - but I can't find a lot of specific information on what I am attempting to do - although I have found enough to believe that this configuration is possible. I have 2 FS hosts (FreeSWITCH Version 1.6.19~64bit) behind a Kamailio server. Kamailio is routing REGISTER and INVITE requests to both using the Dispatcher module. I have a single MySQL server shared by both FS hosts and mod_sofia is properly using the shared MySQL server via ODBC on the back-end. I have two softphones registering through Kamailio and all is good. I can see both registrations in the "sip_registrations" table. The problem comes in when I try to call one of the extensions. If the INVITE arrives on the opposite server to the last REGISTER for that extension - the FS server can't see the registered user since the "hostname" is different now. I can experiment with it by using "sofia_contact 1000" on both servers - and only the last server that accepted the REGISTER sees the user, the other returns "error/user_not_registered". Can anyone point me in the right direction to understand what might need to be changed to allow both FS servers to share the same records in "sip_registrations"? Thanks, Geoff

1 1

RTP Loss with Freeswitch in Default Mode
by vikas sharma 25 Sep '17

25 Sep '17

Hi I am making a linphone to linphone call using Freeswitch server in Defualt mode but i am facing huge RTP loss and the video call is getting stuck in between.I have also found that there is a difference between the RTP packets received on A Leg and sent from B-leg and vice versa. Can somebody help me to understand what exactly happens with the RTP packets on server and why there is a difference in the RTP count?? How can we improve the video call quality in moderate network conditions. Thanks

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Re: [Freeswitch-users] Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread
by Robert Mundkowsky 25 Sep '17

25 Sep '17

Any reason you are against Python event handlers? -------- Original message -------- From: Kamal Mustafa <kamal.mustafa(a)gmail.com> Date: 9/24/17 10:20 PM (GMT-05:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread On 10 May 2017 at 02:39, Lesley Pervis <lesley.pervis(a)gmail.com> wrote: > Late to answer. Seems no one did, so I'd say don't write event listeners > that run in a Python interpreter that FreeSWITCH knows nothing about and > uses threading. Hi, thanks for the reply and advice. I'd appreciate that. It's been quite sometimes after I'm working on this (forgot about this email already) but with recent problem I have, I think we have to open new connection in the handler thread if we want to send another command over the esl socket. We can't reuse the same connection in the main thread. Since we're at this, what you think about doing billing in python ? It sound crazy but after seeing cgrates also doing the billing over the event socket, I think maybe it feasible - https://github.com/cgrates/cgrates/blob/master/docs/freeswitch.rst#811-sess… I've started a PoC of billing in python and it seem to work, although I haven't put it under load and concurrent tests yet. _________________________________________________________________________ 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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Python-ESL: ESLconnection.execute() blocked when handling channel_park under thread
by Mohd Kamal Bin Mustafa 25 Sep '17

25 Sep '17

I have this python code:- import time import json import logging import threading from freeswitchESL import ESL class EventThread(threading.Thread): def __init__(self, target, args): super(EventThread, self).__init__() self.target = target self.args = args def run(self): print("Running target") self.target(*self.args) def on_channel_park(e): uuid = e.getHeader('Unique-ID') print("Ringing") conn.execute('ring_ready ', '', uuid) conn.execute('sleep', '5500', uuid) conn.execute('answer', '', uuid) print("answered, playing media ...") media_url = 'http://www.noiseaddicts.com/samples_1w72b820/17.mp3' resp = conn.execute("playback", media_url, uuid) print(resp.getHeader("Reply-Text")) print("playback done") print(resp) if __name__ == '__main__': conn = ESL.ESLconnection('127.0.0.1', '8021', 'mypass') conn.events('plain', 'all') while True: e = conn.recvEvent() uuid = e.getHeader('Unique-ID') events_message = json.loads(e.serialize('json')) event_name = events_message['Event-Name'].lower() method_name = 'on_%s' % event_name print(method_name) if event_name == 'channel_park': print("spawning thread") t = EventThread(on_channel_park, args=(e,)) t.start() if event_name == 'playback_stop': conn.execute("hangup", "", uuid) print("Hanging up") Above, the code will stuck at conn.execute('ring_ready ', '', uuid) in on_channel_park(). It never went pass through until I hangup the call. In freeswitch log, it stopped at executing park(). But if I moved ring_ready, sleep, answer outside of the thread it will work, i.e I can hear a playback:- def on_channel_park(e): uuid = e.getHeader('Unique-ID') print("answered, playing media ...") media_url = 'http://www.noiseaddicts.com/samples_1w72b820/17.mp3' resp = conn.execute("playback", media_url, uuid) print(resp.getHeader("Reply-Text")) print("playback done") print(resp) if __name__ == '__main__': conn = ESL.ESLconnection('127.0.0.1', '8021', 'mypass') conn.events('plain', 'all') while True: e = conn.recvEvent() uuid = e.getHeader('Unique-ID') events_message = json.loads(e.serialize('json')) event_name = events_message['Event-Name'].lower() method_name = 'on_%s' % event_name print(method_name) if event_name == 'channel_park': print("Ringing") conn.execute('ring_ready ', '', uuid) conn.execute('sleep', '5500', uuid) conn.execute('answer', '', uuid) print("spawning thread") t = EventThread(on_channel_park, args=(e,)) t.start() Any idea why this happen ?

3 2

gsmopen one-way voice issue
by Jeff Chua 24 Sep '17

24 Sep '17

I'm configured freeswitch with gsmopen but can't seem to figure where why I only get voice out from SIP-client to mobile-phone but not voice from mobile to SIP client. When I tried to call from mobile-phone to IVR extension 5000 and press 2 for the standard echo test (9196), I could hear the echo back. When I tried to call 9195 for the five second delay echo test, I couldn't anything. I've add these settings in vars.xml as well. <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=true"/> <X-PRE-PROCESS cmd="set" data="inbound-late-negotiation=false"/> <X-PRE-PROCESS cmd="set" data="inbound-zrtp-passthru=false"/> <X-PRE-PROCESS cmd="set" data="disable-transcoding=false"/> <X-PRE-PROCESS cmd="set" data="inherit_codec=false"/> <X-PRE-PROCESS cmd="set" data="bypass_media=false"/> <X-PRE-PROCESS cmd="set" data="proxy_media=false"/> What am I missing? Thanks, Jeff

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JSON API
by Rick Jarvis 24 Sep '17

24 Sep '17

Did anything ever come of this http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115482.h… <http://lists.freeswitch.org/pipermail/freeswitch-users/2015-August/115482.h…> ? I’ve given up with trying to get information over WSS, so am back to using Event Socket - it would be good to know exactly which commands I can use to get JSON, or whether I’ll need to write a parser… ‘show channels’ for instance just errors...

1 0

tls srtp calls between users white noise
by Bipin Patel 23 Sep '17

23 Sep '17

hi, i got tls and srtp working between client and FS, client registers to FS using TLS and SRTP and if that call is forwarded to gateway in SIP and RTP everything works fine. Problem arises when 2 clients using zoiper register to FS using TLS and SRTP and any calls routed between them the calls goes through and the phone rings but on answering there is white noise constantly, i checked in FS and it does send the call using TLS and SRTP and key exchange and codec negotiation all is fine using g711u but i cant seem to get rid of this white noise almost making it impossible to each party to hear the other. any idea what could be wrong my dialplan looks like this <extension name="unloop"> <condition field="${unroll_loops}" expression="^true$"/> <condition field="${sip_looped_call}" expression="^true$"> <action application="deflect" data="${destination_number}"/> </condition> </extension> <extension name="general_stuff" continue="true"> <condition> <action application="set" data="suppress_cng=true"/> </condition> </extension> <extension name="local_call"> <condition field="destination_number" expression="^000$"> <action application="export" data="rtp_secure_media=true"/> <action application="bridge" data="${sofia_contact(${vps_id})}"/> </condition> </extension> my porfile looks like this <param name="inbound-bypass-media" value="false"/> <param name="inbound-late-negotiation" value="false"/> <param name="inbound-zrtp-passthru" value="false"/> <param name="disable-transcoding" value="true"/> <param name="vad" value="none"/> <param name="rtp-autofix-timing" value="false"/> <param name="rtp-rewrite-timestamps" value="false"/> <param name="send-display-update" value="false"/> <param name="accept-blind-reg" value="false"/> -- Regards, Bipin ------------------------------------------------------------------------

1 1

Modifying RURI for outbound calls
by mcooley 22 Sep '17

22 Sep '17

Hi, How do I modify the RURI for outbound calls? I'd like to add the port, protocol and user=phone So, this: INVITE sip:3035551212@10.10.10.10 SIP/2.0 to this: INVITE sip:3035551212@10.10.10.10:5060;user=phone;trasport=udp SIP/2.0 thanks, mike -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

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Re: [Freeswitch-users] Session recording performance
by jungle Boogie 22 Sep '17

22 Sep '17

What occurs if you send identical audio for the same duration? IOW, play the sound file for each call.

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