September 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Session recording performance
by Stanislav Sinyagin 20 Sep '17

20 Sep '17

I'm building an environment for voice quality testing, and found a strange behavior. Maybe someone has an explanation. Two identical physical computers (PC Engines APU2), connected back to back (A and B). A initiates the test calls to B's SIP address, and then both ends go through a small dialplan (less than 100 lines of XML, similar to https://github.com/voxserv/fsqa/blob/master/fsqa.xml and start recording the session. The dialplan initiates session recording to a Raw file, and plays back a 20-second audio sample, and then inserts the filename into a Redis queue, then hangs up. Then the files get processed by Sevana AQuA software in lowest CPU priority, and deleted. The files are written into a tmpfs filesystem. With 2 calls per second, I get 48 simultaneous calls, and B indicates frame drops in the audio recordings, while A shows good results. The CPU average idle time is always above 50% on both computers. Why is the B side (the one receiving INVITEs) getting more frame drops than the A (the one originating the calls)? I flipped the sides, and then the other machine started to get more drops. At first I started recording in WAV format, and similar effect started to appear at a call every 2 seconds, resulting in 12 simultaneous calls. With raw recording, the performance is much better, but still it's a mistery why there are dropped frames in the audio. The CPU is always running at its highest frequency, under "performance" governor.

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sox L16 to G711, inconsistent results
by Stanislav Sinyagin 20 Sep '17

20 Sep '17

In order to utilize mod_native_file, I converted the original L16 8000Hz audio into G711 with sox: sox ${SRCFILE} -t raw -r 8k -b 8 -c 1 -e u-law ${DSTDIR}/${BASENAME}.PCMU I just discovered that if I do this command twice and compare resulting files, they differ. They also differ if I take 44100Hz source audio. Anyone got a clue what's going on? FreeSWITCH is always producing the same bitstream when converting L16 to G711.

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Unable to catch extra sip headers
by Aqs Younas 20 Sep '17

20 Sep '17

Greetings list, I am trying to catch extra sip header in dialplan but of no avail. I have tried info app and uuid_dump on the call to make show variables are appearing in console, but they were not. Though I can see header in sip trace when i run "sofia global siptrace on". This is how it is appearing in sip trace. 2017/09/20 11:22:37.025662 35.189.47.13:5060 -> 10.240.0.23:5063 INVITE sip:0061862083277@sip99.mondotalk.com:5060;transport=udp SIP/2.0 Record-Route: <sip:35.189.47.13;lr;ftag=9d3880-6709294b-13c4-50029-f25-3460e079-f25;did=985.187c7a27> From: "8300005"<sip:8300005@sip99.mondotalk.com:5060 >;tag=9d3880-6709294b-13c4-50029-f25-3460e079-f25 To: <sip:0061862083277@sip99.mondotalk.com:5060> *SRC-IP: 103.9.41.75* *Customer: 8300005* *Fax: out* Call-ID: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 CSeq: 32590 INVITE Via: SIP/2.0/UDP 35.189.47.13:5060;branch=z9hG4bK9fed.42448b27.0 Via: SIP/2.0/UDP 103.9.41.75:5060 ;rport=5060;received=103.9.41.75;branch=z9hG4bK-f25-3b2a1d-491db0d7 Max-Forwards: 69 Supported: replaces,100rel Contact: <sip:8300005@103.9.41.75:5060;transport=udp> User-Agent: TITAN Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,INFO,NOTIFY,PRACK,MESSAGE,UPDATE Content-Type: application/sdp Content-Length: 461 v=0 o=a0000 8966 6672 IN IP4 103.9.41.75 s=SIP Call c=IN IP4 103.9.41.75 t=0 0 m=audio 53930 RTP/AVP 8 0 101 103 100 104 9 4 102 18 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:103 G726-32/8000 a=rtpmap:100 G726-16/8000 a=rtpmap:104 G726-40/8000 a=rtpmap:9 G722/16000 a=rtpmap:4 G723-63/8000 a=rtpmap:102 G726-24/8000 a=rtpmap:18 G729/8000 a=ptime:10 a=sendrecv a=rtcp-xr:voip-metrics and freeswitch@fax-syd-dev-01> uuid_dump 01784ffa-9df6-11e7-9716-291629b5fb60 Event-Name: CHANNEL_DATA Core-UUID: 499e114c-8330-11e7-b5e4-291629b5fb60 FreeSWITCH-Hostname: fax-syd-dev-01 FreeSWITCH-Switchname: fax-syd-dev-01 FreeSWITCH-IPv4: 10.240.0.23 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2017-09-20%2011%3A22%3A50 Event-Date-GMT: Wed,%2020%20Sep%202017%2011%3A22%3A50%20GMT Event-Date-Timestamp: 1505906570912351 Event-Calling-File: mod_commands.c Event-Calling-Function: uuid_dump_function Event-Calling-Line-Number: 6138 Event-Sequence: 461193 Channel-State: CS_EXECUTE Channel-Call-State: ACTIVE Channel-State-Number: 4 Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 Call-Direction: inbound Presence-Call-Direction: inbound Channel-HIT-Dialplan: true Channel-Call-UUID: 01784ffa-9df6-11e7-9716-291629b5fb60 Answer-State: answered Channel-Read-Codec-Name: L16 Channel-Read-Codec-Rate: 8000 Channel-Read-Codec-Bit-Rate: 128000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Channel-Write-Codec-Bit-Rate: 64000 Caller-Direction: inbound Caller-Logical-Direction: inbound Caller-Username: 8300005 Caller-Dialplan: XML Caller-Caller-ID-Name: 8300005 Caller-Caller-ID-Number: 8300005 Caller-Orig-Caller-ID-Name: 8300005 Caller-Orig-Caller-ID-Number: 8300005 Caller-Network-Addr: 35.189.47.13 Caller-ANI: 8300005 Caller-Destination-Number: 0061862083277 Caller-Unique-ID: 01784ffa-9df6-11e7-9716-291629b5fb60 Caller-Source: mod_sofia Caller-Context: public Caller-Channel-Name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1505906554292369 Caller-Channel-Created-Time: 1505906554292369 Caller-Channel-Answered-Time: 1505906557192357 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 1505906557192357 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Channel-Resurrect-Time: 0 Caller-Channel-Bridged-Time: 0 Caller-Channel-Last-Hold: 0 Caller-Channel-Hold-Accum: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_direction: inbound variable_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 variable_session_id: 4193 variable_video_media_flow: sendrecv variable_channel_name: sofia/external/8300005%40sip99.mondotalk.com%3A5060 variable_sip_invite_stamp: 1505906554292369 variable_sip_received_ip: 35.189.47.13 variable_sip_received_port: 5060 variable_sip_via_protocol: udp variable_sip_from_user_stripped: 8300005 variable_sofia_profile_name: external variable_recovery_profile_name: external variable_sip_invite_route_uri: %3Csip%3A35.189.47.13%3Blr%3Bftag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid%3D985.187c7a27%3E variable_sip_invite_record_route: %3Csip%3A35.189.47.13%3Blr%3Bftag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25%3Bdid%3D985.187c7a27%3E variable_sip_req_params: transport%3Dudp variable_sip_req_user: 0061862083277 variable_sip_req_port: 5060 variable_sip_req_uri: 0061862083277%40sip99.mondotalk.com%3A5060 variable_sip_req_host: sip99.mondotalk.com variable_sip_via_host: 35.189.47.13 variable_sip_via_port: 5060 variable_max_forwards: 69 variable_call_uuid: 01784ffa-9df6-11e7-9716-291629b5fb60 variable_audio_media_flow: sendrecv variable_rtp_audio_recv_pt: 8 variable_rtp_use_codec_name: PCMA variable_rtp_use_codec_rate: 8000 variable_rtp_use_codec_ptime: 10 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: PCMA%408000h%4010i%401c variable_original_read_codec: PCMA variable_original_read_rate: 8000 variable_write_codec: PCMA variable_write_rate: 8000 variable_dtmf_type: rfc2833 variable_local_media_ip: 10.240.0.23 variable_local_media_port: 18268 variable_advertised_media_ip: 35.197.168.74 variable_rtp_use_timer_name: soft variable_rtp_use_pt: 8 variable_rtp_use_ssrc: 1844781053 variable_rtp_2833_send_payload: 101 variable_rtp_2833_recv_payload: 101 variable_remote_media_ip: 103.9.41.75 variable_remote_media_port: 53930 variable_endpoint_disposition: ANSWER variable_api_hangup_hook: lua%20fax_hangup.lua variable_fax_enable_t38_request: true variable_fax_enable_t38: true variable_current_application_data: / variable_current_application: rxfax variable_read_codec: L16 variable_read_rate: 8000 variable_rtp_local_sdp_str: v%3D0%0D%0Ao%3DFreeSWITCH%201505888289%201505888291%20IN%20IP4%2035.197.168.74%0D%0As%3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2035.197.168.74%0D%0At%3D0%200%0D%0Am%3Dimage%2018268%20udptl%20t38%0D%0Aa%3DT38FaxVersion%3A0%0D%0Aa%3DT38MaxBitRate%3A14400%0D%0Aa%3DT38FaxFillBitRemoval%0D%0Aa%3DT38FaxRateManagement%3AtransferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A2000%0D%0Aa%3DT38FaxMaxDatagram%3A400%0D%0Aa%3DT38FaxUdpEC%3At38UDPRedundancy%0D%0A variable_sip_local_network_addr: 35.197.168.74 variable_sip_reply_host: 35.189.47.13 variable_sip_reply_port: 5060 variable_sip_network_ip: 35.189.47.13 variable_sip_network_port: 5060 variable_ep_codec_string: t38 variable_sip_user_agent: TITAN variable_sip_allow: INVITE,%20ACK,%20CANCEL,%20BYE,%20OPTIONS,%20REFER,%20INFO,%20NOTIFY,%20PRACK,%20MESSAGE,%20UPDATE variable_sip_full_via: SIP/2.0/UDP%2035.197.168.74%3A5063%3Breceived%3D35.197.168.74%3Brport%3D5063%3Bbranch%3Dz9hG4bKmeU9KQHKjjS3m variable_sip_full_from: %3Csip%3A0061862083277%40sip99.mondotalk.com %3A5060%3E%3Btag%3D2HQ8NmB9v6Uca variable_sip_to_display: 8300005 variable_sip_full_to: %228300005%22%20%3Csip%3A8300005%40sip99.mondotalk.com %3A5060%3E%3Btag%3D9d3880-6709294b-13c4-50029-f25-3460e079-f25 variable_sip_from_user: 0061862083277 variable_sip_from_port: 5060 variable_sip_from_uri: 0061862083277%40sip99.mondotalk.com%3A5060 variable_sip_from_host: sip99.mondotalk.com variable_sip_to_user: 8300005 variable_sip_to_port: 5060 variable_sip_to_uri: 8300005%40sip99.mondotalk.com%3A5060 variable_sip_to_host: sip99.mondotalk.com variable_sip_contact_params: transport%3Dudp variable_sip_contact_user: 8300005 variable_sip_contact_port: 5060 variable_sip_contact_uri: 8300005%40103.9.41.75%3A5060 variable_sip_contact_host: 103.9.41.75 variable_sip_to_tag: 9d3880-6709294b-13c4-50029-f25-3460e079-f25 variable_sip_from_tag: 2HQ8NmB9v6Uca variable_sip_cseq: 112617216 variable_sip_call_id: 9eda40-6709294b-13c4-50029-f25-31be0b2b-f25 variable_switch_r_sdp: v%3D0%0D%0Ao%3Da0000%208966%206673%20IN%20IP4%20103.9.41.75%0D%0As%3DSIP%20Call%0D%0Ac%3DIN%20IP4%20103.9.41.75%0D%0At%3D0%200%0D%0Am%3Dimage%2053930%20udptl%20t38%0D%0Aa%3DT38FaxVersion%3A0%0D%0Aa%3DT38MaxBitRate%3A14400%0D%0Aa%3DT38FaxRateManagement%3AtransferredTCF%0D%0Aa%3DT38FaxMaxBuffer%3A1800%0D%0Aa%3DT38FaxMaxDatagram%3A256%0D%0Aa%3DT38FaxUdpEC%3At38UDPRedundancy%0D%0Aa%3DT38Fax%0D%0A variable_rtp_use_codec_string: OPUS,G722,PCMU,PCMA,VP8 variable_has_t38: true variable_fax_v17_disabled: 0 variable_fax_ecm_requested: 1 variable_fax_filename: / I am stuck here, please advise. Best Regards,

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FreeSWITCH on Raspberry board
by youssef elouam 20 Sep '17

20 Sep '17

hi all ; I am trying install freeSWITCH in raspberry board (Raspbian OS). but when I run "make && make install" I get as result : make: *** No targets. Stop. could you please help fix this issue. thanks in advance Bestregards --------------------------Youssef ELOUAM.

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RTP Loss with Freeswitch in Default Mode
by vikas sharma 20 Sep '17

20 Sep '17

Hi I am using Freeswitch in Default mode for video calls.But while i am making video calls using two linphone applications on android mobile phone with H264 codec support ,video is getting stuck in between the calls on linphone i.e video is not continuous. *aleg*:*bleg*: Here is my analysis on RTP packet dump taken on the Freeswitch server side: 1. Audio RTP received on *A leg*= *5184* but* forwarded *on *B leg *= 5005 + 35(*private port*) =* 5040*. 2. Audio RTP received on* B leg* = *5178* but forwarded on* A leg* = 4763 + 68(*private port*) = *4831*. 3. Video RTP received on* A leg* = *3145* but forwarded on* B leg* = 3050 + 77(*private port)* = *3127*. 4. Video RTP received on *B leg* = *6402* but forwarded on *A leg* = *6252*. I have also attached here the screenshot of RTP streams from wireshark. This RTP Loss is causing video call quality very bad if network is not too good.Kindly help me to improve the video call quality and reduce RTP loss. Questions: 1. Why some of the RTP packets are being forwarded to private ports? 2. Why there is so much difference in the RTP packets received on the Freeswitch and sent out from the Freeswitch Kindly let me know if i am doing something wrong.

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mod_callcenter events variables
by Peter Nerád 20 Sep '17

20 Sep '17

Hi, Im parsing events from mod_callcenter but I have problem with variables like CC-*-Time .. what is meaning of those numbers ? CC-Member-Joined-Time, CC-Member-Leaving-Time .. I have no idea. ( they are not unix timestamps in microseconds ) Sorry for me bad English :-D

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Verto
by Rick Jarvis 19 Sep '17

19 Sep '17

Looking to get into Verto, in particular handling voice calls with JS. Going through the source, I’m wondering: 1. What’s the difference between the Verto source and the Verto Communicator source? 2. What’s the best way to start from the bottom up - by this I mean that it seems hugely comprehensive, but rather than just use grunt to set it all up, I’d like to start simply with the basics… is there for instance a list of the bare minimum scripts / file structure to use? Apologies if this is a silly question, I’m still relatively new to JS and I don’t want to blow my mind in one go ;) Rick

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JAVA CODE ERROR
by Freeswitch user 19 Sep '17

19 Sep '17

Error:- [ERR] switch_cpp.cpp:683 object is not initalized import org.freeswitch.*; import org.freeswitch.swig.*; public class PhoneTest implements FreeswitchScript, DTMFCallback, HangupHook { public PhoneTest() { } public String onDTMF(Object object, int i, String arg) { if (object instanceof String) freeswitch.console_log("notice", "DTMF: " + (String)object + " ARG: " + arg + "\n"); else freeswitch.console_log("notice", "WOW GOT AN EVENT: " + object.toString()); return "true"; } public void onHangup() { freeswitch.console_log("notice", "HANGUP!\n"); } public void run(String sessionUuid, String args) { freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: " + args + "\n"); JavaSession session = null; try { session = new JavaSession(sessionUuid); session.setDTMFCallback(this, "TEST"); // Getting error in this line. session object not initilizing session.setHangupHook(this); session.answer(); session.streamFile(args, 0); session.hangup(""); } finally { if (session != null) session.delete(); } } }

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install source error
by Макаревич М.А. 19 Sep '17

19 Sep '17

Hi, I am installing freeswitch 1.6.19 with RPI-3,but getting some errors Linux raspberrypi 4.9.41-v7+ #1023 SMP Tue Aug 8 16:00:15 BST 2017 armv7l GNU/Linux. pi@raspberrypi:~/freeswitch-1.6.19 $ make -j3 touch /home/pi/freeswitch-1.6.19/src/include/switch.h CC libfreeswitch_spandsp_la-plc.lo make[1]: Entering directory '/home/pi/freeswitch-1.6.19/libs/srtp' /bin/bash ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c -o srtp.lo `test -f 'srtp/srtp.c' || echo './'`srtp/srtp.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c srtp/srtp.c -fPIC -DPIC -o srtp.o In file included from ./crypto/include/prng.h:17:0, from ./crypto/include/crypto_kernel.h:50, from ./include/srtp.h:53, from srtp/srtp.c:46: ./crypto/include/aes_icm_ossl.h:66:20: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ In file included from srtp/srtp.c:50:0: ./crypto/include/aes_gcm_ossl.h:58:18: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ config.status: creating Makefile config.status: creating include/apr.h config.status: creating build/apr_rules.mk Makefile:513: recipe for target 'srtp.lo' failed make[1]: *** [srtp.lo] Error 1 Please suggest me.

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Dynamic Conference Calls
by Akshaya j 19 Sep '17

19 Sep '17

Hello FS users, I'm a fresher working around with FreeSwitch features and came across this conference room feature. This conference room actually acts like a rooms where anyone can join the room by dialling some number. There is a also moderator who can add any user into the conference room. What i need to know is whether the conventional conference calls is available in FreeSwitch? The conventional conference calls is what we use in our normal mobile phones where any person in a call can add any other person by dialling his number and then all 3 persons can talk together. That is anybody in a call can act as a moderator to add a person to join. I don't find any content regarding this conference in web, is there any other terminology for this feature in Freeswitch which i dont see or whether this is possible? If so how to proceed with this ?? Any help would be highly appreciated.!!! Thanks Advance, PurpleWinged

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