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[Freeswitch-dev] Questions
by defender oogway 07 Aug '21

07 Aug '21

Hi, I would like to first start off by saying I love freeswitch, this industry. - What I'm trying to do: - Get an audio bytes stream from FreeSwitch of a channel, that has been answered and forwarded to the 8084 socket (using ESL Connection) - I'd like to listen live on the stream, and intercept and put new content audio on the stream. - Language I'm using is python - What I've tried: - Using Unicast ESL feature (no data gets sent but I do see "connection initiated) - Using the MOD_RTMP module but have not been able to bridge the the 2 CHANNELS together and read from the MOD_RTMP stream to listen and insert data to that stream - What i'm trying to look into but getting stuck with finding the correct documentation to start: - Learning how modules work, how to run a module, and setting variables so I can create my own unicast stream or something similar to do this. I really love this project and want to contribute. Could someone please help. Sincerely, David Germiquet

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[Freeswitch-dev] Heartbeat between WebRTC Client and Freeswitch
by Qihong Wang 24 May '21

24 May '21

Hi, I'm new to Freeswitch and real time communication. However, I'm involving a project which uses Freeswitch gateway for WebRTC communication. The client, mostly JavaScript code, uses Websocket to send media data and also the heartbeat regardless the ping/pong of Websocket. Freeswitch must respond that heartbeat. I'm not sure where to start. I'm thinking src/mod/endpoints/mod_rtc/mod_rtc.c or mod_verto.c/mod_verto.c, ws.c might be the right place to add heartbeat handling? The second thought is using Freeswitch's event system, especially the buildin heartbeat event. If so, our client needs to give up its own heartbeat, although the possibility is very low. I'd love to hear how to utilize Freeswitch's heartbeat event between our client and Freeswitch gateway. Thanks in advance! Qihong

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[Freeswitch-dev] DTMF dropped due to remote party re-starting packet numbers from 0 for each digit
by Andy Newlands 19 Jan '21

19 Jan '21

Hi, A 3rd-party is sending each RFC2833 DTMF digit with a new SSRC and starts the packet sequence numbering at ZERO for each new digit. I can see, in switch_rtp.c: handle_rfc2833(), that FS makes a note (in the session) of the last DTMF packet it processed, then drops any subsequent packets with a sequence number less than that. My questions are: 1. Is the remote-party compliant with relevant RFCs when they set a new SSRC for each DTMF digit? 2. Is the remote party compliant with relevant RFCs when they re-start packet sequence numbers from 0 for each DTMF digit? 3. Looking at the source for handle_rfc2833(), I can't see any setting which could be used to force FS to 'tolerate' the above behaviour 4. If I alter handle_rfc2833(), to cope with that behaviour, am I likely to run into problems with other remote systems (E.G. possibly ending up with duplicate digits) Thanks, Andy

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[Freeswitch-dev] xmlrpc websocket bug and fix
by Royce Mitchell III 14 Jan '21

14 Jan '21

I have found and fixed a bug in mod_xml_rpc's websocket code. I know my fix is not comprehensive, but it does prevent the 100% reproducible catastrophic behavior in FreeSWITCH that I'm experiencing specifically. Would one of the developers be willing to review my findings and convert them into a more proper fix please? https://github.com/signalwire/freeswitch/issues/1031 Thanks! Royce Mitchell, IT Consultant ITAS Solutions royce3(a)itas-solutions.com There are three hard problems in computer science: naming things, and off-by-one errors.

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Re: [Freeswitch-dev] HELP DTLS HANDSHAKE to FAIL chromeV87 (Kaiduan Xie)
by Cédric Clavier 08 Dec '20

08 Dec '20

Thank you very much it works. It save me from a very difficult and speed server migration (ubuntu 16.04 -> ubuntu 20.04). Cédric Clavier Chef de projet R&D 06 25 00 45 88 4 rue André Marie Ampère - 22300 Lannion <https://saooti.com/> <https://twitter.com/saooti> <https://www.linkedin.com/company/291543/admin/> > Le 8 déc. 2020 à 13:00, freeswitch-dev-request(a)lists.freeswitch.org a écrit : > > Send FreeSWITCH-dev mailing list submissions to > freeswitch-dev(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > or, via email, send a message with subject or body 'help' to > freeswitch-dev-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-dev-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-dev digest..." > Today's Topics: > > 1. Re: HELP DTLS HANDSHAKE to FAIL chromeV87 (Kaiduan Xie) > > De: Kaiduan Xie <kaiduanx(a)gmail.com> > Objet: Rép. : [Freeswitch-dev] HELP DTLS HANDSHAKE to FAIL chromeV87 > Date: 4 décembre 2020 à 21:00:39 UTC+1 > À: freeswitch-dev(a)lists.freeswitch.org > Cc: "freeswitch-users-request(a)lists.freeswitch.org" <freeswitch-users-request(a)lists.freeswitch.org>, FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > If you can build Freeswitch, change the code in switch_rtp.c as below, > > #if OPENSSL_VERSION_NUMBER >= 0x10100000 > dtls->ssl_ctx = SSL_CTX_new((type & DTLS_TYPE_SERVER) ? DTLS_server_method() : DTLS_client_method()); > #else > - dtls->ssl_ctx = SSL_CTX_new((type & DTLS_TYPE_SERVER) ? DTLSv1_server_method() : DTLSv1_client_method()); > + switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_INFO, > + "Negotiated DTLS1.2 between remote-host:%s, local-host:%s\n", rtp_session->remote_host_str, rtp_session->local_host_str); > + dtls->ssl_ctx = SSL_CTX_new((type & DTLS_TYPE_SERVER) ? DTLSv1_2_server_method() : DTLSv1_2_client_method()); > #endif > > /Kaiduan > > On Fri, Dec 4, 2020 at 1:09 PM Jazmin Marina Florez Marino <Jazmin.Marino(a)on24.com <mailto:Jazmin.Marino@on24.com>> wrote: > Hi > > I have a verto client trying to connect to FS using Chrome V87 but we have DTLS problems. > > The openssl version we are using on FS side is OpenSSL 1.0.2k-fips > > FreeSWITCH (Version 1.8.4 64bit) > > > > > > FS LOG - Chrome V87 FAILED CONNECTION > > > > 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure 1 > > 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS state from HANDSHAKE to FAIL > > 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure 1 > > 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS state from HANDSHAKE to FAIL > > 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] > > 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 Video thread ended > > 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving conference, cause: DESTINATION_OUT_OF_ORDER > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 skip receive message [TRANSFER] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 Restore previous codec opus:116. > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup already) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 (verto.rtc/3520) State EXECUTE going to sleep > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER > > 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP going to sleep > > > > The weird thing is the connection works ok when the version of chrome is V86. Some weeks ago chrome was updated and we detected this issue. > > > > FS LOG - ChromeV86 CONNECTION OK > > > > 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS state from HANDSHAKE to SETUP > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint Verified. > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video Secure RTP SEND > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video Secure RTP RECV > > 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS state from SETUP to READY > > 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: video ssrc[2719546543] base_seq[1999] > > 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS state from HANDSHAKE to SETUP > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint Verified. > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV > > 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS state from SETUP to READY > > 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community <https://signalwire.community/> > > Professional FreeSWITCH Services > sales(a)freeswitch.com <mailto:sales@freeswitch.com> > https://freeswitch.com <https://freeswitch.com/> > > Official FreeSWITCH Sites > https://freeswitch.com/oss <https://freeswitch.com/oss> > https://freeswitch.org/confluence <https://freeswitch.org/confluence> > https://cluecon.com <https://cluecon.com/> > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org <mailto:FreeSWITCH-dev@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev <http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev <http://lists.freeswitch.org/mailman/options/freeswitch-dev> > https://freeswitch.com <https://freeswitch.com/> > > -- > Founder of Goodstartsoft > https://www.goodstartsoft.com <https://www.goodstartsoft.com/> > > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > https://freeswitch.com

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[Freeswitch-dev] HELP DTLS HANDSHAKE to FAIL chromeV87
by Jazmin Marina Florez Marino 04 Dec '20

04 Dec '20

Hi I have a verto client trying to connect to FS using Chrome V87 but we have DTLS problems. The openssl version we are using on FS side is OpenSSL 1.0.2k-fips FreeSWITCH (Version 1.8.4 64bit) FS LOG - Chrome V87 FAILED CONNECTION 2020-12-03 11:23:06.762291 [ERR] switch_rtp.c:3199 video Handshake failure 1 2020-12-03 11:23:06.762291 [INFO] switch_rtp.c:3200 Changing video DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.832295 [ERR] switch_rtp.c:3199 audio Handshake failure 1 2020-12-03 11:23:06.832295 [INFO] switch_rtp.c:3200 Changing audio DTLS state from HANDSHAKE to FAIL 2020-12-03 11:23:06.852295 [NOTICE] switch_rtp.c:3181 Hangup verto.rtc/3520 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2020-12-03 11:23:06.852295 [DEBUG] switch_core_media.c:7470 verto.rtc/3520 Video thread ended 2020-12-03 11:23:06.872305 [INFO] conference_loop.c:1670 Channel leaving conference, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2639 verto.rtc/3520 skip receive message [DISPLAY] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] mod_conference.c:2467 verto.rtc/3520 skip receive message [TRANSFER] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_media.c:12220 verto.rtc/3520 skip receive message [BITRATE_REQ] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_codec.c:248 verto.rtc/3520 Restore previous codec opus:116. 2020-12-03 11:23:06.872305 [DEBUG] switch_core_session.c:2886 verto.rtc/3520 skip receive message [PHONE_EVENT] (channel is hungup already) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:650 (verto.rtc/3520) State EXECUTE going to sleep 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:584 (verto.rtc/3520) Running State Change CS_HANGUP (Cur 1 Tot 3) 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:847 (verto.rtc/3520) Callstate Change ACTIVE -> HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:60 verto.rtc/3520 Standard HANGUP, cause: DESTINATION_OUT_OF_ORDER 2020-12-03 11:23:06.872305 [DEBUG] switch_core_state_machine.c:849 (verto.rtc/3520) State HANGUP going to sleep The weird thing is the connection works ok when the version of chrome is V86. Some weeks ago chrome was updated and we detected this issue. FS LOG - ChromeV86 CONNECTION OK 2020-12-03 11:19:19.622294 [INFO] switch_rtp.c:3206 Changing video DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3113 video Fingerprint Verified. 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4104 Activating video Secure RTP SEND 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:4082 Activating video Secure RTP RECV 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.672317 [INFO] switch_rtp.c:3155 Changing video DTLS state from SETUP to READY 2020-12-03 11:19:19.672317 [DEBUG] switch_rtp.c:1890 rtcp_stats_init: video ssrc[2719546543] base_seq[1999] 2020-12-03 11:19:19.672317 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3206 Changing audio DTLS state from HANDSHAKE to SETUP 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3113 audio Fingerprint Verified. 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80 2020-12-03 11:19:19.702298 [INFO] switch_rtp.c:3155 Changing audio DTLS state from SETUP to READY 2020-12-03 11:19:19.702298 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:dtls:AES_CM_128_HMAC_SHA1_80

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[Freeswitch-dev] Regarding Registration error
by Alaghu Sundram 28 Oct '20

28 Oct '20

Hello Team, I found error when I register vertility freeswitch(a)fsm.intelliverse.com> sofia_reg.c:2458 Changing expire time to 60 by request of proxy sip:sofia_reg.c:2458 Changing expire time to 60 by request of proxy I also found error registering other voip Registration failed. Too much hope list Thanks Alaghu sundram

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[Freeswitch-dev] A question about 16 bit linearisation in mod_conference
by Ghazi Triki 25 Sep '20

25 Sep '20

Hello guys, I have few questions regarding the 16 bit linearisation in mod_conference. Especially on those lines https://github.com/signalwire/freeswitch/blob/master/src/mod/applications/m… 1 - What does 16 bit linearisation do? And what it is technically? 2 - Does it have an impact on the mixing quality? 3 - Is it considered as a transcoding, an intermediate encoding between the input streams and the final output stream? Thanks

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[Freeswitch-dev] switch_app_execute() ???
by Andrew Clayton 12 Aug '20

12 Aug '20

Hi, So there is switch_api_execute() for executing API commands (such as those defined via SWITCH_ADD_API()). However I'm not seeing a switch_app_execute() equivalent for executing application commands (such as those defined via SWITCH_ADD_APP()). What am I missing? Cheers, Andrew

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[Freeswitch-dev] Server Requirement for Running 1000 Con current G729 Calls
by amitsharma＠coraltele.com 12 Aug '20

12 Aug '20

Hello, Need one thing to know what will be the Server Requirement for running 1000 Concurrent G729 call with Free Switch. Any information will be appreciated. Thanks & Regards, Amit Sharma (Sr. Team Leader) (An ISO 9001:2008 company) Mobile: <tel:9891612004> tel:9891612004 PH: +91 120 2595870 Ext.: <tel:870> tel:870 Email : <mailto:amitsharma@coraltele.com> amitsharma(a)coraltele.com Web : <blocked::http://www.coraltele.com> www.coraltele.com

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