_________________________________________________________________________That seems to be a nat problem, or an ice misconfiguration, like using wrong stun (that results in a nat problem, again).I assume all is equal in server.Check to see if you can give different parameters to clients, maybe between releases Freeswitch changed in subtle modes how it react to subtle client parameters. I would check all nat/ice/stun client parameters and experiment with them...Just a possible line of investigation, no warranties ;)-giovanniOn Wed, Feb 20, 2019, 16:18 Jurijs Ivolga <jurijs.ivolga@gmail.com wrote:Hi Giovanni,Thank you for tips!There is nothing happening after what I sent, so call just stuck there. Only if I hangup on any side.Only I can to add that after 30 seconds I just get this line:2019-02-20 14:22:07.195060 [ERR] switch_rtp.c:834 No audio stun for a long time!If I hangup calls on both side, call still shows in freeswitch as active if I do show calls.I believe I have everything needed for trans-coding in config files, cause calls where no DTLS involved but with transcoding, calls works fine. In that case I have RTP(ALAW) => Freeswitch => SRTP(OPUS)Regarding ICE:2019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8690 Activating Audio ICE2019-02-20 12:51:21.532911 [NOTICE] switch_rtp.c:4799 Activating RTP audio ICE: 1cf2011d:u4NGJi1myDzrpRoh 10.99.88.122:554652019-02-20 12:51:21.532911 [DEBUG] switch_core_media.c:8731 Activating RTCP PORT 554652019-02-20 12:51:21.532911 [DEBUG] switch_rtp.c:4696 RTCP send rate is: 1000 and packet rate is: 20000 Remote Port: 554652019-02-20 12:51:21.532911 [INFO] switch_core_media.c:8742 Skipping RTCP ICE (Same as RTP)Jurijs_________________________________________________________________________On Wed, Feb 20, 2019 at 4:17 PM Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:You stopped the log much before the call ends... Can you pastebin a complete log, up to DESTROY CS ?Also, out of memory, do you have the things related to transcode media and mixing media activated in config? Do FreeSWITCH find a media candidate during ICE bargaining?If you pastebin a complete log maybe we can help more-giovanniOn Wed, Feb 20, 2019 at 2:58 PM Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:Hi,I doubt that this is in network or configuration error. Everything are same between 1.6 & 1.8. I'm using docker and difference are only in Docker file where I build freeswitch. If I rebuild with old Docker file(1.6) then it works fine, but with new Docker file for 1.8, I have problems. Everything else seems to work, but DTLS calls...Logs:Jurijs_________________________________________________________________________On Wed, Feb 20, 2019 at 2:47 PM Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:_________________________________________________________________________On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <jurijs.ivolga@gmail.com> wrote:Hi,I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem with DTLS calls(media_webrtc=true).Does anybody has same issues as me?I have following setup:RTP=>Freeswitch=>DTLSIn 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never sends it back to RTP endpoint.maybe you have a network problem, or a configuration problem network related in 1.8 deployment?Also, for us to better understand: from fs cli, "sofia global siptrace on", after this, copy and paste the ENTIRE debug level output from beginning to end in a pastebin.Then put here the pastebin link-giovanni_________________________________________________________________________Here some snippet from logs, where call stuck:2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio DTLS state from OFF to HANDSHAKE
2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815 sofia/internal/9999999999999999999@sip.myapp.net Set 2833 dtmf send payload to 101
2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822 sofia/internal/9999999999999999999@sip.myapp.net Set 2833 dtmf receive payload to 101
2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845 sofia/internal/9999999999999999999@sip.myapp.net Set rtp dtmf delay to 40
In 1.6.20 everything works as expected:
2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180 sofia/internal/9999999999999999999@sip.myapp.net Set 2833 dtmf send payload to 101
2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187 sofia/internal/9999999999999999999@sip.myapp.net Set 2833 dtmf receive payload to 101
2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210 sofia/internal/9999999999999999999@sip.myapp.net Set rtp dtmf delay to 40
2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel [sofia/internal/9999999999999999999@sip.myapp.net] has been answered
2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773 (sofia/internal/9999999999999999999@sip.myapp.net) Callstate Change RINGING -> ACTIVE
2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248 sofia/internal/1234567890@sip.myapp.net Restore previous codec PCMA:8.
2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/internal/1234567890@sip.myapp.net.
2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP sofia/internal/1234567890@sip.myapp.net:Jurijs
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--Sincerely,_________________________________________________________________________
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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