[Freeswitch-users] 1.8 DTLS calls

[Giovanni Maruzzelli]

On Wed, Feb 20, 2019 at 12:59 PM Jurijs Ivolga <jurijs.ivolga at 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=>DTLS
>
> In 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 at 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 at 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 at 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 at 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 at 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 at sip.myapp.net Set rtp dtmf delay to 40
> 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel [sofia/internal/
> 9999999999999999999 at sip.myapp.net] has been answered
> 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773 (sofia/internal/
> 9999999999999999999 at sip.myapp.net) Callstate Change RINGING -> ACTIVE
> 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248 sofia/internal/
> 1234567890 at 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 at sip.myapp.net.
> 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP
> sofia/internal/1234567890 at sip.myapp.net:
>
> Jurijs
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-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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