[asterisk-bugs] [JIRA] (ASTERISK-29957) Transport autoselection is broken on FreeBSD

[Joshua C. Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29957?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258321#comment-258321 ] 

Joshua C. Colp commented on ASTERISK-29957:
-------------------------------------------

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Additionally: FreeBSD is not a project supported environment. Once debug information is provided I will mark this as open, but it will be up to the community to do anything.

> Transport autoselection is broken on FreeBSD
> --------------------------------------------
>
>                 Key: ASTERISK-29957
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29957
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 16.24.1, 18.10.1
>         Environment: OS: FreeBSD 13.0-RELEASE
> Architecture: amd64
> Environment: jail(8)
>            Reporter: 04dco
>
> PJSIP transport autoselection fails and the last defined transport in pjsip.conf is used if not explicitly specified on endpoint, as a result no audio gets through. For example if the last transport in pjsip.conf is over IPv6, calls over IPv4 have no audio. RTP Packet Debugging says packets of -13 bytes are send to the endpoint but none are received back. When the endpoint tries to send RTP packets to Asterisk, they bounce with ICMP port-unreachable.
> Issue is as described in this older forum post:
> https://community.asterisk.org/t/13-14-0-pjsip-rtp-problem-no-audio/70244
> RTP Packet Debugging output excerpt:
> [Mar  6 23:17:40] Sent RTP packet to      10.0.1.5:33024 (type 09, seq 020789, ts 000160, len -000013)
> [Mar  6 23:17:40] Sent RTP packet to      10.0.1.5:33024 (type 09, seq 020790, ts 000320, len -000013)
> [Mar  6 23:17:40] Sent RTP packet to      10.0.1.5:33024 (type 09, seq 020791, ts 000480, len -000013)
> Forcing transport fixes it but for one address family only. If I force an IPv4 UDP transport then RTP over IPv6 breaks with the usual -13 byte packets and vice-versa. If I force an IPv4 UDP transport, TLS over IPv4 works normally.
> Steps to reproduce: specify two UDP transports in pjsip.conf, one over IPv4 then one over IPv6 (in that order). Make a call over IPv4 and fail to get audio.
> Workarounds: explicitly specify transport for endpoints



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