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

[04dco (JIRA)]

04dco created ASTERISK-29957:
--------------------------------

             Summary: 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: pjproject/pjsip
    Affects Versions: 18.10.1, 16.24.1
         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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