Interoperability: Asterisk 13 on re-INVITE

[traud]

When Asterisk started the call but the remote party sends a re-INVITE on the SIP layer, the call might face one-way audio. One such a call scenario is call hold, for example. One-way audio happens because this codec here uses a dynamic RTP payload type in SDP negotiation, which Asterisk 13 LTS does not support correctly. Complete details are documented in ASTERISK-27056. Even if you are not affected, applying the patch provided there is recommended for any user. If you face an issues with that patch, report in ASTERISK-27056, please. If possible, consider upgrading to a newer Asterisk branch.