Bug#855014: src:asterisk: Asymetric RTP codec leads to one-way audio

[Bernhard Schmidt]

Package: src:asterisk
Version: 1:13.13.1~dfsg-4
Severity: important
Tags: patch upstream
Forwarded: https://issues.asterisk.org/jira/browse/ASTERISK-26603

Normal SIP behaviour is to use one RTP codec in both directions of a call. 
PJSIP has recently gained support for this, but it is incomplete and broken.

This behaviour leads to one-way audio when using PJSIP.

A fix will be released in 13.14.0.