[asterisk-bugs] [JIRA] (ASTERISK-21374) [patch] One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX

[Michael L. Young (JIRA)]

Michael L. Young created ASTERISK-21374:
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             Summary: [patch] One-way Audio With auto_* NAT Settings When SIP Calls Initiated By PBX
                 Key: ASTERISK-21374
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21374
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/General
    Affects Versions: 11.3.0
            Reporter: Michael L. Young


I found another case where the force_rport and comedia flags are not being set automatically when using the new auto_* settings.  This time it involves calls initiated by the PBX.

When we reload asterisk the default flags turned on and off by auto_force_rport (force_rport) and auto_comedia (comedia) go back to the default setting of off.  These flags are turned on, as needed, when a peer re-registers or initiates a call.  This would apply to even just having the default global setting "nat=auto_force_rport".

Everything is good except in the following scenario:

We reload Asterisk and the peer's registration has not expired.  We load in the default settings for the peer which turns force_rport and comedia back to off.  Since the peer has not re-registered or placed a call yet, they remain off.  We then initiate a call to the peer from the PBX.  The force_rport and comedia flags stay off.  If NAT is involved, we end up with one-way audio since we never checked to see if the peer is behind NAT or not.

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