About this case: the old SIP channel behaves correctly.
On Sun, May 17, 2020 at 2:44 AM Saint Michael <[email protected]> wrote:
> My phone is located behind a NAT, 172.16.0.0/21.
> Asterisk 16 is on a public IP.
> PJSIP has the config below:
> force_rport=yes
> direct_media=yes
> disable_direct_media_on_nat = yes
> direct_media_method=invite
>
> But when I send a call I see the RTP being sent to my private address, vs
> the public IP. This only happens when Asterisk has dialed the call to
> another carrier. If instead of Dial I choose Answer() and MusicOnHold, then
> the RTP gets shipped to the right address.
> This is a sample of the erroneous behavior:
> Got RTP packet from XX.XX.XX.XX:17510 (type 00, seq 024786, ts 017440,
> len 000160)
> Sent RTP packet to 172.16.7.254:50798 (type 00, seq 010736, ts
> 017440, len 000160)
>
> 172.16.7.254 is my private address.
> What am I missing? Should I open a bug?
> Asterisk should never, ever send RTP to a private address when Asterisk
> itself is on a public IP.
> Before you ask, the dialplan is 3 lines,
> '_X.' => 1. NoOP()
> 2. Dial(PJSIP/${EXTEN}@carrier)
> 3. Hangup()
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