Re: PJSIP Losing knowledge of external_media_address

On 8/18/23 12:41, Joshua C. Colp wrote:
> On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski 
> <[email protected]> wrote:
>
>     I've seen this happen three times in the wild now.  I've been
>     trying to
>     isolate the source of the issue, but so far it seems like there's not
>     enough debug output to know why this occurs.
>
>     Long story short:
>     - Start Asterisk
>     - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is
>     behind
>     NAT).  SIP is handled correctly, Asterisk responds OK with RTP media
>     address of external_media_address
>     - After 30 minutes to an hour or sometimes months later after
>     startup,
>     upon receiving INVITE from ITSP via WAN, Asterisk responds OK with
>     INTERNAL LAN IP instead of external_media_address
>     - I've observed this occur after 30 minutes from startup with no
>     configuration changes that were made or any pjsip reloads done during
>     this period
>
>
> <snip>
>
>
>
>     Attached sip sessions and debug log... the only thing I found
>     interesting was finding a lack of a log item
>     We SHOULD be seeing:
>     DEBUG[XXXXX] res_pjsip_session.c: (null session): Setting external
>     media
>     address to 152.X.Y.Z
>     This message is clearly lacking from the debug session where the
>     incorrect media address is sent.  But there's not enough detail in
>     the
>     debugs to see why this decision was not made to use
>     external_media_address
>
>
> Can't you just extend the debug and add further logging to understand 
> the choices being made and why?

Doing that now!

>
>     By default we use nat settings for all our endpoints, but
>     obviously it's
>     not required here for an ITSP that has trustworthy media ports in the
>     SDP.  Maybe a bandaid is turning off rewrite_contact for this
>     endpoint?
>     Going to try that as soon as possible.
>
>
> I believe I've stated this once or twice when you've brought this 
> issue up on IRC but rewrite_contact has no influence or impact on 
> this. It rewrites incoming Contact headers to the source IP address 
> and port of the SIP message. You can turn it on if you wish, but it is 
> unlikely to do anything.


Sorry, I missed this on IRC.  Thanks.  Makes sense

>
> With the limited insight into things it could be a bug. I haven't seen 
> any other reports, and haven't received any reports from other Sangoma 
> products. Is this with a mainline Asterisk, or is it your patched 
> version of Asterisk? It should be confirmed on normal Asterisk.

Thanks, very curious if this has come up for anyone else.  This is a 
slightly patched asterisk but nothing that would change the outcome of 
any nat handling or decision making (additional logging updates to pjsip 
only)

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