[asterisk-users] PJSIP Losing knowledge of external_media_address
Mark Murawski
markm-lists at intellasoft.net
Fri Aug 18 11:46:34 CDT 2023
On 8/18/23 12:41, Joshua C. Colp wrote:
> On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski
> <markm-lists at intellasoft.net> 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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