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<div class="moz-cite-prefix">On 8/18/23 12:41, Joshua C. Colp wrote:<br>
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<div dir="ltr">On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski
<<a href="mailto:markm-lists@intellasoft.net"
moz-do-not-send="true" class="moz-txt-link-freetext">markm-lists@intellasoft.net</a>>
wrote:<br>
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style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">I've
seen this happen three times in the wild now. I've been
trying to <br>
isolate the source of the issue, but so far it seems like
there's not <br>
enough debug output to know why this occurs.<br>
<br>
Long story short:<br>
- Start Asterisk<br>
- PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk
is behind <br>
NAT). SIP is handled correctly, Asterisk responds OK with
RTP media <br>
address of external_media_address<br>
- After 30 minutes to an hour or sometimes months later
after startup, <br>
upon receiving INVITE from ITSP via WAN, Asterisk responds
OK with <br>
INTERNAL LAN IP instead of external_media_address<br>
- I've observed this occur after 30 minutes from startup
with no <br>
configuration changes that were made or any pjsip reloads
done during <br>
this period<br>
<br>
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<div><snip></div>
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<br>
<br>
Attached sip sessions and debug log... the only thing I
found <br>
interesting was finding a lack of a log item<br>
We SHOULD be seeing:<br>
DEBUG[XXXXX] res_pjsip_session.c: (null session): Setting
external media <br>
address to 152.X.Y.Z<br>
This message is clearly lacking from the debug session where
the <br>
incorrect media address is sent. But there's not enough
detail in the <br>
debugs to see why this decision was not made to use
external_media_address<br>
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<div>Can't you just extend the debug and add further logging
to understand the choices being made and why?</div>
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<br>
Doing that now!<br>
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<br>
By default we use nat settings for all our endpoints, but
obviously it's <br>
not required here for an ITSP that has trustworthy media
ports in the <br>
SDP. Maybe a bandaid is turning off rewrite_contact for
this endpoint? <br>
Going to try that as soon as possible.<br>
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<div>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.</div>
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<br>
Sorry, I missed this on IRC. Thanks. Makes sense<br>
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<div>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.</div>
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<br>
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)<br>
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