Does anyone know if one of the "strictrtp" options disables RTP learning?
As far as I can tell from the documentation the values "no" and "seqno" are
more permissive in allowing other sources rather than less, but I thought
I'd check.


On Thu, 23 Feb 2023 at 12:13, David Cunningham <dcunning...@voisonics.com>

> Hello,
> We have a system that interoperates with an external service, so that the
> basic call flow is:
> PSTN origination -> Asterisk A -> External service -> Asterisk B
> Initially the SDP from the external service tells the two Asterisks to
> send RTP directly to each other. Part way through the call the external
> service sends re-INVITEs both Asterisks to change the address for audio to
> itself, but this fails to work intermittently. The problem seems to be one
> of timing.
> If there's no RTP between the two re-INVITEs then it works fine, and both
> Asterisks send future RTP to the external service as instructed.
> The problem is if RTP is transmitted/received in the fraction of the
> second between the two re-INVITEs. If Asterisk A receives the re-INVITE
> first, and then receives RTP from Asterisk B (which hasn't yet received its
> re-INVITE), then it re-learns the media address of Asterisk B and sends
> audio there instead of the new address. Asterisk B gets the second
> re-INVITE with the new media address, but soon re-learns the media address
> of Asterisk A because it's getting RTP from it.
> Note we have "canreinvite = no" in sip.conf, but I don't think that's
> relevant to the problem.
> Can anyone suggest how to prevent this problem? Is it possible to turn off
> learning the media address per call or per peer?
> Thanks for your help.
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782

David Cunningham, Voisonics Limited
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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