Hello, 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.
Thanks. On Thu, 23 Feb 2023 at 12:13, David Cunningham <dcunning...@voisonics.com> wrote: > 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 http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782
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